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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Date: Fri, 4 Oct 2013 18:48:29 -0400
Thread-Topic: NOMCOM 2013 - Second Call for Nominations - two weeks left
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Subject: [tsvwg] FW: NOMCOM 2013 - Second Call for Nominations - two weeks left
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From michawe@ifi.uio.no  Tue Oct  8 04:58:08 2013
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From: Michael Welzl <michawe@ifi.uio.no>
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Subject: [tsvwg] Call for participation: Transport Services
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Dear all,

Sorry if you receive multiple copies of this! I sent it to all the lists =
with potentially interested folks...  (this should be okay to do =
according to RFC5434, which says "various mailing lists").

A community of interest is being formed to gauge whether there is =
sufficient interest to host a BOF at the London IETF, on the topic of =
"Transport Services". The intention is to create a WG that would define =
the set of services that transport protocols offer to applications, such =
that applications could at some point in the future request a service, =
not a protocol. This could foster innovation (protocols could be tried =
out, with a fall-back, without requiring the application programmer to =
include such functionality); it could also give more freedom to whatever =
resides below the API to automatically make the best out of what it =
currently has available.

If you're interested in this problem space, please visit the related =
webpage that I have set up:
https://sites.google.com/site/transportprotocolservices/
There you'll find an initial stab at a charter and all information about =
the mailing list - please join us and participate in discussions! =
Thanks!

Cheers,
Michael=

--Apple-Mail=_24569281-1FE8-4470-8D5D-E83F1FF5D84E
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div>Dear all,</div><div><br></div><div>Sorry if you receive multiple =
copies of this! I sent it to all the lists with potentially interested =
folks... &nbsp;(this should be okay to do according to RFC5434, which =
says "various mailing lists").</div><div><br></div><div>A community of =
interest is being formed to gauge whether there is sufficient interest =
to host a BOF at the London IETF, on the topic of "Transport Services". =
The intention is to create a WG that would define the set of services =
that transport protocols offer to applications, such that applications =
could at some point in the future request a service, not a protocol. =
This could foster innovation (protocols could be tried out, with a =
fall-back, without requiring the application programmer to include such =
functionality); it could also give more freedom to whatever resides =
below the API to automatically make the best out of what it currently =
has available.</div><div><br></div><div>If you're interested in this =
problem space, please visit the related webpage that I have set =
up:</div><div><a =
href=3D"https://sites.google.com/site/transportprotocolservices/">https://=
sites.google.com/site/transportprotocolservices/</a></div><div>There =
you'll find an initial stab at a charter and all information about =
the&nbsp;mailing list - please join us and participate in discussions! =
Thanks!</div><div><br></div><div>Cheers,</div><div>Michael</div></body></h=
tml>=

--Apple-Mail=_24569281-1FE8-4470-8D5D-E83F1FF5D84E--

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From: Nico Williams <nico@cryptonector.com>
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Subject: Re: [tsvwg] The List (of application-layer desired features)
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--047d7b5d9c6b68680204e52d03df
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- HTTP needs an XID so as to make it less dependent on octet stream
transports.  HTTP chunks, so there's no need for sub-channels, as long as
an XID can be used to dissociate reply order from request order.

- SCTP and similar should be implemented in user-space -- the OS/NIC should
know nothing more than how to classify packets for dispatch purposes.  UDP
obviously facilitates this and has the benefit that middle boxes deal with
it.

- Multi-homing and name resolution should be abstracted from the app, even
if they are handled within the app (in a library).

- The tricky things are a) security (SYN forgery type attacks), and b)
congestion control.  We know how to do (a) with low latency sometimes
(e.g., see TCP Fast Open).  (b) is trickier; read with care (has the IAB
said anything about this?).

- User/service authentication and session crypto are orthogonal to the
above: TLS and similar can be adapted to unreliable, unordered datagram
delivery, and they have been (DTLS).  This is not to say that we have
nothing to do in this space, just that it's mostly orthogonal to the
transport issues/wish-list.

Nico
--

--047d7b5d9c6b68680204e52d03df
Content-Type: text/html; charset=UTF-8
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<p dir=3D"ltr">- HTTP needs an XID so as to make it less dependent on octet=
 stream transports.=C2=A0 HTTP chunks, so there&#39;s no need for sub-chann=
els, as long as an XID can be used to dissociate reply order from request o=
rder.</p>

<p dir=3D"ltr">- SCTP and similar should be implemented in user-space -- th=
e OS/NIC should know nothing more than how to classify packets for dispatch=
 purposes.=C2=A0 UDP obviously facilitates this and has the benefit that mi=
ddle boxes deal with it.</p>

<p dir=3D"ltr">- Multi-homing and name resolution should be abstracted from=
 the app, even if they are handled within the app (in a library).</p>
<p dir=3D"ltr">- The tricky things are a) security (SYN forgery type attack=
s), and b) congestion control.=C2=A0 We know how to do (a) with low latency=
 sometimes (e.g., see TCP Fast Open).=C2=A0 (b) is trickier; read with care=
 (has the IAB said anything about this?).</p>

<p dir=3D"ltr">- User/service authentication and session crypto are orthogo=
nal to the above: TLS and similar can be adapted to unreliable, unordered d=
atagram delivery, and they have been (DTLS).=C2=A0 This is not to say that =
we have nothing to do in this space, just that it&#39;s mostly orthogonal t=
o the transport issues/wish-list.</p>

<p dir=3D"ltr">Nico<br>
-- </p>

--047d7b5d9c6b68680204e52d03df--

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From: Nico Williams <nico@cryptonector.com>
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Cc: tsvwg <tsvwg@ietf.org>, Joe Touch <touch@isi.edu>, HTTP Working Group <ietf-http-wg@w3.org>, Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Subject: Re: [tsvwg] The List (of application-layer desired features)
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Date: Thu, 05 Sep 2013 03:14:21 -0000
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Yes, deployable means: doesn't require middle-box updates
(NAAAAAAAAATs) and doesn't require OS vendors / kernel developers /
distros to develop, package, and deliver an implementation.  That
pretty much means: plain TCP and plain UDP.  Not IPsec (sure, IPsec
can run over UDP, but apps can't all run their own IKEs), not SCTP
(unless it's over UDP), not TCP-AO or anything of the sort.

Things like SCTP over UDP will do.  Lighter-weight protocols over UDP
too.  Just make sure to handle congestion.

DJB's MinimaLT looks pretty good.

Nico
--

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Subject: Re: [tsvwg] The List (of application-layer desired features)
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On 9/4/2013 8:14 PM, Nico Williams wrote:
> Yes, deployable means: doesn't require middle-box updates
> (NAAAAAAAAATs) and doesn't require OS vendors / kernel developers /
> distros to develop, package, and deliver an implementation.  That
> pretty much means: plain TCP and plain UDP.  Not IPsec (sure, IPsec
> can run over UDP, but apps can't all run their own IKEs), not SCTP
> (unless it's over UDP), not TCP-AO or anything of the sort.
>
> Things like SCTP over UDP will do.

By your set of requirements, *anything* over UDP would work fine. 
Including IPsec, TCP-AO, and anything else you can dream up.

Thank you for moving the stack up one level, adding 20 bytes of header, 
and constraining the number of connections between two hosts (due to the 
limit of port numbers).

Joe


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Subject: Re: [tsvwg] The List (of application-layer desired features)
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On 10/10/2013 10:28 AM, Joe Touch wrote:
>
>
> On 9/4/2013 8:14 PM, Nico Williams wrote:
>> Yes, deployable means: doesn't require middle-box updates
>> (NAAAAAAAAATs) and doesn't require OS vendors / kernel developers /
>> distros to develop, package, and deliver an implementation.  That
>> pretty much means: plain TCP and plain UDP.  Not IPsec (sure, IPsec
>> can run over UDP, but apps can't all run their own IKEs), not SCTP
>> (unless it's over UDP), not TCP-AO or anything of the sort.
>>
>> Things like SCTP over UDP will do.
>
> By your set of requirements, *anything* over UDP would work fine.
> Including IPsec, TCP-AO, and anything else you can dream up.
>
> Thank you for moving the stack up one level, adding 20 bytes of header,

(I intended to make that 8, 28, or 48, depending on whether you embed 
another layer of IPv4 or IPv6 header too).

> and constraining the number of connections between two hosts (due to the
> limit of port numbers).
>
> Joe

From michawe@ifi.uio.no  Fri Oct 11 01:14:37 2013
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On 30. aug. 2013, at 18:50, Nico Williams wrote:

> - HTTP needs an XID so as to make it less dependent on octet stream =
transports.  HTTP chunks, so there's no need for sub-channels, as long =
as an XID can be used to dissociate reply order from request order.
>=20
> - SCTP and similar should be implemented in user-space -- the OS/NIC =
should know nothing more than how to classify packets for dispatch =
purposes.  UDP obviously facilitates this and has the benefit that =
middle boxes deal with it.
>=20
>=20
AFAIK a pretty well working version of SCTP in user-space exists, and =
this is also what's used in Firefox for the rtcweb data channel.

> - Multi-homing and name resolution should be abstracted from the app, =
even if they are handled within the app (in a library).
>=20
+1

> - The tricky things are a) security (SYN forgery type attacks), and b) =
congestion control.  We know how to do (a) with low latency sometimes =
(e.g., see TCP Fast Open).  (b) is trickier; read with care (has the IAB =
said anything about this?).
>=20
>=20
RFC 3714 is a congestion control related RFC from the IAB. There's also =
more RFCs with general recommendations, and a RFC 5783 gives an overview =
of all of them (up to the date of its publication).

> - User/service authentication and session crypto are orthogonal to the =
above: TLS and similar can be adapted to unreliable, unordered datagram =
delivery, and they have been (DTLS).  This is not to say that we have =
nothing to do in this space, just that it's mostly orthogonal to the =
transport issues/wish-list.
>=20
> Nico
> --
>=20

Cheers,
Michael


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On 30. aug. 2013, at 18:50, Nico Williams =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><p dir=3D"ltr">- HTTP needs an XID so as to make it less =
dependent on octet stream transports.&nbsp; HTTP chunks, so there's no =
need for sub-channels, as long as an XID can be used to dissociate reply =
order from request order.</p><p dir=3D"ltr">- SCTP and similar should be =
implemented in user-space -- the OS/NIC should know nothing more than =
how to classify packets for dispatch purposes.&nbsp; UDP obviously =
facilitates this and has the benefit that middle boxes deal with =
it.</p><div><br></div></blockquote>AFAIK a pretty well working version =
of SCTP in user-space exists, and this is also what's used in Firefox =
for the rtcweb data channel.</div><div><br><blockquote type=3D"cite"><p =
dir=3D"ltr">- Multi-homing and name resolution should be abstracted from =
the app, even if they are handled within the app (in a =
library).</p></blockquote>+1</div><div><br></div><div><blockquote =
type=3D"cite"><p dir=3D"ltr">- The tricky things are a) security (SYN =
forgery type attacks), and b) congestion control.&nbsp; We know how to =
do (a) with low latency sometimes (e.g., see TCP Fast Open).&nbsp; (b) =
is trickier; read with care (has the IAB said anything about =
this?).</p><div><br></div></blockquote>RFC 3714 is a congestion control =
related RFC from the IAB. There's also more RFCs with general =
recommendations, and a RFC 5783 gives an overview of all of them (up to =
the date of its publication).</div><div><br><blockquote type=3D"cite"><p =
dir=3D"ltr">- User/service authentication and session crypto are =
orthogonal to the above: TLS and similar can be adapted to unreliable, =
unordered datagram delivery, and they have been (DTLS).&nbsp; This is =
not to say that we have nothing to do in this space, just that it's =
mostly orthogonal to the transport issues/wish-list.</p><p =
dir=3D"ltr">Nico<br>
-- </p>
=
</blockquote></div><br><div>Cheers,</div><div>Michael</div><div><br></div>=
</body></html>=

--Apple-Mail=_87E28245-9E70-4F58-9ABD-C1FD39F20AEE--

From nico@cryptonector.com  Fri Oct 11 09:25:02 2013
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On Thu, Oct 10, 2013 at 12:28 PM, Joe Touch <touch@isi.edu> wrote:
> On 9/4/2013 8:14 PM, Nico Williams wrote:
>> Things like SCTP over UDP will do.
>
> By your set of requirements, *anything* over UDP would work fine. Including
> IPsec, TCP-AO, and anything else you can dream up.

Well, not TCP-AO.   But, yes.

> Thank you for moving the stack up one level, adding 20 bytes of header, and
> constraining the number of connections between two hosts (due to the limit
> of port numbers).

As if it were my fault.  It's not.  Or as if blame for this can be
properly allocated.

And anyways, no, we could run lots of [end-to-end] things over UDP in
such a way that the port numbers from the UDP header need not be
repeated at higher layers.  But being able to use a single port for
lots of communications streams is useful to deal with the second
problem you note.

Nico
--

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Cc: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, HTTP Working Group <ietf-http-wg@w3.org>, tsvwg <tsvwg@ietf.org>
Subject: Re: [tsvwg] The List (of application-layer desired features)
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On 10/11/2013 9:24 AM, Nico Williams wrote:
> On Thu, Oct 10, 2013 at 12:28 PM, Joe Touch <touch@isi.edu> wrote:
>> On 9/4/2013 8:14 PM, Nico Williams wrote:
>>> Things like SCTP over UDP will do.
>>
>> By your set of requirements, *anything* over UDP would work fine. Including
>> IPsec, TCP-AO, and anything else you can dream up.
>
> Well, not TCP-AO.   But, yes.

TCP-AO would work fine, presuming it's TCP over UDP, or TCP over IP over 
UDP.

>> Thank you for moving the stack up one level, adding 20 bytes of header, and
>> constraining the number of connections between two hosts (due to the limit
>> of port numbers).
>
> As if it were my fault.  It's not.  Or as if blame for this can be
> properly allocated.

"you" is also plural, and by encouraging this sort of approach, you are 
a member of the group.

> And anyways, no, we could run lots of [end-to-end] things over UDP in
> such a way that the port numbers from the UDP header need not be
> repeated at higher layers.

You can reinvent the entire IP stack over a single UDP port. We've been 
around this block before with HTTP.

> But being able to use a single port for
> lots of communications streams is useful to deal with the second
> problem you note.

Useful in what sense? More fragile when dealing with changes in routing 
that alter which NATs are traversed? Providing the ability to reinvent 
the entire IP stack above UDP in a way that won't (can't) interact 
correctly with the native IP stack?

Joe


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Subject: [tsvwg] [Extended Deadline] Call for Papers of the PAMS 2014
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We apologize if you receive multiple copies of this CFP.


CALL FOR PAPERS

The 4th International Workshop on Protocols and Applications with Multi=
-Homing=20
Support (PAMS 2014)
In conjunction with the 28th IEEE International Conference on Advanced=20=

Information Networking and Applications (AINA 2014)
Victoria, BC/Canada, May 13-16, 2014
http://simula.no/pams2014

The intent of our workshop is to bring together people from research an=
d
industry, in order to provide a discussion forum for state-of-the-art t=
opics
related to multi-homing on Network, Transport, Session and Application =
Layers.
The PAMS workshop will include full-paper sessions as well as a poster =
session
(with short presentations) to introduce preliminary ideas as well as wo=
rk in
progress.

Workshops proceedings will be published by the IEEE CS Conference Publi=
shing
Service and will be included in the Digital Library.

The main topics to be addressed include (but are not limited to):

    * Network Resilience by Multi-Homing
    * Network Architectures for Multi-Homed Systems
    * Performance Evaluation of Multi-Homed Systems
    * Deployment of Multi-Homed Protocols in Existing Networks
      with Middleboxes
    * Design and Implementation of Multi-Homed Systems
    * Testbeds for Multi-Homed Systems
    * Load Sharing and Load Balancing for Multi-Homed Systems
    * Mobility for Multi-Homed Systems
    * Protocols with Multi-Homing Support
    * Congestion and Flow Control of Multi-Homed Systems
    * Quality of Service for Multi-Homed Systems
    * Multi-Homing for Interconnection Networks
    * Security of Multi-Homed Systems
    * Application Deployment and Support for Legacy Applications
    * Cross-Layer Optimisation for Multi-Homed Systems
    * Multi-Homing in the Context of the Future Internet

GENERAL CHAIRS:

    * Thomas Dreibholz, Simula Research Laboratory, Norway
    * Hakim Adhari, University of Duisburg-Essen, Germany

PROGRAM CHAIRS:

    * Martin Becke, University of Duisburg-Essen, Germany
    * Thomas Dreibholz, Simula Research Laboratory, Norway

PUBLICITY CHAIRS:

    * Glenford Mapp, Middlesex University, United Kingdom
    * Xing Zhou, Hainan University, China

PROGRAM COMMITTEE

    * Hakim Adhari, University of Duisburg-Essen, Germany
    * =C4=B0lknur Ayd=C4=B1n, Farmingdale State College, U.S.A.
    * =C5=81ukasz Budzisz, Technische Universit=C3=A4t Berlin, Germany
    * Nicola Altan, ista International, Germany
    * Paul Amer, University of Delaware, U.S.A.
    * Anna Brunstr=C3=B6m, Karlstad University, Sweden
    * Ahmed Elmokashfi, Simula Research Laboratory, Norway
    * Kristian Evensen, Simula Research Laboratory, Norway
    * Simone Ferlin, Simula Research Laboratory, Norway
    * Ernst Gunnar Gran, Simula Research Laboratory, Norway
    * Michio Honda, Keio University, Japan
    * Seok Joo Koh, Kyungpook National University, South Korea
    * Amund Kvalbein, Simula Research Laboratory, Norway
    * Glenford Mapp, Middlesex University, United Kingdom
    * Michael Menth, University of T=C3=BCbingen, Germany
    * Preethi Natarajan, Cisco, U.S.A.
    * Brad Penoff, Google, U.S.A.
    * Esbold Unurkhaan, Mongolian University of Science and Technology,=
=20
Mongolia
    * Alan Wagner, University of British Columbia, Canada
    * Thomas Volkert, Ilmenau University of Technology, Germany
    * Michael Welzl, University of Oslo, Norway
    * Xing Zhou, Hainan University, China


IMPORTANT DATES

    * Paper Submission Deadline:       November 1, 2013 (extended)
    * Author Notification:             December 20, 2013
    * Author Registration:             January 28, 2014
    * Camera-Ready Paper Submission:   February 11, 2014


CONTACT

For further information, please contact:

    * Thomas Dreibholz (dreibh@simula.no)
    * Hakim Adhari (hakim.adhari@uni-due.de)


WEBSITE

http://simula.no/pams2014

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From gorry@erg.abdn.ac.uk  Wed Oct 16 10:58:43 2013
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Subject: [tsvwg] IETF TSVWG at Vancouver - Agenda Requests
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If you still would like to present at the planned TSVWG meeting in 
Vancouver, please let the chairs know asap:

* Presentation title / draft
* Presenter
* Presentation time

- Last meeting we again received requests for a large number of drafts, 
so this time we plan again to focus must time on drafts that have seen 
recent discussion on the list (hint: If it has not been discussed on the 
list, then  it probably isn't ready for tsvwg) and those drafts that 
have changed significantly since the last meeting.

- Time for WG drafts will be allocated in precedence of other presentations.

Best wishes,

Gorry, James, and David



---  Working Agenda below ---

The working agenda is below, slots marked ??? have not been assigned a 
presentation and may be removed if the need for time is not confirmed.

--

TSVWG Agenda
    IETF-87, Berlin
    WG Chairs:
    Gorry Fairhurst <gorry@erg.abdn.ac.uk>
    James Polk <jmpolk@cisco.com>
    David Black <david.black@emc.com>

Please note there are 2 sessions and some drafts may need to be moved
between sessions to accommodate time keeping.


TSVWG Session 1 0900-1130 MONDAY, July 29, 2013
Note Taker: ???

1) Chairs  (10)
    NOTE WELL

    Document Status and Accomplishments

    Charter & Milestones Review

2) Chairs - Drafts Under review for publication (5)

    2.1) Bob/Jukka - Byte and Packet Congestion Notification (0)
    draft-ietf-tsvwg-byte-pkt-congest
    - On agenda of 2013-10-24 IESG telechat

    2.2) Randy and Michael - SCTP SACK Immediately (0)
    draft-ietf-tsvwg-sctp-sack-immediately
    - TRFC Ed Queue


3) Working Group Drafts

    3.1) Joe Touch (proxy Gorry Fairhurst) - Recommendations for 
Transport Port Uses  (???)
    draft-ietf-tsvwg-port-use

    3.2) Randy and Michael - SCTP NAT Support (??)
    draft-ietf-tsvwg-natsupp
    - Thought ready for WGLC.

    3.3) James Polk - RSVP Application-ID Profiles (???)
    draft-ietf-tsvwg-rsvp-app-id-vv-profiles
    - Thought ready for WGLC.

    3.4)Anurag Bhargava - RSVP for PCN (???)
    draft-ietf-tsvwg-rsvp-pcn
    - Awaiting author update

    3.5) Randy and Michael - DTLS Encap of SCTP for RTCWEB (???)
    draft-ietf-tsvwg-sctp-dtls-encaps

    3.6) SCTP Failover (???)
    draft-ietf-tsvwg-sctp-failover

4) Other Drafts

    4.1) Edward Crabbe - Generic UDP Encapsulation for IP Tunneling (???)
    draft-yong-tsvwg-gre-in-udp-encap
    - Proposed as a work item

==================================================================




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



From Ruediger.Geib@telekom.de  Fri Oct 18 01:28:21 2013
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Dear TSVers

I've updated my draft on terminology and on treatment of Network Control
traffic at carrier interconnection interfaces.

Regards,

Ruediger

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
Sent: Friday, October 18, 2013 10:25 AM
To: Geib, R=FCdiger; Geib, R=FCdiger
Subject: New Version Notification for draft-geib-tsvwg-diffserv-intercon-04=
.txt


A new version of I-D, draft-geib-tsvwg-diffserv-intercon-04.txt
has been successfully submitted by Ruediger Geib and posted to the
IETF repository.

Filename:        draft-geib-tsvwg-diffserv-intercon
Revision:        04
Title:           DiffServ interconnection classes and practice
Creation date:   2013-10-18
Group:           Individual Submission
Number of pages: 15
URL:             http://www.ietf.org/internet-drafts/draft-geib-tsvwg-diffs=
erv-intercon-04.txt
Status:          http://datatracker.ietf.org/doc/draft-geib-tsvwg-diffserv-=
intercon
Htmlized:        http://tools.ietf.org/html/draft-geib-tsvwg-diffserv-inter=
con-04
Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-geib-tsvwg-diffse=
rv-intercon-04

Abstract:
   This document proposes a limited set of interconnection QoS PHBs and
   PHB groups.  It further introduces some DiffServ deployment aspects.
   The proposals made here should be integrated into a revised version
   of RFC5127.




Please note that it may take a couple of minutes from the time of submissio=
n
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat


From paul.hoffman@vpnc.org  Sat Oct 19 07:20:55 2013
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Subject: [tsvwg] Fwd: [IPsec] Beginning of WG Last Call on draft-ietf-ipsecme-ikev2-fragmentation
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Greetings. The IPsecME WG has started WG Last Call on a draft that deals =
with IKEv2 fragmentation. The abstract says:

   This document describes the way to avoid IP fragmentation of large
   IKEv2 messages.  This allows IKEv2 messages to traverse network
   devices that don't allow IP fragments to pass through.

The WG welcomes document reviews from anyone on this list, particularly =
those who have an interest in fragmentation.

Begin forwarded message:

> From: Paul Hoffman <paul.hoffman@vpnc.org>
> Subject: [IPsec] Beginning of WG Last Call on =
draft-ietf-ipsecme-ikev2-fragmentation
> Date: October 18, 2013 6:49:43 PM PDT
> To: "ipsec@ietf.org WG" <ipsec@ietf.org>
>=20
> Greetings again. The WG's draft on IKEv2 fragmentation has gotten a =
fair amount of review in the past few months, and is thus ready for a =
two-week WG Last Call (ending November 1). The draft is at:
>=20
> http://tools.ietf.org/html/draft-ietf-ipsecme-ikev2-fragmentation
>=20
> Please read this draft and send any comments to the WG mailing list, =
even if the comments are "I see no problems". Comments such as "I do not =
understand this part" or "this part could be explained better in this =
way" are particularly useful at this point.
>=20
> If you have an IKEv2 implementation, or are using someone else's IKEv2 =
implementation in your products, you really should be reviewing and =
commenting on this draft. This will have a significant impact on IKEv2 =
implementations in the future, and having anything wrong or even unclear =
in the draft will make everyone's life worse. Please do a review and =
send any comments to the list.
>=20
> --Paul Hoffman


From internet-drafts@ietf.org  Sun Oct 20 10:51:07 2013
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

	Title           : Generic Aggregation of Resource ReSerVation Protocol (RS=
VP) for IPv4 And IPv6 Reservations over PCN domains
	Author(s)       : Georgios Karagiannis
                          Anurag Bhargava
	Filename        : draft-ietf-tsvwg-rsvp-pcn-07.txt
	Pages           : 31
	Date            : 2013-10-20

Abstract:
   This document specifies extensions to Generic Aggregated RSVP
   [RFC4860] for support of the PCN Controlled Load (CL) and Single
   Marking (SM) edge behaviors over a Diffserv cloud using Pre-
   Congestion Notification.





The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-07


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From karagian@cs.utwente.nl  Sun Oct 20 11:02:26 2013
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--_000_FF1A9612A94D5C4A81ED7DE1039AB80F4F3F104AEXMBX23adutwent_
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Hi all,



Please note that a new version of the draft-ietf-tsvwg-rsvp-pcn has been su=
bmitted!

http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-rsvp-pcn-07.txt

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-07


We have worked out the comments that have been posted by Francois during th=
e WGLC on this draft.



In particular, we have addressed the following comments:



o) clarify the split of functionality between PCN-ingress-node (Aggregator)=
 and PCN-egress-node (Deaggregator)
Note that this was a signifficant change!

o) *** section 1.1:
expand the PBAC acronym in Introduction


o) ***section 1.2:
OLD:
"This is also the reason of choosing Approach (1)"
NEW:
"This is the reason for choosing Approach (1)"


o) *** section 6:
expand the text to indicate to IANA that they need to allocate all the C-Ty=
pes defined in section 4.1 and  indicate the policy for allocating addition=
al C-Types. Doing that we also updated Section 4.1.


o) *** Add an example message flow in an Appendix, similar to Appendix A of=
 RFC4860. This considerably helps getting the general idea in a short time.

Best regards,

Georgios



________________________________
Van: tsvwg-bounces@ietf.org [tsvwg-bounces@ietf.org] namens internet-drafts=
@ietf.org [internet-drafts@ietf.org]
Verzonden: zondag 20 oktober 2013 19:51
To: i-d-announce@ietf.org
Cc: tsvwg@ietf.org
Onderwerp: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-07.txt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

        Title           : Generic Aggregation of Resource ReSerVation Proto=
col (RSVP) for IPv4 And IPv6 Reservations over PCN domains
        Author(s)       : Georgios Karagiannis
                          Anurag Bhargava
        Filename        : draft-ietf-tsvwg-rsvp-pcn-07.txt
        Pages           : 31
        Date            : 2013-10-20

Abstract:
   This document specifies extensions to Generic Aggregated RSVP
   [RFC4860] for support of the PCN Controlled Load (CL) and Single
   Marking (SM) edge behaviors over a Diffserv cloud using Pre-
   Congestion Notification.





The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-07


Please note that it may take a couple of minutes from the time of submissio=
n
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


--_000_FF1A9612A94D5C4A81ED7DE1039AB80F4F3F104AEXMBX23adutwent_
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<div style=3D"FONT-FAMILY: Tahoma; DIRECTION: ltr; COLOR: #000000; FONT-SIZ=
E: 10pt">
<p>Hi all,</p>
<p>&nbsp;</p>
<p>Please note that a new version of the draft-ietf-tsvwg-rsvp-pcn has been=
 submitted!</p>
<p><a href=3D"http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-rsvp-pcn=
-07.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-t=
svwg-rsvp-pcn-07.txt</a><br>
</p>
<p>A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-07"=
 target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp=
-pcn-07</a><br>
<br>
<br>
We have worked out the comments that have been posted by Francois during th=
e WGLC on this draft.</p>
<p>&nbsp;</p>
<p>In particular, we have addressed the following comments:</p>
<p>&nbsp;</p>
<p>o) clarify the split of functionality between&nbsp;PCN-ingress-node (Agg=
regator)&nbsp;and&nbsp;PCN-egress-node (Deaggregator)&nbsp;&nbsp;<br>
Note that this was a signifficant change!</p>
<p><br>
o) *** section 1.1:<br>
expand the PBAC acronym in Introduction<br>
<br>
<br>
o) ***section 1.2:<br>
OLD:<br>
&quot;This is also the reason of choosing Approach (1)&quot;<br>
NEW:<br>
&quot;This is the reason for choosing Approach (1)&quot;<br>
<br>
<br>
o) *** section 6:<br>
expand the text to indicate to IANA that they need to allocate all the C-Ty=
pes defined in section 4.1 and &nbsp;indicate the policy for allocating add=
itional C-Types. Doing that we also updated Section 4.1.</p>
<p><br>
<br>
o) *** Add an example message flow in an Appendix, similar to Appendix A of=
 RFC4860. This considerably helps getting the general idea in a short time.=
<br>
<br>
Best regards,</p>
<p>Georgios</p>
<p>&nbsp;</p>
<div>
<hr tabindex=3D"-1">
<div id=3D"x_divRplyFwdMsg"><font color=3D"#000000" size=3D"2" face=3D"Taho=
ma"><b>Van:</b> tsvwg-bounces@ietf.org [tsvwg-bounces@ietf.org] namens inte=
rnet-drafts@ietf.org [internet-drafts@ietf.org]<br>
<b>Verzonden:</b> zondag 20 oktober 2013 19:51<br>
<b>To:</b> i-d-announce@ietf.org<br>
<b>Cc:</b> tsvwg@ietf.org<br>
<b>Onderwerp:</b> [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-07.txt<br>
</font><br>
</div>
<div></div>
</div>
<font size=3D"2"><span style=3D"FONT-SIZE: 10pt">
<div class=3D"PlainText"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
&nbsp;This draft is a work item of the Transport Area Working Group Working=
 Group of the IETF.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Generic Aggregation of Resource ReSerVa=
tion Protocol (RSVP) for IPv4 And IPv6 Reservations over PCN domains<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Author(s)&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp; : Georgios Karagiannis<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 Anurag Bhargava<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; : draft-ietf-tsvwg-rsvp-pcn-07.txt<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 31<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2013-10-20<br>
<br>
Abstract:<br>
&nbsp;&nbsp; This document specifies extensions to Generic Aggregated RSVP<=
br>
&nbsp;&nbsp; [RFC4860] for support of the PCN Controlled Load (CL) and Sing=
le<br>
&nbsp;&nbsp; Marking (SM) edge behaviors over a Diffserv cloud using Pre-<b=
r>
&nbsp;&nbsp; Congestion Notification.<br>
<br>
<br>
<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn" targ=
et=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn</a=
><br>
<br>
There's also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-07" target=
=3D"_blank">http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-07</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-07"=
 target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp=
-pcn-07</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at tools.ietf.org.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
</div>
</span></font></div>
</body>
</html>

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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-sctp-dtls-encaps-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

	Title           : DTLS Encapsulation of SCTP Packets
	Author(s)       : Michael Tuexen
                          Randall R. Stewart
                          Randell Jesup
                          Salvatore Loreto
	Filename        : draft-ietf-tsvwg-sctp-dtls-encaps-02.txt
	Pages           : 7
	Date            : 2013-10-20

Abstract:
   The Stream Control Transmission Protocol (SCTP) is a transport
   protocol originally defined to run on top of the network protocols
   IPv4 or IPv6.  This document specifies how SCTP can be used on top of
   the Datagram Transport Layer Security (DTLS) protocol.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-dtls-encaps

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-sctp-dtls-encaps-02


Please note that it may take a couple of minutes from the time of submission
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Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From eckelcu@cisco.com  Mon Oct 21 14:07:32 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "int-area@ietf.org" <int-area@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: New Version Notification for draft-eckert-intarea-flow-metadata-framework-02.txt
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Subject: [tsvwg] FW: New Version Notification for draft-eckert-intarea-flow-metadata-framework-02.txt
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This update of the flow metadata framework addresses feedback received
regarding:

1) the relationship of this proposed framework with existing heuristic
based mechanisms
2) clarification on signed/unsigned values
3) change of delay tolerance and loss tolerance attributes to be defined
in terms of a discrete and ordered set of values
4) addition of jitter tolerance attribute

We did not have time to adequately expand on the security considerations,
so it is being kept on the todo list for the next revision.

The previous version of the draft was presented in INTAREA during IETF 87.
A slot has been requested in TSVWG at IETF 88 to present "Flow Metadata
Signaling with RSVP"
(http://tools.ietf.org/id/draft-zamfir-tsvwg-flow-metadata-rsvp), which
applies this framework to RSVP.

Please have a look and and share your comments.

Cheers,
Charles


On 10/21/13 1:45 PM, "internet-drafts@ietf.org" <internet-drafts@ietf.org>
wrote:

>
>A new version of I-D, draft-eckert-intarea-flow-metadata-framework-02.txt
>has been successfully submitted by Toerless Eckert and posted to the
>IETF repository.
>
>Filename:	 draft-eckert-intarea-flow-metadata-framework
>Revision:	 02
>Title:		 A Framework for Signaling Flow Characteristics between
>Applications and the Network
>Creation date:	 2013-10-21
>Group:		 Individual Submission
>Number of pages: 18
>URL:            =20
>http://www.ietf.org/internet-drafts/draft-eckert-intarea-flow-metadata-fra
>mework-02.txt
>Status:         =20
>http://datatracker.ietf.org/doc/draft-eckert-intarea-flow-metadata-framewo
>rk
>Htmlized:       =20
>http://tools.ietf.org/html/draft-eckert-intarea-flow-metadata-framework-02
>Diff:           =20
>http://www.ietf.org/rfcdiff?url2=3Ddraft-eckert-intarea-flow-metadata-fram=
ew
>ork-02
>
>Abstract:
>   This document provides a framework for communicating information
>   elements (a.k.a. metadata) in a consistent manner between
>   applications and the network to provide better visibility of
>   application flows, thereby enabling differentiated treatment of those
>   flows.  These information elements can be conveyed using various
>   signaling protocols, including PCP, RSVP, and STUN.
>
>                 =20
>       =20
>
>
>Please note that it may take a couple of minutes from the time of
>submission
>until the htmlized version and diff are available at tools.ietf.org.
>
>The IETF Secretariat
>


From internet-drafts@ietf.org  Mon Oct 21 15:59:18 2013
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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-sctp-failover-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

	Title           : Quick Failover Algorithm in SCTP
	Author(s)       : Yoshifumi Nishida
                          Preethi Natarajan
                          Armando Caro
                          Paul D. Amer
	Filename        : draft-ietf-tsvwg-sctp-failover-02.txt
	Pages           : 17
	Date            : 2013-10-21

Abstract:
   One of the major advantages in SCTP is supporting multi-homing
   communication.  If a multi-homed end-point has redundant network
   connections, SCTP sessions can have a good chance to survive from
   network failures by migrating inactive network to active one.
   However, if we follow the SCTP standard, there can be significant
   delay for the network migration.  During this migration period, SCTP
   cannot transmit much data to the destination.  This issue drastically
   impairs the usability of SCTP in some situations.  This memo
   describes the issue of SCTP failover mechanism and discuss its
   solutions which require minimal modification to the current standard.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-failover

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-failover-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-sctp-failover-02


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From bob.briscoe@bt.com  Mon Oct 21 16:41:30 2013
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Date: Tue, 22 Oct 2013 00:40:39 +0100
To: Dave Taht <dave.taht@gmail.com>
From: Bob Briscoe <bob.briscoe@bt.com>
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Cc: "aqm@ietf.org" <aqm@ietf.org>, tsvwg IETF list <tsvwg@ietf.org>
Subject: [tsvwg] Immediate and Explicit Congestion Notification (was: Guidelines for Adding Congestion Notification to Protocols that Encapsulate IP)
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Dave,

Thx for cross-posting. It's quite possible there are aqm people not on=
 tsvwg.

1) Did you have views on the ecn-encap-guidelines I-D itself? Is it helpful?
I think it's fairly non-controversial.

2) In case I'm misunderstood, I don't see much=20
latency gain from ECN as currently defined=20
(RFC3168). I'm still banging on about ECN in=20
lower layers 'cos early tests show promise that=20
we may have found a way to make the ultra-low=20
queuing delay of data centre TCP incrementally=20
deployable on the public Internet

         We =3D Mirja Kuehlewind, David Wagner, Juan Manuel Reyes Espinosa=
 and I.

It can be deployed with most existing kit,=20
without any new AQM implementation needed. We use=20
WRED in an unusual configuration. The idea in a nutshell:
* For ECN-capable packets, shift the job of hiding bursts from network to=
 host:
    - ECN-capable packets map to a WRED policy with queue smoothing turned=
 off
    - so the network signals ECN with zero smoothing delay
    - then the transport can hide bursts of ECN=20
signals from itself (like DCTCP does)
    - the transport knows
      o that it's ECN-capable
      o whether it's TCP or RTP etc,
      o whether its in congestion avoidance or slow-start,
      o and it knows its RTT,
      o so it can know whether to respond immediately or to smooth the=
 signals,
      o and if so, over what time
    - then short RTT flows can smooth the signals=20
with only the delay of their /own/ RTT
      o so they can fill troughs and absorb peaks that longer RTT flows=
 cannot
    - a TCP only needs to smooth the signals if in congestion avoidance
      o in slow start, it can respond immediately, thus reducing overshoot

There's more to it - I've asked to present the whole story in Vancouver.



Bob

At 05:35 15/10/2013, Dave Taht wrote:
>It is my general assumption that membership in the aqm wg is a subset
>of tsvwg, but in case it isn't...
>
>
>---------- Forwarded message ----------
>From: Bob Briscoe <bob.briscoe@bt.com>
>Date: Tue, Sep 17, 2013 at 5:33 AM
>Subject: [tsvwg] Guidelines for Adding Congestion Notification to
>Protocols that Encapsulate IP
>To: tsvwg IETF list <tsvwg@ietf.org>
>Cc: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
>
>
>TSV folks,
>
>We've just posted an update to the guidelines for adding congestion
>notification to L2 and tunnelling protocols, e.g. Ethernet, L2TP, GRE,
>PPTP, GTP etc. Especially of current interest in mobile and separately
>in data centres.
>
>Pls read/review and send any comments to the list.
><http://tools.ietf.org/html/draft-briscoe-tsvwg-ecn-encap-guidelines>
>
>It's currently still an individual draft; intended status: BCP.
>We're looking for support for adoption as a item of work in tsvwg.
>
>We've dealt with most of the outstanding issues from previous reviews,
>esp. those from Gorry (see summary of diffs below).
>
>--------------------------------
>Summary of Diffs from -02 to -03
>
>       *  Scope section:
>
>          +  Added dependence on correct propagation of traffic class
>             information
>
>          +  For the feed-backward mode, deemed multicast and anycast out
>             of scope
>
>       *  Ensured all guidelines referring to subnet technologies also
>          refer to tunnels and vice versa by adding applicability
>          sentences at the start of sections 4.1, 4.2, 4.3, 4.4, 4.6 and
>          5.
>
>       *  Added Security Considerations on ensuring congestion signal
>          fields are classed as immutable and on using end-to-end
>          congestion signal integrity technologies rather than hop-by-
>          hop.
>
>
>Bob
>
>
> > From: <internet-drafts@ietf.org>
> > To: John Kaippallimalil <john.kaippallimalil@huawei.com>,
> >         Bob Briscoe
> >         <bob.briscoe@bt.com>,
> >         "Bob J.Briscoe" <bob.briscoe@bt.com>,
> >         Pat Thaler
> >         <pthaler@broadcom.com>,
> >         Patricia Thaler <pthaler@broadcom.com>
> > Subject: New Version Notification for
> >         draft-briscoe-tsvwg-ecn-encap-guidelines-03.txt
> >
> >
> > A new version of I-D, draft-briscoe-tsvwg-ecn-encap-guidelines-03.txt
> > has been successfully submitted by Bob Briscoe and posted to the
> > IETF repository.
> >
> > Filename:       draft-briscoe-tsvwg-ecn-encap-guidelines
> > Revision:       03
> > Title:          Guidelines for Adding=20
> Congestion Notification to Protocols that Encapsulate IP
> > Creation date:  2013-09-17
> > Group:          Individual Submission
> > Number of pages: 29
> > URL:=20
>=
 http://www.ietf.org/internet-drafts/draft-briscoe-tsvwg-ecn-encap-guideline=
s-03.txt
> > Status:=20
> http://datatracker.ietf.org/doc/draft-briscoe-tsvwg-ecn-encap-guidelines
> > Htmlized:=20
> http://tools.ietf.org/html/draft-briscoe-tsvwg-ecn-encap-guidelines-03
> > Diff:=20
>=
 http://www.ietf.org/rfcdiff?url2=3Ddraft-briscoe-tsvwg-ecn-encap-guidelines=
-03
> >
> > Abstract:
> >    The purpose of this document is to guide the design of congestion
> >    notification in any lower layer or tunnelling protocol that
> >    encapsulates IP.  The aim is for explicit congestion signals to
> >    propagate consistently from lower layer protocols into IP.  Then the
> >    IP internetwork layer can act as a portability layer to carry
> >    congestion notification from non-IP-aware congested nodes up to the
> >    transport layer (L4).  Following these guidelines should assure
> >    interworking between new lower layer congestion notification
> >    mechanisms, whether specified by the IETF or other standards bodies.
> >
> >
> >
> >
> >
> > Please note that it may take a couple of=20
> minutes from the time of submission
> > until the htmlized version and diff are available at tools.ietf.org.
> >
> > The IETF Secretariat
>
>
>________________________________________________________________
>Bob Briscoe,                                                  BT
>
>
>--
>Dave T=E4ht
>
>Fixing bufferbloat with cerowrt:=20
>http://www.teklibre.com/cerowrt/subscribe.html
>_______________________________________________
>aqm mailing list
>aqm@ietf.org
>https://www.ietf.org/mailman/listinfo/aqm

________________________________________________________________
Bob Briscoe,                                                  BT=20


From touch@isi.edu  Mon Oct 21 17:06:52 2013
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From: Joe Touch <touch@isi.edu>
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Hi, all,

I've reviewed the following doc for TSVDIR:
	draft-ietf-ipsecme-ikev2-fragmentation-04

Although this is not intended as a complete TSVDIR review, I have 
checked for the typical issues.

Joe

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

draft-ietf-ipsecme-ikev2-fragmentation-04
	
This doc makes the case that IKEv2 should implement its own 
frag/reassembly mechanism, because some NATs don't pass IP fragments.

First, the issue with NATs and fragments should be made more clear, 
especially citing existing descriptions of this issue, e.g., RFC4787. 
Note that NATs which do not pass fragments violate RFQ-14 of that BCP 
RFC, and it might be useful to report this issue to the vendor.

Second, it is not clear why this issue is being handled inside IKE. 
Appendix A provides a design rationale, but it clearly trades 
computation and space overhead for local storage, and still permits DOS 
attacks (e.g., overloading the receiver with false fragments).

An equally viable solution (see ** below) would involve placing a single 
signed IKE message into a single IP packet, fragmenting that packet, and 
transmitting those fragments inside UDP datagrams. This approach could 
use the innermost IP encapsulation as the reassembly layer.

**: Although this approach is susceptible to fragmentation overload, so 
too is transmitting the original message over IP that is naturally 
fragmented at the network layer anyway - this introducing no new 
vulnerability.

This IP-encapsulated IKE message would need the same negotiation 
described in section 2.3, but would avoid the complexity of most of the 
rest of the document, relying instead on existing IP reassembly.

In conclusion, I don't see the need for the level of detail and 
mechanism proscribed, basically because it reinvents the wheel, but see 
no other substantial issues with it.

Additionally, regarding PMTUD, this should be done using PLMTUD (RFC 
4821), and should be described as such.

---



From touch@isi.edu  Mon Oct 21 17:26:50 2013
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To: Martin Stiemerling <martin.stiemerling@neclab.eu>, ipsec@ietf.org, draft-amjads-ipsecme-ikev2-data-channel@tools.ietf.org, "tsv-ads@tools.ietf.org" <tsv-ads@tools.ietf.org>, tsvwg <tsvwg@ietf.org>
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Subject: [tsvwg] TSVDIR-ish review of draft-amjads-ipsecme-ikev2-data-channel-00
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Hi, all,

I've reviewed the following doc for TSVDIR:
      draft-amjads-ipsecme-ikev2-data-channel-00

Although this is not intended as a complete TSVDIR review, I have
checked for the typical issues.

Joe

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

draft-amjads-ipsecme-ikev2-data-channel-00

This doc makes the case that IKEv2 can provide a secure data channel for
arbitrary communication, rather than being used (as designed) to
configure IPsec channels for that purpose.

This mechanism lacks congestion control, and so needs to be used only 
where its load is known to be a small fraction of capacity. In specific, 
IKE's window mechanism allows for increasing the window size but not 
decreasing it, as is needed to react to network congestion.

The acknowledged data transfer mode uses IKE's window mechanism, which 
is presumably set to a small value, and may result in very low 
throughput. Attempts to increase this window size to overcome this 
limitation can easily increase burstiness and network loss.

This mechanism includes its own fragmentation mechanism based on a 
pre-configiured MTU, where it should use an adaptive size based on 
PLMTUD (RFC4821). The mechanism described replicates that of IP, and so 
introduces no new issues. Fragment reassembly appears to rely on the IKE 
sequence number, and the relationship between the two should be more 
clear, especially on the reuse of the IKE sequence number and how that 
affects reassembly timeout.

---


From ford@isoc.org  Tue Oct 22 03:56:25 2013
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On 22 Oct 2013, at 00:40, Bob Briscoe <bob.briscoe@bt.com> wrote:

> There's more to it - I've asked to present the whole story in =
Vancouver.

Cool, looking forward to it. I'll prime the question pump now if I may:

Does the transport need to know that it is connected to a network that =
will signal ECN in this fashion?

Mat


From touch@isi.edu  Tue Oct 22 08:28:14 2013
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Cc: tsvwg <tsvwg@ietf.org>, tsv-ads@tools.ietf.org, ipsec@ietf.org, draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org, tsv-dir@ietf.org
Subject: Re: [tsvwg] [IPsec] TSVDIR-ish review ofdraft-ietf-ipsecme-ikev2-fragmentation-04
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Hi, Valery,

On 10/22/2013 5:50 AM, Valery Smyslov wrote:
> Hi Joe,
>
> thank you for your review.
> Please, see my comments inline.
>
>> Hi, all,
>>
>> I've reviewed the following doc for TSVDIR:
>> draft-ietf-ipsecme-ikev2-fragmentation-04
>>
>> Although this is not intended as a complete TSVDIR review, I have
>> checked for the typical issues.
>>
>> Joe
>>
>> -------------------------------------------------------------------
>>
>> draft-ietf-ipsecme-ikev2-fragmentation-04
>>
>> This doc makes the case that IKEv2 should implement its own
>> frag/reassembly mechanism, because some NATs don't pass IP fragments.
>>
>> First, the issue with NATs and fragments should be made more clear,
>> especially citing existing descriptions of this issue, e.g., RFC4787.
>> Note that NATs which do not pass fragments violate RFQ-14 of that BCP
>> RFC, and it might be useful to report this issue to the vendor.
>>
>> Second, it is not clear why this issue is being handled inside IKE.
>> Appendix A provides a design rationale, but it clearly trades
>> computation and space overhead for local storage, and still permits DOS
>
> I don't see a strict computation/memory tradeoff here. You must
> verify message validity in any case, whether it is done before
> reassempling (on each fragment), or after it is complete (including
> the case when you get garbage message due to false fragments).
> The amount of consumped CPU power will be roughly the same.

The cost to verify several individual messages is higher than the cost 
to verify a single message, especially when the individual messages are 
not exactly the blocksize of the algorithm used.

So in the case of no attack, the overhead of your proposed mechanism is 
higher. You can't drop an entire message when a single false fragment 
arrives, because there could be other valid fragments on the way. So 
either way you end up checking the same amount of data, but your 
computational overhead is higher.

> But the amount of memory consumption will be smaller if
> you are able to discard false fragments.

Yes, but you also need to keep some state about pending partial valid 
messages.

AFAICT, the memory difference is more likely to be negligible, but the 
processing difference high.

>> attacks (e.g., overloading the receiver with false fragments).
>
> This attack is always possible both with and without fragmentation
> (just overloading with false full messages) and IKE is designed to
> withstand it.
> Handling fragmentation in IKE doesn't make it more vulnerable to this
> attack.
> On the other hand, handling fragmentation outside IKE (on IP level)
> or using unauthenticated fragments in IKE, makes it susceptible to an
> attack,
> described in the rationale, i.e. when attacker spending virtually zero
> resources
> can stop IKE communication by infrequently injecting bad fragments,
> which results in "poisoning" of reassembling queue. Using authenticated
> fragments
> prevents this kind of attack.

Avoiding fragmentation poisoning attacks is not a motivation of this 
work. If you want to make that case, you need to explain why you think 
that is a significant issue and worth the effort to fix inside IKE.

Further, this method does not necessarily avoid IP fragmentation. 
Fragments are supposed to pass through a NAT (see my previous 
reference), and when they do your mechanism wouldn't adjust the IKE size 
downward.

>> An equally viable solution (see ** below) would involve placing a
>> single signed IKE message into a single IP packet, fragmenting that
>> packet, and transmitting those fragments inside UDP datagrams. This
>> approach could use the innermost IP encapsulation as the reassembly
>> layer.
>>
>> **: Although this approach is susceptible to fragmentation overload,
>> so too is transmitting the original message over IP that is naturally
>> fragmented at the network layer anyway - this introducing no new
>> vulnerability.
>
> But using IKE fragmentation in this case might eliminate this
> vulnerability.

It might, but again that's not your threat model and there are other 
vulnerabilities this method opens up (e.g., replay attacks have a higher 
CPU impact).

>> This IP-encapsulated IKE message would need the same negotiation
>> described in section 2.3, but would avoid the complexity of most of
>> the rest of the document, relying instead on existing IP reassembly.
>>
>> In conclusion, I don't see the need for the level of detail and
>> mechanism proscribed, basically because it reinvents the wheel, but
>> see no other substantial issues with it.
>>
>> Additionally, regarding PMTUD, this should be done using PLMTUD (RFC
>> 4821), and should be described as such.
>
> IMHO, PLMTUD doesn't suite well for IKE. First, IKE is UDP-based
> and RFC4821 has many restrictions when used with connectionless
> datagram protocols, some of which are often difficult to satisfy in
> environments,
> where IKE is used.

You are re-inventing PLMUTD. In places where you claim RFC4821 is too 
complex - those are the places where your mechanism is deficient IMO. 
E.g., positive verification that a message gets through as a 
confirmation of when you've achieved a reasonable size.

> Second, the goal of PLMTUD is to find _maximum_
> MTU size that is not fragmented by IP, in other words, to maximize
> throughput. For IKE the goal of PMTUD is to find _any_reasonable_
> MTU size that won't be fragmented by IP. Throughput is not important.

Agreed, but if that's the case, why not always send the smallest 
possible message size (68 bytes)? It's because the overhead is too high 
(additional headers, blocksize mismatch, and computational startup cost 
on each block). Further, most transports don't care about 1400 vs. 1500 
bytes; they're searching for a rough approximation of a 'large' packet, 
as are you IMO.

> Third, PMTU discovery in the draft is completely optional and one may
> use any other mechanisms (including PLMTUD, if it is available)
> to obtain MTU information.

The transport folks spent a lot of time on PLMTUD; there are a lot of 
lessons there that can and should be applied here. PLMTUD is already 
flexible enough to apply here directly. You can always reinvent the 
wheel, of course.

Joe

From brian.e.carpenter@gmail.com  Tue Oct 22 16:57:12 2013
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Subject: Re: [tsvwg] I-D Action: draft-geib-tsvwg-diffserv-intercon-04.txt
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Hi R=C3=BCdiger,

>    DSCP Precedence Prefix  The bits 0-2 of the DSCP (marked "x" in this=

>            generic DSCP field: xxxddd) are called the DSCP Precedence
>            Prefix [RFC2474] in the following.  By ignoring the value of=

>            bits 3-6 ( d stands for don' care), a simple aggregation of
>            PHBs differed by DSCP is possible in IP and MPLS backbones,
>            but also if Ethernet transport is applied. =20

Thankyou for providing explicit terminology for this. I am still worried
that by reviving the word "precedence" we would seriously damage outsider=
s'
understanding of differentiated services. The specific reason we got rid
of "precedence" and the notion of simple priority (except for implementin=
g
the legacy code points) was to underline that precedence doesn't provide
quality of service; if anything, it provides quality of disservice.

Couldn't you call those three bits "DSCP Aggregation Prefix"? Then they
would refer to 8 traffic classes, and only in a very primitive
implementation would they be mapped to priority levels.

    Brian


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From svanru@gmail.com  Tue Oct 22 05:51:05 2013
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From: "Valery Smyslov" <svanru@gmail.com>
To: "Joe Touch" <touch@isi.edu>, "Martin Stiemerling" <martin.stiemerling@neclab.eu>, <ipsec@ietf.org>, <draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org>, <tsv-ads@tools.ietf.org>, "tsvwg" <tsvwg@ietf.org>
References: <51FA6A6C.8090803@gmail.com> <523CB65D.1090904@neclab.eu><523CB9CB.7060507@isi.edu> <5265C19B.30108@isi.edu>
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish review ofdraft-ietf-ipsecme-ikev2-fragmentation-04
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Hi Joe,

thank you for your review.
Please, see my comments inline.

> Hi, all,
>
> I've reviewed the following doc for TSVDIR:
> draft-ietf-ipsecme-ikev2-fragmentation-04
>
> Although this is not intended as a complete TSVDIR review, I have checked 
> for the typical issues.
>
> Joe
>
> -------------------------------------------------------------------
>
> draft-ietf-ipsecme-ikev2-fragmentation-04
>
> This doc makes the case that IKEv2 should implement its own 
> frag/reassembly mechanism, because some NATs don't pass IP fragments.
>
> First, the issue with NATs and fragments should be made more clear, 
> especially citing existing descriptions of this issue, e.g., RFC4787. Note 
> that NATs which do not pass fragments violate RFQ-14 of that BCP RFC, and 
> it might be useful to report this issue to the vendor.
>
> Second, it is not clear why this issue is being handled inside IKE. 
> Appendix A provides a design rationale, but it clearly trades computation 
> and space overhead for local storage, and still permits DOS

I don't see a strict computation/memory tradeoff here. You must
verify message validity in any case, whether it is done before
reassempling (on each fragment), or after it is complete (including
the case when you get garbage message due to false fragments).
The amount of consumped CPU power will be roughly the same.
But the amount of memory consumption will be smaller if
you are able to discard false fragments.

> attacks (e.g., overloading the receiver with false fragments).

This attack is always possible both with and without fragmentation
(just overloading with false full messages) and IKE is designed to withstand 
it.
Handling fragmentation in IKE doesn't make it more vulnerable to this 
attack.
On the other hand, handling fragmentation outside IKE (on IP level)
or using unauthenticated fragments in IKE, makes it susceptible to an 
attack,
described in the rationale, i.e. when attacker spending virtually zero 
resources
can stop IKE communication by infrequently injecting bad fragments,
which results in "poisoning" of reassembling queue. Using authenticated 
fragments
prevents this kind of attack.

> An equally viable solution (see ** below) would involve placing a single 
> signed IKE message into a single IP packet, fragmenting that packet, and 
> transmitting those fragments inside UDP datagrams. This approach could use 
> the innermost IP encapsulation as the reassembly layer.
>
> **: Although this approach is susceptible to fragmentation overload, so 
> too is transmitting the original message over IP that is naturally 
> fragmented at the network layer anyway - this introducing no new 
> vulnerability.

But using IKE fragmentation in this case might eliminate this vulnerability.

> This IP-encapsulated IKE message would need the same negotiation described 
> in section 2.3, but would avoid the complexity of most of the rest of the 
> document, relying instead on existing IP reassembly.
>
> In conclusion, I don't see the need for the level of detail and mechanism 
> proscribed, basically because it reinvents the wheel, but see no other 
> substantial issues with it.
>
> Additionally, regarding PMTUD, this should be done using PLMTUD (RFC 
> 4821), and should be described as such.

IMHO, PLMTUD doesn't suite well for IKE. First, IKE is UDP-based
and RFC4821 has many restrictions when used with connectionless
datagram protocols, some of which are often difficult to satisfy in 
environments,
where IKE is used. Second, the goal of PLMTUD is to find _maximum_
MTU size that is not fragmented by IP, in other words, to maximize
throughput. For IKE the goal of PMTUD is to find _any_reasonable_
MTU size that won't be fragmented by IP. Throughput is not important.
Third, PMTU discovery in the draft is completely optional and one may
use any other mechanisms (including PLMTUD, if it is available)
to obtain MTU information.

Thanks,
Valery Smyslov.



From paul@cypherpunks.ca  Tue Oct 22 08:35:49 2013
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish review ofdraft-ietf-ipsecme-ikev2-fragmentation-04
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On Tue, 22 Oct 2013, Valery Smyslov wrote:

>> attacks (e.g., overloading the receiver with false fragments).
>
> This attack is always possible both with and without fragmentation
> (just overloading with false full messages) and IKE is designed to withstand 
> it.

Partially. For unfragmented IKE, the cookies/SPI mitigate this if
you give preference to IKE messages further in the processing queue.
With this draft, I'm forced to do more crypto on IKE packets before I
can decide if these need prioritization or not (with the exception of
the first packet, which cannot get fragmented anyway)

> Handling fragmentation in IKE doesn't make it more vulnerable to this attack.

Handling encrypted fragments does, because you will spend more time
doing the crypto. Although this still happens after the cookie/spi
verification. You are doing an additional round of crypto on each fragment
in addition to the regular IKE crypto you need to perform on the assembled
packet. Fragments could be made artificially long and the IKE packet
artificial big, resulting in much more crypto. Now I think this might
not matter much, because the attacker needs to be local and they will
only be local to a remote endpoint which has enough CPU, not the ipsec
core gateway that actually might run out of CPU power when under attack.

> On the other hand, handling fragmentation outside IKE (on IP level)
> or using unauthenticated fragments in IKE, makes it susceptible to an attack,
> described in the rationale, i.e. when attacker spending virtually zero 
> resources
> can stop IKE communication by infrequently injecting bad fragments,
> which results in "poisoning" of reassembling queue.

But the attack is theoretical and of no practical value.

1) on-path attacker (remote)

They can just DOS you regardless by withholding packets.

2) off-path attacker (remote)

These attackers don't know the cookies/spi's so they can't really sent
a forged fragment you would use for assembly - protected with crypto or
not. It is also impossible for them to time their attack so that one of
their fragments intersect with one of your fragments so they would need
to start sending fragments before the endpoint does but using mostly
wrong cookies/spis and would swamp an endpoint with easilly detectable
forged packets.

3) shared medium like wifi hotspot (local)

These are the only attackers with enough knowledge to sent sufficiently
advanced bad fragments interfering with incoming fragments. They can
already do all the damage they want without resorting to IKE fragment
spoofing.  They can arp spoof the gateway, sent a Wifi de-association
request, sent enough garbage to fill the air waves, and interfere with
the IKE in such a way that you think the remote peer does not support
fragmentation at all. So this draft protects against a local attacker
that wants to masquerade as a network failure and only prevent you from
doing IKE, hoping you will fall back to plaintext traffic. Software
could notify the user that they seem to be under attack, which IMHO
would be more valuable to the user than quiet resilience against the
attack.

> Using authenticated fragments prevents this kind of attack.

Yes, it prevents one of the low hanging fruits. Which I don't think is
worth the complexity of this draft compared to the existing
fragmentation code out in the wild.

> But using IKE fragmentation in this case might eliminate this vulnerability.

At the price of adding more complexity to IKE. With the attacker not
stopped from their goal.

>> Additionally, regarding PMTUD, this should be done using PLMTUD (RFC 4821), 
>> and should be described as such.
>
> IMHO, PLMTUD doesn't suite well for IKE. First, IKE is UDP-based
> and RFC4821 has many restrictions when used with connectionless
> datagram protocols, some of which are often difficult to satisfy in 
> environments,
> where IKE is used. Second, the goal of PLMTUD is to find _maximum_
> MTU size that is not fragmented by IP, in other words, to maximize
> throughput. For IKE the goal of PMTUD is to find _any_reasonable_
> MTU size that won't be fragmented by IP. Throughput is not important.
> Third, PMTU discovery in the draft is completely optional and one may
> use any other mechanisms (including PLMTUD, if it is available)
> to obtain MTU information.

I agree. There is going to be like 2 to 4 fragments. Size does not
matter. It's better to cut down on latency/RTT by using an MTU size
that is for sure to work on the first go.

But that's also why this complexity is not worth it. Fragmentation does
only mean going from 1 IKE packet to 2-4 IKE packets. The timeframe
to mess with these are in the 5ms range. If an attacker has that
accuracy against you, you've lost already regardless. I doubt the
attacker can even pull it off on wifi.

The original IKE fragmentation is really nice and simple. Copy part of
the original IKE header, add a single payload with fragment payload
type and fragment number/max, sent fragments in a burst.

This draft looks like it is fixing a security issue, while it really
should be just about sending smaller packets as similar as possible
to the original single packet. Messing with one IKE packet versus 2-4
IKE packets is not a real problem that I think needs fixing.

Paul

From svan@elvis.ru  Tue Oct 22 23:26:28 2013
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From: Valery Smyslov <svan@elvis.ru>
To: Joe Touch <touch@isi.edu>
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish review ofdraft-ietf-ipsecme-ikev2-fragmentation-04
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>>> This doc makes the case that IKEv2 should implement its own
>>> frag/reassembly mechanism, because some NATs don't pass IP fragments.
>>>
>>> First, the issue with NATs and fragments should be made more clear,
>>> especially citing existing descriptions of this issue, e.g., RFC4787.
>>> Note that NATs which do not pass fragments violate RFQ-14 of that BCP
>>> RFC, and it might be useful to report this issue to the vendor.
>>>
>>> Second, it is not clear why this issue is being handled inside IKE.
>>> Appendix A provides a design rationale, but it clearly trades
>>> computation and space overhead for local storage, and still permits DOS
>>
>> I don't see a strict computation/memory tradeoff here. You must
>> verify message validity in any case, whether it is done before
>> reassempling (on each fragment), or after it is complete (including
>> the case when you get garbage message due to false fragments).
>> The amount of consumped CPU power will be roughly the same.
>
> The cost to verify several individual messages is higher than the cost to 
> verify a single message, especially when the individual messages are not 
> exactly the blocksize of the algorithm used.

Right. That's why I wrote "_roughly_ the same". Of course there is some
overhead, not only due to blocksize, but mainly due to the inclusion of IKE 
Header
into ICV calculations for each fragment message.

> So in the case of no attack, the overhead of your proposed mechanism is 
> higher. You can't drop an entire message when a single false fragment 
> arrives, because there could be other valid fragments on the way. So 
> either way you end up checking the same amount of data, but your 
> computational overhead is higher.

Yes, it is higher, but the amount of the overhead could be estimated
about 10%. If some system cannot bear this overhead, I think it is broken.

>> But the amount of memory consumption will be smaller if
>> you are able to discard false fragments.
>
> Yes, but you also need to keep some state about pending partial valid 
> messages.

Right. But only after fragment is validated.

> AFAICT, the memory difference is more likely to be negligible, but the 
> processing difference high.

No. In solution you have proposed (relying on IP fragmentation
inside UDP datagrams) the attacker could send you millions
of packets, all with different IP Fragmentation IDs in inner packet. As you
has no way to know which Fragmentation ID is valid (from real peer)
you will need to try to reassemble all of them, consuming
as much memory, as attacker wishes. If attacker is smart enough,
s(he) could easy make so, that all the false fragments will eventually
be reassembled by your system (into garbage), and you will also consume
the CPU power trying to validate them. In this case you will consume
roughly the same CPU power as with authenticated fragments,
but the amount of your memory consumption will be under attacker's control.

>>> attacks (e.g., overloading the receiver with false fragments).
>>
>> This attack is always possible both with and without fragmentation
>> (just overloading with false full messages) and IKE is designed to
>> withstand it.
>> Handling fragmentation in IKE doesn't make it more vulnerable to this
>> attack.
>> On the other hand, handling fragmentation outside IKE (on IP level)
>> or using unauthenticated fragments in IKE, makes it susceptible to an
>> attack,
>> described in the rationale, i.e. when attacker spending virtually zero
>> resources
>> can stop IKE communication by infrequently injecting bad fragments,
>> which results in "poisoning" of reassembling queue. Using authenticated
>> fragments
>> prevents this kind of attack.
>
> Avoiding fragmentation poisoning attacks is not a motivation of this work. 
> If you want to make that case, you need to explain why you think that is a 
> significant issue and worth the effort to fix inside IKE.

Actually, Security Consideration Section of RFC5996 has already
stated that IP fragmentation is a significant issue for IKE:

   If the messages of IKEv2 are long enough that IP-level fragmentation
   is necessary, it is possible that attackers could prevent the
   exchange from completing by exhausting the reassembly buffers.

RFC5996 advices to avoid IP fragmentation by using Hash and URL
format for certificates. For some reasons, this is not widely supported
by implementations and is not generic solution (some systems
transfer a fair amount of data in Configuration Payloads, in which
case Hash and URL won't help).

The immediate goal of IKE Fragmentation was to deal with
devices, that block IP fragments. But the solution is all in line
with RFC5996, advicing avoid IP fragmentation whenever possible.

> Further, this method does not necessarily avoid IP fragmentation. 
> Fragments are supposed to pass through a NAT (see my previous reference), 
> and when they do your mechanism wouldn't adjust the IKE size downward.

It doesn't matter. If IP fragmentation works in some environment - it is ok.
If it doesn't (either because some intermediate devices block
IP fragments or because an attacker is trying to exploit it),
then peers detect it and start using IKE Fragmentation.

>>> An equally viable solution (see ** below) would involve placing a
>>> single signed IKE message into a single IP packet, fragmenting that
>>> packet, and transmitting those fragments inside UDP datagrams. This
>>> approach could use the innermost IP encapsulation as the reassembly
>>> layer.
>>>
>>> **: Although this approach is susceptible to fragmentation overload,
>>> so too is transmitting the original message over IP that is naturally
>>> fragmented at the network layer anyway - this introducing no new
>>> vulnerability.
>>
>> But using IKE fragmentation in this case might eliminate this
>> vulnerability.
>
> It might, but again that's not your threat model and there are other 
> vulnerabilities this method opens up (e.g., replay attacks have a higher 
> CPU impact).

Why? Are you reffering to the fact, that with IKE Fragmentation you need to
check 4 additional bytes to detect a replay? Is it really a CPU impact?

>>> This IP-encapsulated IKE message would need the same negotiation
>>> described in section 2.3, but would avoid the complexity of most of
>>> the rest of the document, relying instead on existing IP reassembly.
>>>
>>> In conclusion, I don't see the need for the level of detail and
>>> mechanism proscribed, basically because it reinvents the wheel, but
>>> see no other substantial issues with it.
>>>
>>> Additionally, regarding PMTUD, this should be done using PLMTUD (RFC
>>> 4821), and should be described as such.
>>
>> IMHO, PLMTUD doesn't suite well for IKE. First, IKE is UDP-based
>> and RFC4821 has many restrictions when used with connectionless
>> datagram protocols, some of which are often difficult to satisfy in
>> environments,
>> where IKE is used.
>
> You are re-inventing PLMUTD. In places where you claim RFC4821 is too 
> complex - those are the places where your mechanism is deficient IMO. 
> E.g., positive verification that a message gets through as a confirmation 
> of when you've achieved a reasonable size.

It is sufficient for IKE.

>> Second, the goal of PLMTUD is to find _maximum_
>> MTU size that is not fragmented by IP, in other words, to maximize
>> throughput. For IKE the goal of PMTUD is to find _any_reasonable_
>> MTU size that won't be fragmented by IP. Throughput is not important.
>
> Agreed, but if that's the case, why not always send the smallest possible 
> message size (68 bytes)? It's because the overhead is too high (additional 
> headers, blocksize mismatch, and computational startup cost on each 
> block). Further, most transports don't care about 1400 vs. 1500 bytes; 
> they're searching for a rough approximation of a 'large' packet, as are 
> you IMO.

I think, the main difference is that PLMTUD is always trying
to increase MTU size when everything works fine, while in IKE
PMTUD is supposed to decrease it only if packets don't get through.
For IKE the former is just not needed.

>> Third, PMTU discovery in the draft is completely optional and one may
>> use any other mechanisms (including PLMTUD, if it is available)
>> to obtain MTU information.
>
> The transport folks spent a lot of time on PLMTUD; there are a lot of 
> lessons there that can and should be applied here. PLMTUD is already 
> flexible enough to apply here directly. You can always reinvent the wheel, 
> of course.

I appreciate the work transport folks has done. I will also appreciate
if you point out what exact lessons should be applied here and why.
And you may consider PMTUD in IKE as simplified PLMTUD,
implemented according with Section 10.4 of RFC4821.

> Joe

Regards,
Valery. 


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From: "Valery Smyslov" <svanru@gmail.com>
To: "Paul Wouters" <paul@cypherpunks.ca>
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Hi Paul,

>>> attacks (e.g., overloading the receiver with false fragments).
>>
>> This attack is always possible both with and without fragmentation
>> (just overloading with false full messages) and IKE is designed to 
>> withstand it.
>
> Partially. For unfragmented IKE, the cookies/SPI mitigate this if
> you give preference to IKE messages further in the processing queue.
> With this draft, I'm forced to do more crypto on IKE packets before I
> can decide if these need prioritization or not (with the exception of
> the first packet, which cannot get fragmented anyway)

The SPIs are still there and you are supposed to check them
before you will do any crypto. What is the difference?

>> Handling fragmentation in IKE doesn't make it more vulnerable to this 
>> attack.
>
> Handling encrypted fragments does, because you will spend more time
> doing the crypto. Although this still happens after the cookie/spi
> verification. You are doing an additional round of crypto on each fragment
> in addition to the regular IKE crypto you need to perform on the assembled
> packet. Fragments could be made artificially long and the IKE packet
> artificial big, resulting in much more crypto. Now I think this might

No, there is no additional round of crypto. After you has verified
and decrypted fragment you will store it in decrypted form.
After reassembly has finished you will get decrypted message.
No more crypto to apply.

> not matter much, because the attacker needs to be local and they will
> only be local to a remote endpoint which has enough CPU, not the ipsec
> core gateway that actually might run out of CPU power when under attack.
>
>> On the other hand, handling fragmentation outside IKE (on IP level)
>> or using unauthenticated fragments in IKE, makes it susceptible to an 
>> attack,
>> described in the rationale, i.e. when attacker spending virtually zero 
>> resources
>> can stop IKE communication by infrequently injecting bad fragments,
>> which results in "poisoning" of reassembling queue.
>
> But the attack is theoretical and of no practical value.

I am not so confident.

> 1) on-path attacker (remote)
>
> They can just DOS you regardless by withholding packets.

There may be cases when attacker, even on-path,
could only eavesdrop and inject packets, but not to block them.

> 2) off-path attacker (remote)
>
> These attackers don't know the cookies/spi's so they can't really sent
> a forged fragment you would use for assembly - protected with crypto or
> not. It is also impossible for them to time their attack so that one of
> their fragments intersect with one of your fragments so they would need
> to start sending fragments before the endpoint does but using mostly
> wrong cookies/spis and would swamp an endpoint with easilly detectable
> forged packets.

Again, I am not so confident that it is impossible, especially for
particular implementation. Remember, that TCP ISN was also
considered impossible to predict untill we saw a successfull attack
on particular implementation of TCP.

> 3) shared medium like wifi hotspot (local)
>
> These are the only attackers with enough knowledge to sent sufficiently
> advanced bad fragments interfering with incoming fragments. They can
> already do all the damage they want without resorting to IKE fragment
> spoofing.  They can arp spoof the gateway, sent a Wifi de-association
> request, sent enough garbage to fill the air waves, and interfere with
> the IKE in such a way that you think the remote peer does not support
> fragmentation at all. So this draft protects against a local attacker
> that wants to masquerade as a network failure and only prevent you from
> doing IKE, hoping you will fall back to plaintext traffic. Software
> could notify the user that they seem to be under attack, which IMHO
> would be more valuable to the user than quiet resilience against the
> attack.

Sorry, I don't follow your logic here. Security software should
thwart the attacks as much, as it could, and not just
notify user that he is under attack and give up. DoS attacks
are difficult to thwart and sometimes are easy to perform.
So what is the reason not to thwart one kind of them?
I agree that others are still possible, but they may have
their own drawback for attacker and why should we
give him/her more options to choose from?

>> Using authenticated fragments prevents this kind of attack.
>
> Yes, it prevents one of the low hanging fruits. Which I don't think is
> worth the complexity of this draft compared to the existing
> fragmentation code out in the wild.

Is it really _much_ complex if you first fragment then encrypt compared
to first encrypt then fragment?

>> But using IKE fragmentation in this case might eliminate this 
>> vulnerability.
>
> At the price of adding more complexity to IKE. With the attacker not
> stopped from their goal.

See above.

>>> Additionally, regarding PMTUD, this should be done using PLMTUD (RFC 
>>> 4821), and should be described as such.
>>
>> IMHO, PLMTUD doesn't suite well for IKE. First, IKE is UDP-based
>> and RFC4821 has many restrictions when used with connectionless
>> datagram protocols, some of which are often difficult to satisfy in 
>> environments,
>> where IKE is used. Second, the goal of PLMTUD is to find _maximum_
>> MTU size that is not fragmented by IP, in other words, to maximize
>> throughput. For IKE the goal of PMTUD is to find _any_reasonable_
>> MTU size that won't be fragmented by IP. Throughput is not important.
>> Third, PMTU discovery in the draft is completely optional and one may
>> use any other mechanisms (including PLMTUD, if it is available)
>> to obtain MTU information.
>
> I agree. There is going to be like 2 to 4 fragments. Size does not
> matter. It's better to cut down on latency/RTT by using an MTU size
> that is for sure to work on the first go.
>
> But that's also why this complexity is not worth it. Fragmentation does
> only mean going from 1 IKE packet to 2-4 IKE packets. The timeframe
> to mess with these are in the 5ms range. If an attacker has that
> accuracy against you, you've lost already regardless. I doubt the
> attacker can even pull it off on wifi.
>
> The original IKE fragmentation is really nice and simple. Copy part of
> the original IKE header, add a single payload with fragment payload
> type and fragment number/max, sent fragments in a burst.

Not much difference from the draft - just move fragmentation
before encryption.

> This draft looks like it is fixing a security issue, while it really
> should be just about sending smaller packets as similar as possible
> to the original single packet. Messing with one IKE packet versus 2-4
> IKE packets is not a real problem that I think needs fixing.

It is about sending smaller packets, but doing so in a secure manner.

> Paul

Regards,
Valery. 


From gorry@erg.abdn.ac.uk  Wed Oct 23 05:00:44 2013
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From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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I've uploaded a draft Agenda. This divides the individual draft talks 
between those on transport protocols and DS/IS:

http://www.ietf.org/proceedings/88/agenda/agenda-88-tsvwg

If there are any presenters not listed here who would like to be 
considered please get back asasp, we can then start confirming slots to 
those marked ???.

Best wishes,

Gorry

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To: Brian E Carpenter <brian.e.carpenter@gmail.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
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From touch@isi.edu  Wed Oct 23 10:23:42 2013
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On 10/22/2013 11:25 PM, Valery Smyslov wrote:
> I appreciate the work transport folks has done. I will also appreciate
> if you point out what exact lessons should be applied here and why.
> And you may consider PMTUD in IKE as simplified PLMTUD,
> implemented according with Section 10.4 of RFC4821.

 From the second sentence of that section: Because you never generate 
distinct probes, finding out when the MTU fails is tied to timeouts in 
your message exchange system, rather than being decoupled.

 From Section 9: You don't talk about trying to make sure that IPv4 DF 
is set, as per Section 9, which means that even the conservative values 
you pick might continue to generate fragments downstream.

Finally, and this is separate from RFC4821, you don't mention the fact 
that dropping fragments is against the recommended behavior for NATs. I 
appreciate trying to get around that issue, but it's equally important 
to indicate this as a flaw of the network, rather than "just the way it is".

Joe


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From: "Valery Smyslov" <svanru@gmail.com>
To: "Joe Touch" <touch@isi.edu>
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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Hi Joe,

thank you for your suggestions. I still have some comments.

> On 10/22/2013 11:25 PM, Valery Smyslov wrote:
>> I appreciate the work transport folks has done. I will also appreciate
>> if you point out what exact lessons should be applied here and why.
>> And you may consider PMTUD in IKE as simplified PLMTUD,
>> implemented according with Section 10.4 of RFC4821.
>
> From the second sentence of that section: Because you never generate 
> distinct probes, finding out when the MTU fails is tied to timeouts in 
> your message exchange system, rather than being decoupled.

That's true. I don't see any requirements in section 10.4. that the probes
must be distinct. Request/reply nature of IKE's messages make them
well suitable for probes and I see no need here to introduce
separate mechanism, adding complexity, consuming bandwith and adding delay.
Moreover, section 10.4 explicitely allows using application data for probes.

> From Section 9: You don't talk about trying to make sure that IPv4 DF is 
> set, as per Section 9, which means that even the conservative values you 
> pick might continue to generate fragments downstream.

With all due respect, I have to disagree that setting DF bit is needed in 
case of IKE.
With IKE the goal is not to make sure that _no_ IP fragmentation takes place
on the path, but to make sure that IP fragmentation issues, if any,
do not prevent IKE communications. So, IP fragmentation is present on the 
path,
but it doesn't bother us - let it be. If it doesn't allow IKE to work
(either by dropping IP fragments or by attacking destination host with
bogus fragments) - we will try to avoid it. So, in your example, even if
IP fragments are generated downstream, but those fragments are passed - let 
them be.

Note, that this approach is in line with advices, given for IKE in the paper

C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
              for UDP-based protocols", ACM Conference on Computer and
              Communications Security, October 2003.

that is reffered by RFC5996 as recommendations to protect against IP
fragmentation issues. Section 3.3 of this paper discusses the ways PMTUD
should be done in IKE and finds it more advantageous not to set DF bit.

> Finally, and this is separate from RFC4821, you don't mention the fact 
> that dropping fragments is against the recommended behavior for NATs. I 
> appreciate trying to get around that issue, but it's equally important to 
> indicate this as a flaw of the network, rather than "just the way it is".

That's true and I will definitely add some words about it.

> Joe

Thank you,
Valery Smyslov. 


From touch@isi.edu  Thu Oct 24 10:05:38 2013
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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On 10/23/2013 11:42 PM, Valery Smyslov wrote:
> Hi Joe,
>
> thank you for your suggestions. I still have some comments.
>
>> On 10/22/2013 11:25 PM, Valery Smyslov wrote:
>>> I appreciate the work transport folks has done. I will also appreciate
>>> if you point out what exact lessons should be applied here and why.
>>> And you may consider PMTUD in IKE as simplified PLMTUD,
>>> implemented according with Section 10.4 of RFC4821.
>>
>> From the second sentence of that section: Because you never generate
>> distinct probes, finding out when the MTU fails is tied to timeouts in
>> your message exchange system, rather than being decoupled.
>
> That's true. I don't see any requirements in section 10.4. that the probes
> must be distinct. Request/reply nature of IKE's messages make them
> well suitable for probes and I see no need here to introduce
> separate mechanism, adding complexity, consuming bandwith and adding delay.
> Moreover, section 10.4 explicitely allows using application data for
> probes.

Yes, but there are two components:
	- the probes
	- the timeouts

You're using existing IKE messages *and* existing timeouts to determine 
when there is a problem. A separate timer would be useful, if only to 
allow you to decouple fragment retransmission from IKE transaction retries.

>> From Section 9: You don't talk about trying to make sure that IPv4 DF
>> is set, as per Section 9, which means that even the conservative
>> values you pick might continue to generate fragments downstream.
>
> With all due respect, I have to disagree that setting DF bit is needed
> in case of IKE.
> With IKE the goal is not to make sure that _no_ IP fragmentation takes
> place
> on the path, but to make sure that IP fragmentation issues, if any,
> do not prevent IKE communications. So, IP fragmentation is present on
> the path,
> but it doesn't bother us - let it be.

Agreed. But that's the case when IKE works with whatever message size 
you start with.

> If it doesn't allow IKE to work
> (either by dropping IP fragments or by attacking destination host with
> bogus fragments) - we will try to avoid it. So, in your example, even if
> IP fragments are generated downstream, but those fragments are passed -
> let them be.

Your mechanism kicks in (presumably) when IKE fails, and then you retry 
with application-layer fragments. App-layer fragments are useless if the 
network level then fragments the messages. That's why you should set DF=1.

> Note, that this approach is in line with advices, given for IKE in the
> paper
>
> C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
>               for UDP-based protocols", ACM Conference on Computer and
>               Communications Security, October 2003.

That paper doesn't consider IKE-level fragmentation, which you're 
introducing. I agree that DF=0 for IKE without IKE-level fragmentation.

> that is reffered by RFC5996 as recommendations to protect against IP
> fragmentation issues. Section 3.3 of this paper discusses the ways PMTUD
> should be done in IKE and finds it more advantageous not to set DF bit.

That too refers to IKE without IKE-level fragmentation (see Sec 2). 
Again, once you introduce fragmentation at the IKE-level to avoid IP 
fragmentation, it makes no sense to continue to allow it at the IP level.

Joe

From touch@isi.edu  Fri Oct 25 13:18:51 2013
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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On 10/24/2013 10:45 PM, Valery Smyslov wrote:
...
>> You're using existing IKE messages *and* existing timeouts to
>> determine when there is a problem. A separate timer would be useful,
>> if only to allow you to decouple fragment retransmission from IKE
>> transaction retries.
>
> No, the timeouts are different. I should have made it more expplicit in
> the draft.

That'd be useful.

...
> Always setting DF bit in this case will probably increase the delay
> before IKE SA is set up (as more probes will need to be done).

Except that if you continue to allow IP fragmentation, you can't claim 
your solution is robust to IP fragment poisoning.

>>> Note, that this approach is in line with advices, given for IKE in the
>>> paper
>>>
>>> C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
>>>               for UDP-based protocols", ACM Conference on Computer and
>>>               Communications Security, October 2003.
>>
>> That paper doesn't consider IKE-level fragmentation, which you're
>> introducing. I agree that DF=0 for IKE without IKE-level fragmentation.
>
> It does, in Section 3.3.

Sorry - I missed that. But that section also gives good reasons why this 
is a bad idea in IKE too.

Joe

From mattmathis@google.com  Fri Oct 25 13:41:08 2013
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Date: Fri, 25 Oct 2013 13:41:05 -0700
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From: Matt Mathis <mattmathis@google.com>
To: Valery Smyslov <svanru@gmail.com>
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Cc: tsvwg <tsvwg@ietf.org>, Joe Touch <touch@isi.edu>, tsv-ads@tools.ietf.org, ipsec@ietf.org, draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org, "tsv-dir@ietf.org" <tsv-dir@ietf.org>
Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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I concur with Joe: once you have enough machinery work well with IPv6
fragmentation semantics, you should use it for IPv4 too, and
unconditionally set DF.   This probably applies to *all* protocols.

IPv4 reassembly is hopelessly out of scale.  IP ID wrap times are likely to
be sub second for any large CGN connecting to any large service.....  They
might even be shorter than the queuing times.

I suspect that if you re-review decade old papers on fragmentation, you
will find some scale assumptions that are no longer correct.  In that time
the Internet has moved at least another two orders of magnitude in packet
rates.

Thanks,
--MM--
The best way to predict the future is to create it.  - Alan Kay

Privacy matters!  We know from recent events that people are using our
services to speak in defiance of unjust governments.   We treat privacy and
security as matters of life and death, because for some users, they are.


On Fri, Oct 25, 2013 at 1:18 PM, Joe Touch <touch@isi.edu> wrote:

>
>
> On 10/24/2013 10:45 PM, Valery Smyslov wrote:
> ...
>
>> You're using existing IKE messages *and* existing timeouts to
>>> determine when there is a problem. A separate timer would be useful,
>>> if only to allow you to decouple fragment retransmission from IKE
>>> transaction retries.
>>>
>>
>> No, the timeouts are different. I should have made it more expplicit in
>> the draft.
>>
>
> That'd be useful.
>
> ...
>
>> Always setting DF bit in this case will probably increase the delay
>> before IKE SA is set up (as more probes will need to be done).
>>
>
> Except that if you continue to allow IP fragmentation, you can't claim
> your solution is robust to IP fragment poisoning.
>
>  Note, that this approach is in line with advices, given for IKE in the
>>>> paper
>>>>
>>>> C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
>>>>               for UDP-based protocols", ACM Conference on Computer and
>>>>               Communications Security, October 2003.
>>>>
>>>
>>> That paper doesn't consider IKE-level fragmentation, which you're
>>> introducing. I agree that DF=0 for IKE without IKE-level fragmentation.
>>>
>>
>> It does, in Section 3.3.
>>
>
> Sorry - I missed that. But that section also gives good reasons why this
> is a bad idea in IKE too.
>
> Joe
>

--089e0158a9aafbfe5404e996c45d
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I concur with Joe: once you have enough machinery work wel=
l with IPv6 fragmentation semantics, you should use it for IPv4 too, and un=
conditionally set DF. =A0 This probably applies to *all* protocols.<div><br=
>
</div><div>IPv4 reassembly is hopelessly out of scale. =A0IP ID wrap times =
are likely to be sub second for any large CGN connecting to any large servi=
ce..... =A0They might even be shorter than the queuing times.<div><br></div=
>
<div>I suspect that if you re-review decade old papers on fragmentation, yo=
u will find some scale assumptions that are no longer correct. =A0In that t=
ime the Internet has moved at least another two orders of magnitude in pack=
et rates.</div>
</div></div><div class=3D"gmail_extra"><br clear=3D"all"><div><div dir=3D"l=
tr"><div>Thanks,</div>--MM--<br>The best way to predict the future is to cr=
eate it. =A0- Alan Kay<br><br>Privacy matters! =A0We know from recent event=
s that people are using our services to speak in defiance of unjust governm=
ents. =A0 We treat privacy and security as matters of life and death, becau=
se for some users, they are.</div>
</div>
<br><br><div class=3D"gmail_quote">On Fri, Oct 25, 2013 at 1:18 PM, Joe Tou=
ch <span dir=3D"ltr">&lt;<a href=3D"mailto:touch@isi.edu" target=3D"_blank"=
>touch@isi.edu</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
<br>
On 10/24/2013 10:45 PM, Valery Smyslov wrote:<br>
...<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">

You&#39;re using existing IKE messages *and* existing timeouts to<br>
determine when there is a problem. A separate timer would be useful,<br>
if only to allow you to decouple fragment retransmission from IKE<br>
transaction retries.<br>
</blockquote>
<br></div>
No, the timeouts are different. I should have made it more expplicit in<br>
the draft.<br>
</blockquote>
<br>
That&#39;d be useful.<br>
<br>
...<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Always setting DF bit in this case will probably increase the delay<br>
before IKE SA is set up (as more probes will need to be done).<br>
</blockquote>
<br>
Except that if you continue to allow IP fragmentation, you can&#39;t claim =
your solution is robust to IP fragment poisoning.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Note, that this approach is in line with advices, given for IKE in the<br>
paper<br>
<br>
C. Kaufman, R. Perlman, and B. Sommerfeld, &quot;DoS protection<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 for UDP-based protocols&quot;, ACM Conference o=
n Computer and<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 Communications Security, October 2003.<br>
</blockquote>
<br>
That paper doesn&#39;t consider IKE-level fragmentation, which you&#39;re<b=
r>
introducing. I agree that DF=3D0 for IKE without IKE-level fragmentation.<b=
r>
</blockquote>
<br></div>
It does, in Section 3.3.<br>
</blockquote>
<br>
Sorry - I missed that. But that section also gives good reasons why this is=
 a bad idea in IKE too.<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
Joe<br>
</font></span></blockquote></div><br></div>

--089e0158a9aafbfe5404e996c45d--

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Subject: Re: [tsvwg] [aqm] Immediate and Explicit Congestion Notification (was: Guidelines for Adding Congestion Notification to Protocols that Encapsulate IP)
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Mat,

At 11:56 22/10/2013, Matthew Ford wrote:

>On 22 Oct 2013, at 00:40, Bob Briscoe <bob.briscoe@bt.com> wrote:
>
> > There's more to it - I've asked to present the whole story in Vancouver.
>
>Cool, looking forward to it. I'll prime the question pump now if I may:
>
>Does the transport need to know that it is connected to a network 
>that will signal ECN in this fashion?

Here's the outline of the talk I'm planning, that I sent to Martin 
originally. Most of the second half addresses your question.

Exec summary
* Early tests show promise that we may have found a way to make the 
ultra-low queuing delay of data centre TCP incrementally deployable 
on the public Internet
* For rtcweb, we need to address
   a) cc for r-t media [rmcat w-g in progress]
   b) Making TCP nicer
   c) minimise ability of TCP to bloat queues [AQM w-g now in progress]
   This addresses b) & c)

The problem
* All AQMs delay dropping for about one (hard-coded) worst-case RTT, 
in case a burst dissipates (allegedly a 'good queue' according to Van Jacobson)
* For a flow with 1/10 or 1/100 of this RTT (e.g. from a CDN or your 
home media server), any congestion signal is delayed tens or hundreds 
of its own RTTs by these AQMs.
* A TCP flow in slow-start doesn't need the burst smoothed anyway
   - delaying the signal just makes slow-start overshoot more
   - a TCP in slow-start knows that it won't allow the burst to 
dissipate anyway

The solution: make ECN also mean "Immediate Congestion Notification"?
* For ECN-capable packets, shift the job of hiding bursts from network to host:
   - the network signals ECN with no smoothing delay
   - then the transport can hide bursts of ECN signals from itself
   - the transport knows
     o whether it's TCP or RTP etc,
     o whether its in congestion avoidance or slow-start,
     o and it knows its RTT,
     o so it can know whether to respond immediately or to smooth the signals,
     o and if so, over what time
   - then short RTT flows can smooth the signals with only the delay 
of their /own/ RTT
     o so they can fill troughs and absorb peaks that longer RTT flows cannot
   - a TCP only needs to smooth the signals if in congestion avoidance
     o in slow start, it can respond immediately, thus reducing overshoot

Incremental Deployment:
* Immediate congestion notification doesn't need new AQM implementation
   - it can use the widely implemented WRED algorithm with an 
unexpected configuration
* The network classifies packets for this AQM treatment based on 
their ECN-capability
   - Without ECN, it smoothes the queue before signalling drops
   - With ECN, it signals immediately, without any smoothing delay
   - (as today, the operator can still use WRED with the Diffserv field too)
* For TCP apps, the stack will use 'DCTCP' (we've tweaked it), if the 
ends negotiate ECN with the accurate feedback capability.
* It should 'just work' if an RTP app or a Reno TCP uses ECN.

The request:
* Much more evaluation to do, but first we want to know:
   - if the idea works, would the IETF have an appetite for tweaking 
the definition of ECN so it is merely equivalent to drop in the long 
term, but the dynamics need not be equivalent.


Much better than the ECN that didn't get deployed
* This is Explicit and Immediate Congestion Notification (EICN?)
   - same wire protocol, much greater benefits
* The advantage of the original ECN (avoiding congestive loss) was 
too small to be worth the deployment hassle
* Predictable ultra-low latency without loss too (similar to 
DCTCP-ECN) would be worth deploying
* But we all thought DCTCP could only be deployed in isolation (e.g. 
data centres)
   - we all thought DCTCP traffic would starve alongside today's TCP traffic
   - because in a DCTCP queue, the ECN threshold is lower than you 
would trigger drop
   - and we thought ECN & drop had to be equivalent.
* We believe we've found a way to ensure DCTCP-ECN traffic doesn't starve
   - we still make DCTCP-ECN equivalent to drop in the long-run, but 
not in its dynamics

More info
A paper with early results is under submission  (available on 
request). It shows neither this approach nor legacy traffic can 
starve the other under a wide range of conditions, but we have a lot 
more work to test the limits.



Bob


>Mat
>
>_______________________________________________
>aqm mailing list
>aqm@ietf.org
>https://www.ietf.org/mailman/listinfo/aqm

________________________________________________________________
Bob Briscoe,                                                  BT 
--=====================_552357903==.ALT
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<html>
<body>
Mat,<br><br>
At 11:56 22/10/2013, Matthew Ford wrote:<br><br>
<blockquote type=cite class=cite cite="">On 22 Oct 2013, at 00:40, Bob
Briscoe &lt;bob.briscoe@bt.com&gt; wrote:<br><br>
&gt; There's more to it - I've asked to present the whole story in
Vancouver.<br><br>
Cool, looking forward to it. I'll prime the question pump now if I
may:<br><br>
Does the transport need to know that it is connected to a network that
will signal ECN in this fashion?</blockquote><br>
Here's the outline of the talk I'm planning, that I sent to Martin
originally. Most of the second half addresses your question.<br><br>
<b><u>Exec summary<br>
</u></b>* Early tests show promise that we may have found a way to make
the ultra-low queuing delay of data centre TCP incrementally deployable
on the public Internet <br>
* For rtcweb, we need to address <br>
&nbsp; a) cc for r-t media [rmcat w-g in progress]<br>
&nbsp; b) Making TCP nicer<br>
&nbsp; c) minimise ability of TCP to bloat queues [AQM w-g now in
progress]<br>
&nbsp; This addresses b) &amp; c)<br><br>
<b><u>The problem<br>
</u></b>* All AQMs delay dropping for about one (hard-coded) worst-case
RTT, in case a burst dissipates (allegedly a 'good queue' according to
Van Jacobson)<br>
* For a flow with 1/10 or 1/100 of this RTT (e.g. from a CDN or your home
media server), any congestion signal is delayed tens or hundreds of its
own RTTs by these AQMs.<br>
* A TCP flow in slow-start doesn't need the burst smoothed anyway<br>
&nbsp; - delaying the signal just makes slow-start overshoot more<br>
&nbsp; - a TCP in slow-start knows that it won't allow the burst to
dissipate anyway<br><br>
<b><u>The solution</u></b>: make ECN also mean &quot;Immediate Congestion
Notification&quot;?<br>
* For ECN-capable packets, shift the job of hiding bursts from network to
host:<br>
&nbsp; - the network signals ECN with no smoothing delay<br>
&nbsp; - then the transport can hide bursts of ECN signals from
itself<br>
&nbsp; - the transport knows <br>
&nbsp;&nbsp;&nbsp; o whether it's TCP or RTP etc, <br>
&nbsp;&nbsp;&nbsp; o whether its in congestion avoidance or slow-start,
<br>
&nbsp;&nbsp;&nbsp; o and it knows its RTT, <br>
&nbsp;&nbsp;&nbsp; o so it can know whether to respond immediately or to
smooth the signals, <br>
&nbsp;&nbsp;&nbsp; o and if so, over what time<br>
&nbsp; - then short RTT flows can smooth the signals with only the delay
of their /own/ RTT <br>
&nbsp;&nbsp;&nbsp; o so they can fill troughs and absorb peaks that
longer RTT flows cannot<br>
&nbsp; - a TCP only needs to smooth the signals if in congestion
avoidance<br>
&nbsp;&nbsp;&nbsp; o in slow start, it can respond immediately, thus
reducing overshoot<br><br>
<b><u>Incremental Deployment:<br>
</u></b>* Immediate congestion notification doesn't need new AQM
implementation <br>
&nbsp; - it can use the widely implemented WRED algorithm with an
unexpected configuration<br>
* The network classifies packets for this AQM treatment based on their
ECN-capability<br>
&nbsp; - Without ECN, it smoothes the queue before signalling drops<br>
&nbsp; - With ECN, it signals immediately, without any smoothing
delay<br>
&nbsp; - (as today, the operator can still use WRED with the Diffserv
field too)<br>
* For TCP apps, the stack will use 'DCTCP' (we've tweaked it), if the
ends negotiate ECN with the accurate feedback capability.<br>
* It should 'just work' if an RTP app or a Reno TCP uses ECN.<br><br>
<b><u>The request:<br>
</u></b>* Much more evaluation to do, but first we want to know:<br>
&nbsp; - if the idea works, would the IETF have an appetite for tweaking
the definition of ECN so it is merely equivalent to drop in the long
term, but the dynamics need not be equivalent.<br><br>
<br>
<b><u>Much better than the ECN that didn't get deployed<br>
</u></b>* This is Explicit and Immediate Congestion Notification
(EICN?)<br>
&nbsp; - same wire protocol, much greater benefits<br>
* The advantage of the original ECN (avoiding congestive loss) was too
small to be worth the deployment hassle<br>
* Predictable ultra-low latency without loss too (similar to DCTCP-ECN)
would be worth deploying<br>
* But we all thought DCTCP could only be deployed in isolation (e.g. data
centres)<br>
&nbsp; - we all thought DCTCP traffic would starve alongside today's TCP
traffic<br>
&nbsp; - because in a DCTCP queue, the ECN threshold is lower than you
would trigger drop<br>
&nbsp; - and we thought ECN &amp; drop had to be equivalent.<br>
* We believe we've found a way to ensure DCTCP-ECN traffic doesn't
starve<br>
&nbsp; - we still make DCTCP-ECN equivalent to drop in the long-run, but
not in its dynamics<br><br>
<b><u>More info<br>
</u></b>A paper with early results is under submission&nbsp; (available
on request). It shows neither this approach nor legacy traffic can starve
the other under a wide range of conditions, but we have a lot more work
to test the limits.<br><br>
<br><br>
Bob<br><br>
<br>
<blockquote type=cite class=cite cite="">Mat<br><br>
_______________________________________________<br>
aqm mailing list<br>
aqm@ietf.org<br>
<a href="https://www.ietf.org/mailman/listinfo/aqm" eudora="autourl">
https://www.ietf.org/mailman/listinfo/aqm</a></blockquote>
<x-sigsep><p></x-sigsep>
________________________________________________________________<br>
Bob
Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
BT</body>
</html>

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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Date: Fri, 25 Oct 2013 21:03:51 -0400
Thread-Topic: Asking for tsvwg meeting time
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Subject: [tsvwg] Asking for tsvwg meeting time
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I thought I'd pass along some wise advice from Melinda Shore, one of
the chairs of the opsawg WG - it also applies to tsvwg WG agenda time:

	It seems to have become a matter of standard practice for
	someone to request time on the agenda to discuss a draft
	that we've never seen before and which has received no discussion.

	We always write back and tell them there really needs to be
	discussion on the mailing list so that we can gauge interest,
	relevance, and quality, and so the author sends email to
	tsvwg at ietf.org.  You can save yourself a round trip and
	possibly a bit of time by starting a mailing list discussion
	first.

http://www.ietf.org/mail-archive/web/opsawg/current/msg02987.html

FYI, --David (for the tsvwg WG co-chairs)
----------------------------------------------------
David L. Black, Distinguished Engineer
EMC Corporation, 176 South St., Hopkinton, MA=A0 01748
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----------------------------------------------------



From john@jlc.net  Sat Oct 26 05:16:43 2013
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Date: Sat, 26 Oct 2013 08:16:32 -0400
From: John Leslie <john@jlc.net>
To: Bob Briscoe <bob.briscoe@bt.com>
Message-ID: <20131026121632.GJ65952@verdi>
References: <201309171233.r8HCXKkV003952@bagheera.jungle.bt.co.uk> <CAA93jw7SaAZtVTYPa0d_d2xi+oqpOJmcx9=zWd6EvraaPXE1KA@mail.gmail.com> <201310212340.r9LNee5p025162@bagheera.jungle.bt.co.uk> <657D1484-1B02-41F5-9FD1-0921439DF754@isoc.org> <201310252116.r9PLGAFp007348@bagheera.jungle.bt.co.uk>
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Cc: "aqm@ietf.org" <aqm@ietf.org>, tsvwg IETF list <tsvwg@ietf.org>
Subject: Re: [tsvwg] [aqm] Immediate and Explicit Congestion Notification (was: Guidelines for Adding Congestion Notification to Protocols that Encapsulate IP)
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   (If this continues, we've got to fix the title and cross-posting...)

Bob Briscoe <bob.briscoe@bt.com> wrote:
> 
> Exec summary
> * Early tests show promise that we may have found a way to make the 
> ultra-low queuing delay of data centre TCP incrementally deployable 
> on the public Internet
> * For rtcweb, we need to address
>   a) cc for r-t media [rmcat w-g in progress]
>   b) Making TCP nicer
>   c) minimise ability of TCP to bloat queues [AQM w-g now in progress]
>   This addresses b) & c)
> 
> The problem
> * All AQMs delay dropping for about one (hard-coded) worst-case RTT, 
> in case a burst dissipates (allegedly a 'good queue' according to Van 
> Jacobson)

   This assertion is going to need a lot of support.

   Bob is a man after my own heart suggesting that an ECN notification
may be sent earlier than a packet drop would be indicated. I don't know
if we can get there; but IMHO that is essential to getting ECN deployed
and used.

   I don't think I agree with Bob that what's hard-coded is necessarily
a "worst-case" RTT -- and I'm quite sure I'm not willing to make any
pronouncement about "all AQMs".

   I suggest the talk might be more useful if Bob outlined the AQMs
currently in widespread use and detailed _how_ they delay dropping
for an estimated RTT.

> * For a flow with 1/10 or 1/100 of this RTT (e.g. from a CDN or your 
> home media server), any congestion signal is delayed tens or hundreds 
> of its own RTTs by these AQMs.

   Clearly, RTTs differing by a factor of ten are quite common at most
nodes traversed in a typical path; and it seems _very_ suboptimal to
have the responsibility for guessing the RTT at the node which must
drop packets.

> * A TCP flow in slow-start doesn't need the burst smoothed anyway
>   - delaying the signal just makes slow-start overshoot more
>   - a TCP in slow-start knows that it won't allow the burst to 
> dissipate anyway

   A critical point! (It seems obvious to me, but is it obvious to
everyone?)

> The solution: make ECN also mean "Immediate Congestion Notification"?
> * For ECN-capable packets, shift the job of hiding bursts from network to 
> host:
>   - the network signals ECN with no smoothing delay
>   - then the transport can hide bursts of ECN signals from itself

   But can we get there from here?

   The node doing the ECN notification _can't_ know how the transport
will react; and the transport receiving and ECN notification can't know
whether the forwarding node has "smoothed" the signal. (It is truly a
shame we haven't left any bits for signals like this!)

>   - the transport knows
>     o whether it's TCP or RTP etc,
>     o whether its in congestion avoidance or slow-start,
>     o and it knows its RTT,
>     o so it can know whether to respond immediately or to smooth the 
>     signals,
>     o and if so, over what time

   Yes, but it can't know what smoothing may already have been applied.

>   - then short RTT flows can smooth the signals with only the delay 
> of their /own/ RTT
>     o so they can fill troughs and absorb peaks that longer RTT flows cannot
>   - a TCP only needs to smooth the signals if in congestion avoidance
>     o in slow start, it can respond immediately, thus reducing overshoot

   This would, IMHO, improve "slow start".

> Incremental Deployment:
> * Immediate congestion notification doesn't need new AQM implementation
>   - it can use the widely implemented WRED algorithm with an 
> unexpected configuration

   Bob is beginning to lose me here. Does he mean that a forwarding node
would apply WRED for both drop and ECN, but with different parameters?

> * The network classifies packets for this AQM treatment based on 
> their ECN-capability
>   - Without ECN, it smoothes the queue before signalling drops

   Bob has lost me now -- apparently he doesn't mean different
parameters... and I don't recognize this "smoothing" step in WRED.

>   - With ECN, it signals immediately, without any smoothing delay
>   - (as today, the operator can still use WRED with the Diffserv field too)

   (Do we need to confuse this discussion by adding diffserv?)

> * For TCP apps, the stack will use 'DCTCP' (we've tweaked it), if the 
> ends negotiate ECN with the accurate feedback capability.

   Have we settled on "accurate feedback" already? I thought that was
still under discussion. (I don't follow exactly what it adds...)

> * It should 'just work' if an RTP app or a Reno TCP uses ECN.

   I don't see any way for a Reno transport using ECN to avoid being
starved if ECN arrives earlier (without notice).

> The request:
> * Much more evaluation to do, but first we want to know:
>   - if the idea works, would the IETF have an appetite for tweaking 
> the definition of ECN so it is merely equivalent to drop in the long 
> term, but the dynamics need not be equivalent.

   There's a good question there; but I don't think we're ready for it.

   I'd really like to discuss the dynamics of responding more quickly
but perhaps less drastically for almost any real-time flow.

   But proving "equivalence in the long term" seems too hard.

> Much better than the ECN that didn't get deployed
> * This is Explicit and Immediate Congestion Notification (EICN?)
>   - same wire protocol, much greater benefits
> * The advantage of the original ECN (avoiding congestive loss) was 
> too small to be worth the deployment hassle

   Actually, I don't agree that was the problem -- instead I believe
the code has been deployed but administratively suppressed because
the operators don't trust the transports. There _is_ a significant
improvement from one-RTT reaction instead of several (to detect a
drop), but the whole process is just too complicated, while the
opportunity for abuse remains obvious.

> * Predictable ultra-low latency without loss too (similar to 
> DCTCP-ECN) would be worth deploying

   I'm optimistic that latency will become an easier argument.

> * But we all thought DCTCP could only be deployed in isolation (e.g. 
> data centres)
>   - we all thought DCTCP traffic would starve alongside today's TCP traffic
>   - because in a DCTCP queue, the ECN threshold is lower than you 
> would trigger drop
>   - and we thought ECN & drop had to be equivalent.

   (I'm not sure we'll succeed at breaking that "equivalence"...)

> * We believe we've found a way to ensure DCTCP-ECN traffic doesn't starve
>   - we still make DCTCP-ECN equivalent to drop in the long-run, but 
> not in its dynamics

   (I'm still not sure it's worth arguing the "long-run".)

--
John Leslie <john@jlc.net>

From rs@netapp.com  Sat Oct 26 15:42:29 2013
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: John Leslie <john@jlc.net>, Bob Briscoe <bob.briscoe@bt.com>
Thread-Topic: [aqm] [tsvwg] Immediate and Explicit Congestion Notification (was: Guidelines for Adding Congestion Notification to	Protocols that Encapsulate IP)
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Subject: Re: [tsvwg] [aqm] Immediate and Explicit Congestion Notification	(was: Guidelines for Adding Congestion Notification to	Protocols that Encapsulate IP)
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HI John,

(aqm chair hat off, and accecn author hat on)

>> * For TCP apps, the stack will use 'DCTCP' (we've tweaked it), if the
>> ends negotiate ECN with the accurate feedback capability.
>=20
>    Have we settled on "accurate feedback" already? I thought that was
> still under discussion. (I don't follow exactly what it adds...)


The current requirements draft give a concise list of what benefits it shou=
ld bring.

There is also the (individual) draft on the TCP ECN modifications, that try=
 to deliver these benefits within the confines of the TCP header without re=
quiring additional bits.

The main issue with traditional ECN/TCP is, that you only get one bit of in=
formation per RTT (delivered over N segments sent in an RTT), thus the exte=
nt of congestion (number of CE marks per this flows RTT), and the fine-grai=
ned sequencing of the marks is lost.

AccECN aims to deliver these two signals to the sender - but what the sende=
r is then doing exactly with this much more detailed information is out of =
scope for the AccECN drafts so far.

http://tools.ietf.org/html/draft-ietf-tcpm-accecn-reqs

http://tools.ietf.org/html/draft-kuehlewind-tcpm-accurate-ecn

( http://tools.ietf.org/html/draft-kuehlewind-tcpm-accurate-ecn-option )

Any feedback on these drafts during the TCPM session is appreciated.


Richard Scheffenegger



From bob.briscoe@bt.com  Sun Oct 27 18:36:53 2013
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Subject: Re: [tsvwg] [aqm] Immediate and Explicit Congestion  Notification
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John, inline...

At 12:16 26/10/2013, John Leslie wrote:
>Bob Briscoe <bob.briscoe@bt.com> wrote:
> >
> > Exec summary
> > * Early tests show promise that we may have found a way to make the
> > ultra-low queuing delay of data centre TCP incrementally deployable
> > on the public Internet
> > * For rtcweb, we need to address
> >   a) cc for r-t media [rmcat w-g in progress]
> >   b) Making TCP nicer
> >   c) minimise ability of TCP to bloat queues [AQM w-g now in progress]
> >   This addresses b) & c)
> >
> > The problem
> > * All AQMs delay dropping for about one (hard-coded) worst-case RTT,
> > in case a burst dissipates (allegedly a 'good queue' according to Van
> > Jacobson)
>
>    This assertion is going to need a lot of support.
>
>    Bob is a man after my own heart suggesting that an ECN notification
>may be sent earlier than a packet drop would be indicated. I don't know
>if we can get there; but IMHO that is essential to getting ECN deployed
>and used.
>
>    I don't think I agree with Bob that what's hard-coded is necessarily
>a "worst-case" RTT -- and I'm quite sure I'm not willing to make any
>pronouncement about "all AQMs".
>
>    I suggest the talk might be more useful if Bob outlined the AQMs
>currently in widespread use and detailed _how_ they delay dropping
>for an estimated RTT.

You're right. The 'about' in my sentence was meant to indicate some 
leeway. The specifics depend on each AQM...

The only AQM I know of that doesn't smooth over some nominal RTT is 
DCTCP itself.

* CoDel was designed for 'interval' to be a worst-case (largest) RTT, 
which it recommends to be set to 100ms. One the queue has exceeded 
threshold, CoDel delays for time 'interval' before starting to signal 
congestion.

  - I already said how sluggish CoDel would be for a flow with a much 
shorter RTT than CoDel (others have made this point, e.g. for data 
centres : 
https://lists.bufferbloat.net/pipermail/codel/2012-August/000448.html).
  - And in the other direction, we already know that utilisation 
suffers fairly badly for flows with RTT significantly larger than 100ms.

* PIE suppresses all drops for time max_burst (set to 100ms by 
default) from when the drop probability it calculates (but doesn't 
necessarily use) first rises above zero. This is very similar to 
CoDel, and similar comments are applicable.

* RED requires the constant for its exponentially weighted moving 
average (w_q) to be set taking into account how many packets are 
likely to arrive at the link in a 'typical' RTT. Reverse engineering 
the values recommended by Sally Floyd in the RED paper and in her 
famous RED parameters Web page 
<http://www.icir.org/floyd/REDparameters.txt>, she recommended a 
'typical' RTT of about 130ms.

[BTW, I know of people who don't calculate w_q, but just use the 
value of "0.002" that Sally recommended for her 45Mb/s link in the 
original RED paper simulations (and repeated at 
<http://www.icir.org/floyd/REDparameters.txt>). This was calculated 
assuming about 500 packets arrive at a link (from all flows) in a 
typical RTT. Links have got a lot faster since 1993. Nonetheless, she 
was considering 45Mb/s for an aggregated link in those days, and it 
happens to be about right for a single user today.]


> > * For a flow with 1/10 or 1/100 of this RTT (e.g. from a CDN or your
> > home media server), any congestion signal is delayed tens or hundreds
> > of its own RTTs by these AQMs.
>
>    Clearly, RTTs differing by a factor of ten are quite common at most
>nodes traversed in a typical path; and it seems _very_ suboptimal to
>have the responsibility for guessing the RTT at the node which must
>drop packets.

For packets that do not support ECN, the dropping node has to make a 
guess at the RTT, so as not to drop packets unnecessarily, because 
drop is an impairment as well as a congestion signal. So a transport 
cannot 'undrop' packets.

Our point though is that a network node doesn't have to mimic this 
behaviour for ECN packets, because ECN is not an impairment. So a 
transport can un-ECN-mark packets (by smoothing out bursts itself).


> > * A TCP flow in slow-start doesn't need the burst smoothed anyway
> >   - delaying the signal just makes slow-start overshoot more
> >   - a TCP in slow-start knows that it won't allow the burst to
> > dissipate anyway
>
>    A critical point! (It seems obvious to me, but is it obvious to
>everyone?)
>
> > The solution: make ECN also mean "Immediate Congestion Notification"?
> > * For ECN-capable packets, shift the job of hiding bursts from network to
> > host:
> >   - the network signals ECN with no smoothing delay
> >   - then the transport can hide bursts of ECN signals from itself
>
>    But can we get there from here?
>
>    The node doing the ECN notification _can't_ know how the transport
>will react; and the transport receiving and ECN notification can't know
>whether the forwarding node has "smoothed" the signal. (It is truly a
>shame we haven't left any bits for signals like this!)

Well, we do have ECT(1) still only assigned experimentally and never 
used, which we could decide to use for this immediate ECN. However, 
first I want to see whether people think it might be feasible to just 
redefine the meaning of CE.

Rationale: So few buffers have ECN support turned on anyway that we 
should be able to redefine ECN so that many more will want to turn it on.

For those AQMs that already support ECN, we believe this 
retrospective change will make them only a little worse than they are 
already (and the operator can update them by simple reconfiguration 
anyway, and is more likely to do so, given these are clearly 
early-adopter networks).


> >   - the transport knows
> >     o whether it's TCP or RTP etc,
> >     o whether its in congestion avoidance or slow-start,
> >     o and it knows its RTT,
> >     o so it can know whether to respond immediately or to smooth the
> >     signals,
> >     o and if so, over what time
>
>    Yes, but it can't know what smoothing may already have been applied.

Yes. If this is a problem, we will have to consider using ECT(1) not CE.
But it's pretty academic when so few buffers support ECN.

The tiny proportion that do support ECN will already smooth by a 
'typical RTT' of about 100ms.

If a 20ms RTT flow adds smoothing over its own RTT to this, it will 
be smooth over 120ms.
The main problem there is not the extra 20ms, it's the original 
100ms, which we won't lose unless we make this change somehow.


> >   - then short RTT flows can smooth the signals with only the delay
> > of their /own/ RTT
> >     o so they can fill troughs and absorb peaks that longer RTT 
> flows cannot
> >   - a TCP only needs to smooth the signals if in congestion avoidance
> >     o in slow start, it can respond immediately, thus reducing overshoot
>
>    This would, IMHO, improve "slow start".
>
> > Incremental Deployment:
> > * Immediate congestion notification doesn't need new AQM implementation
> >   - it can use the widely implemented WRED algorithm with an
> > unexpected configuration
>
>    Bob is beginning to lose me here. Does he mean that a forwarding node
>would apply WRED for both drop and ECN, but with different parameters?
>
> > * The network classifies packets for this AQM treatment based on
> > their ECN-capability
> >   - Without ECN, it smoothes the queue before signalling drops
>
>    Bob has lost me now -- apparently he doesn't mean different
>parameters... and I don't recognize this "smoothing" step in WRED.

I do mean that a forwarding node would apply WRED for both drop and 
ECN, but with different parameters.

Each WRED policy-map includes a setting for this smoothing parameter, 
which Cisco calls the exponential-weighting-constant. Many people 
don't notice it's there and they just leave it at the default. For 
instance, Cisco set it to
2^(-9) ~ 0.002 by default for each of the WRED policy-maps (see 
http://www.cisco.com/en/US/docs/ios/12_0s/feature/guide/fswfq26.html#wp1039982).


> >   - With ECN, it signals immediately, without any smoothing delay
> >   - (as today, the operator can still use WRED with the Diffserv field too)
>
>    (Do we need to confuse this discussion by adding diffserv?)

A non-Diffserv network still doesn't need to worry about Diffserv.

I put this in parentheses because, if WRED is used today, it is 
usually used with Diffserv, and I didn't want anyone to worry that 
they wouldn't be able to continue to do this (e.g. BT use WRED with 
Diffserv in enterprise networks, as do many other carriers).


> > * For TCP apps, the stack will use 'DCTCP' (we've tweaked it), if the
> > ends negotiate ECN with the accurate feedback capability.
>
>    Have we settled on "accurate feedback" already? I thought that was
>still under discussion. (I don't follow exactly what it adds...)

See response from Richard Scheffenegger. Essentially the TCPM WG has 
accepted the requirements doc, but not decided between the mechanisms on offer.


> > * It should 'just work' if an RTP app or a Reno TCP uses ECN.
>
>    I don't see any way for a Reno transport using ECN to avoid being
>starved if ECN arrives earlier (without notice).

We haven't tested legacy Reno with ECN yet (we figured legacy Reno 
without ECN is a lot more prevalent, so focused on this first). 
Nonetheless, Reno-ECN is unlikely to starve, because starvation is 
about long-running behaviour, and once a flow has run for more than a 
couple of 100ms RTTs, the immediate ECN signals should be no 
different from a smoothed ECN. I suspect Reno-ECN might be worse in 
its short-term dynamics. But remember Reno-ECN is likely to be a tiny 
corner-case.


> > The request:
> > * Much more evaluation to do, but first we want to know:
> >   - if the idea works, would the IETF have an appetite for tweaking
> > the definition of ECN so it is merely equivalent to drop in the long
> > term, but the dynamics need not be equivalent.
>
>    There's a good question there; but I don't think we're ready for it.

At this stage, even we haven't got many answers. So I'm not asking 
the IETF to answer the question right now. I'm merely saying, /if/ 
our idea works, is there at least an /appetite/ in the IETF for 
reconsidering the definition of ECN?

We wanted to make the IETF aware of this research early, because it 
might want to at least hold off on any actions that would otherwise 
close off this option.

And if we find that any change is completely out of the question, we 
have to try a different tack (e.g. ECT(1)).


>    I'd really like to discuss the dynamics of responding more quickly
>but perhaps less drastically for almost any real-time flow.
>
>    But proving "equivalence in the long term" seems too hard.

This should be the easy part, because the longer that conditions are 
stable, a smoothed signal should tend towards an unsmoothed signal, 
all other factors being equal.

Equivalence during dynamics is the hard part, and I'm suggesting we 
don't sweat too much about that, as long as the performance 
evaluations are not too far apart.


> > Much better than the ECN that didn't get deployed
> > * This is Explicit and Immediate Congestion Notification (EICN?)
> >   - same wire protocol, much greater benefits
> > * The advantage of the original ECN (avoiding congestive loss) was
> > too small to be worth the deployment hassle
>
>    Actually, I don't agree that was the problem -- instead I believe
>the code has been deployed but administratively suppressed because
>the operators don't trust the transports. There _is_ a significant
>improvement from one-RTT reaction instead of several (to detect a
>drop), but the whole process is just too complicated, while the
>opportunity for abuse remains obvious.

I agree. That's the 'deployment hassle' side of my sentence - the 
extra trust-enhancing mechanisms that seemed necessary were too much 
pain for the small gain.


> > * Predictable ultra-low latency without loss too (similar to
> > DCTCP-ECN) would be worth deploying
>
>    I'm optimistic that latency will become an easier argument.
>
> > * But we all thought DCTCP could only be deployed in isolation (e.g.
> > data centres)
> >   - we all thought DCTCP traffic would starve alongside today's TCP traffic
> >   - because in a DCTCP queue, the ECN threshold is lower than you
> > would trigger drop
> >   - and we thought ECN & drop had to be equivalent.
>
>    (I'm not sure we'll succeed at breaking that "equivalence"...)
>
> > * We believe we've found a way to ensure DCTCP-ECN traffic doesn't starve
> >   - we still make DCTCP-ECN equivalent to drop in the long-run, but
> > not in its dynamics
>
>    (I'm still not sure it's worth arguing the "long-run".)

I mean competing long-running ECN & non-ECN flows stabilise at 
predictable rates, rather than one ratchetting itself down to nothing 
over time (starvation).

That's the primary concern of congestion control 'fairness', before 
anyone starts worrying about what the relative rates are. Given apps 
get different relative rates with different RTTs, with different size 
objects or by opening multiple flows, we don't need to sweat so much 
about precisely equal flow rates; but we must sweat about stable convergence.

Results so far show that the proposed idea is at least very robust 
against starvation.


Bob


>--
>John Leslie <john@jlc.net>

________________________________________________________________
Bob Briscoe,                                                  BT 


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As is probably well known at this point, I make a clear distinction
between networking problems in the data center and ones in the wild
and wooly world.

The data center portion of the universe is a couple hundred meters in
diameter, the other, I dunno, let's say 8x from here to the moon
(3,075,200,000m).

There are all sorts of things that work and are needed in the data
center that probably won't work outside it. People run straighter
cables, do microwave, use pause frames at layer 2, etc, etc, in order
to wring out the last nanosecond of performance, and certainly running
without loss is important in that world.

On Sun, Oct 27, 2013 at 6:36 PM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> John, inline...
>
> At 12:16 26/10/2013, John Leslie wrote:
>>
>> Bob Briscoe <bob.briscoe@bt.com> wrote:
>> >
>> > Exec summary
>> > * Early tests show promise that we may have found a way to make the

I'm failing to get excited lacking paper, testable code, and other proofs.

>> > ultra-low queuing delay of data centre TCP incrementally deployable
>> > on the public Internet

At the moment the edge of the internet is using things like cable and
gpon which have 2ms or more of inherent latency built into their
grant/response structures.

>> > * For rtcweb, we need to address
>> >   a) cc for r-t media [rmcat w-g in progress]
>> >   b) Making TCP nicer
>> >   c) minimise ability of TCP to bloat queues [AQM w-g now in progress]
>> >   This addresses b) & c)

I do strongly feel that webrtc needs good aqm and packet scheduling in
order to succeed, and that the webrtc folk should be testing out what
has already been developed (red, sfqred, codel, fq_codel, pie) and the
aqm folk testing the webrtc code.

I look forward to some interesting conference calls.

I gave the webrtc codebases a whirl against what aqm and packet
scheduling techniques we have already this past summer, the initial
results were encouraging, but I was able to easily crash most of the
browsers sooner than I could take a good, repeatable set of
measurements. That said, things over there are moving along smartly.

>> >
>> > The problem
>> > * All AQMs delay dropping for about one (hard-coded) worst-case RTT,
>> > in case a burst dissipates (allegedly a 'good queue' according to Van
>> > Jacobson)
>>
>>    This assertion is going to need a lot of support.
>>
>>    Bob is a man after my own heart suggesting that an ECN notification
>> may be sent earlier than a packet drop would be indicated. I don't know
>> if we can get there; but IMHO that is essential to getting ECN deployed
>> and used.
>>
>>    I don't think I agree with Bob that what's hard-coded is necessarily
>> a "worst-case" RTT -- and I'm quite sure I'm not willing to make any
>> pronouncement about "all AQMs".
>>
>>    I suggest the talk might be more useful if Bob outlined the AQMs
>> currently in widespread use and detailed _how_ they delay dropping
>> for an estimated RTT.
>
>
> You're right. The 'about' in my sentence was meant to indicate some leewa=
y.
> The specifics depend on each AQM...
>
> The only AQM I know of that doesn't smooth over some nominal RTT is DCTCP
> itself.

DCTCP fits uncomfortably into the aqm catagory.

>
> * CoDel was designed for 'interval' to be a worst-case (largest) RTT, whi=
ch
> it recommends to be set to 100ms. One the queue has exceeded threshold,

Which is 5ms of delay.

> CoDel delays for time 'interval' before starting to signal congestion.

Which it then decreases until it finds something approximating the
RTT. For a system in a reasonably steady state, it will find a decent
value and stay there.

For bursty traffic it is not necessarily helpful, and slow start is
interesting...

>  - I already said how sluggish CoDel would be for a flow with a much shor=
ter

But you've never published a measurement, unless there is one in your new p=
aper?

> RTT than CoDel (others have made this point, e.g. for data centres :
> https://lists.bufferbloat.net/pipermail/codel/2012-August/000448.html).

You might want to read the full thread... a stumbling block on fixing
the srtt was the stochastic hashing... so now there is a full blown
non-approximated fq scheduler that implements pacing.

http://www.ietf.org/mail-archive/web/aqm/current/msg00259.html

And I look forward to a certain upcoming presentation in ICCRG with
great anticipation as to additional fallout from this....

>  - And in the other direction, we already know that utilisation suffers
> fairly badly for flows with RTT significantly larger than 100ms.

"target" and "interval" have always been variables in the fq_codel and
codel codebase and ecn is supported.

tc qdisc add dev your_device root fq_codel target 500us interval 10ms ecn

I don't think the pain of adding that one line of configuration would
affect a data center deployment any. I would certainly like to see
benchmarks in a data center environment. I personally lack 10Gig hw.

I note that once you get below 1ms on a typical box today, you start
hitting other bottlenecks in (for example) the cpu scheduler.

> * PIE suppresses all drops for time max_burst (set to 100ms by default) f=
rom
> when the drop probability it calculates (but doesn't necessarily use) fir=
st
> rises above zero. This is very similar to CoDel, and similar comments are
> applicable.

This too has a variable target value, although there are several other
magic constants that I don't fully understand. I would certainly like
to see some benchmarks in a data center environment. The pie code I
have for linux shoots for a target of 20ms by default, for some
reason.

> * RED requires the constant for its exponentially weighted moving average
> (w_q) to be set taking into account how many packets are likely to arrive=
 at
> the link in a 'typical' RTT. Reverse engineering the values recommended b=
y
> Sally Floyd in the RED paper and in her famous RED parameters Web page
> <http://www.icir.org/floyd/REDparameters.txt>, she recommended a 'typical=
'
> RTT of about 130ms.
>
> [BTW, I know of people who don't calculate w_q, but just use the value of
> "0.002" that Sally recommended for her 45Mb/s link in the original RED pa=
per
> simulations (and repeated at <http://www.icir.org/floyd/REDparameters.txt=
>).
> This was calculated assuming about 500 packets arrive at a link (from all
> flows) in a typical RTT. Links have got a lot faster since 1993.
> Nonetheless, she was considering 45Mb/s for an aggregated link in those
> days, and it happens to be about right for a single user today.]
>
>
>> > * For a flow with 1/10 or 1/100 of this RTT (e.g. from a CDN or your
>> > home media server), any congestion signal is delayed tens or hundreds
>> > of its own RTTs by these AQMs.

Congestion takes time to occur. Consider tcp dynamics as flows also
only increase relative to RTT, as well.

A core question is what do you want response time to congestion to be under=
?

100ms? 5ms? 1ms? 500us?

>>    Clearly, RTTs differing by a factor of ten are quite common at most
>> nodes traversed in a typical path; and it seems _very_ suboptimal to

2.4 billion folk connect to the internet generally at RTTs between 20 and 8=
0ms.

In a data center environment I have no data.

>> have the responsibility for guessing the RTT at the node which must
>> drop packets.

I certainly favor the ongoing development of end to end cc capable of
monitoring the RTT and doing the right thing.

> For packets that do not support ECN, the dropping node has to make a gues=
s
> at the RTT, so as not to drop packets unnecessarily, because drop is an
> impairment as well as a congestion signal. So a transport cannot 'undrop'
> packets.
>
> Our point though is that a network node doesn't have to mimic this behavi=
our
> for ECN packets, because ECN is not an impairment. So a transport can
> un-ECN-mark packets (by smoothing out bursts itself).
>
>
>> > * A TCP flow in slow-start doesn't need the burst smoothed anyway
>> >   - delaying the signal just makes slow-start overshoot more
>> >   - a TCP in slow-start knows that it won't allow the burst to
>> > dissipate anyway

The end of slow start is an ECN notification or a packet drop. What am
I missing here?

>>    A critical point! (It seems obvious to me, but is it obvious to
>> everyone?)

>>
>> > The solution: make ECN also mean "Immediate Congestion Notification"?
>> > * For ECN-capable packets, shift the job of hiding bursts from network
>> > to
>> > host:
>> >   - the network signals ECN with no smoothing delay
>> >   - then the transport can hide bursts of ECN signals from itself
>>
>>    But can we get there from here?
>>
>>    The node doing the ECN notification _can't_ know how the transport
>> will react; and the transport receiving and ECN notification can't know
>> whether the forwarding node has "smoothed" the signal. (It is truly a
>> shame we haven't left any bits for signals like this!)
>
>
> Well, we do have ECT(1) still only assigned experimentally and never used=
,
> which we could decide to use for this immediate ECN. However, first I wan=
t
> to see whether people think it might be feasible to just redefine the
> meaning of CE.

A lot of the NONCE logic has been discussed in other rfcs.

>
> Rationale: So few buffers have ECN support turned on anyway that we shoul=
d
> be able to redefine ECN so that many more will want to turn it on.
>
> For those AQMs that already support ECN, we believe this retrospective
> change will make them only a little worse than they are already (and the
> operator can update them by simple reconfiguration anyway, and is more
> likely to do so, given these are clearly early-adopter networks).
>
>
>> >   - the transport knows
>> >     o whether it's TCP or RTP etc,
>> >     o whether its in congestion avoidance or slow-start,
>> >     o and it knows its RTT,
>> >     o so it can know whether to respond immediately or to smooth the
>> >     signals,
>> >     o and if so, over what time
>>
>>    Yes, but it can't know what smoothing may already have been applied.
>
>
> Yes. If this is a problem, we will have to consider using ECT(1) not CE.
> But it's pretty academic when so few buffers support ECN.
>
> The tiny proportion that do support ECN will already smooth by a 'typical
> RTT' of about 100ms.
>
> If a 20ms RTT flow adds smoothing over its own RTT to this, it will be
> smooth over 120ms.
> The main problem there is not the extra 20ms, it's the original 100ms, wh=
ich
> we won't lose unless we make this change somehow.
>
>
>> >   - then short RTT flows can smooth the signals with only the delay
>> > of their /own/ RTT
>> >     o so they can fill troughs and absorb peaks that longer RTT flows
>> > cannot
>> >   - a TCP only needs to smooth the signals if in congestion avoidance
>> >     o in slow start, it can respond immediately, thus reducing oversho=
ot
>>
>>    This would, IMHO, improve "slow start".
>>
>> > Incremental Deployment:
>> > * Immediate congestion notification doesn't need new AQM implementatio=
n
>> >   - it can use the widely implemented WRED algorithm with an
>> > unexpected configuration
>>
>>    Bob is beginning to lose me here. Does he mean that a forwarding node
>> would apply WRED for both drop and ECN, but with different parameters?
>>
>> > * The network classifies packets for this AQM treatment based on
>> > their ECN-capability
>> >   - Without ECN, it smoothes the queue before signalling drops
>>
>>    Bob has lost me now -- apparently he doesn't mean different
>> parameters... and I don't recognize this "smoothing" step in WRED.
>
>
> I do mean that a forwarding node would apply WRED for both drop and ECN, =
but
> with different parameters.
>
> Each WRED policy-map includes a setting for this smoothing parameter, whi=
ch
> Cisco calls the exponential-weighting-constant. Many people don't notice
> it's there and they just leave it at the default. For instance, Cisco set=
 it
> to
> 2^(-9) ~ 0.002 by default for each of the WRED policy-maps (see
> http://www.cisco.com/en/US/docs/ios/12_0s/feature/guide/fswfq26.html#wp10=
39982).
>
>
>> >   - With ECN, it signals immediately, without any smoothing delay
>> >   - (as today, the operator can still use WRED with the Diffserv field
>> > too)
>>
>>    (Do we need to confuse this discussion by adding diffserv?)
>
>
> A non-Diffserv network still doesn't need to worry about Diffserv.
>
> I put this in parentheses because, if WRED is used today, it is usually u=
sed
> with Diffserv, and I didn't want anyone to worry that they wouldn't be ab=
le
> to continue to do this (e.g. BT use WRED with Diffserv in enterprise
> networks, as do many other carriers).

Perhaps we need a taxonomy so we can all talk to areas of the network
we care about? RED as currently defined will not work correctly on
variable rate networks, like wireless...

I have never thought we'd end up with one aqm or tcp to rule them all..

DC: Data Center
WI: Wireless
FL: Fixed Line

>
>> > * For TCP apps, the stack will use 'DCTCP' (we've tweaked it), if the
>> > ends negotiate ECN with the accurate feedback capability.
>>
>>    Have we settled on "accurate feedback" already? I thought that was
>> still under discussion. (I don't follow exactly what it adds...)
>
>
> See response from Richard Scheffenegger. Essentially the TCPM WG has
> accepted the requirements doc, but not decided between the mechanisms on
> offer.
>
>
>> > * It should 'just work' if an RTP app or a Reno TCP uses ECN.
>>
>>    I don't see any way for a Reno transport using ECN to avoid being
>> starved if ECN arrives earlier (without notice).
>
>
> We haven't tested legacy Reno with ECN yet (we figured legacy Reno withou=
t
> ECN is a lot more prevalent, so focused on this first). Nonetheless,
> Reno-ECN is unlikely to starve, because starvation is about long-running
> behaviour, and once a flow has run for more than a couple of 100ms RTTs, =
the
> immediate ECN signals should be no different from a smoothed ECN. I suspe=
ct
> Reno-ECN might be worse in its short-term dynamics. But remember Reno-ECN=
 is
> likely to be a tiny corner-case.
>
>
>> > The request:
>> > * Much more evaluation to do, but first we want to know:
>> >   - if the idea works, would the IETF have an appetite for tweaking
>> > the definition of ECN so it is merely equivalent to drop in the long
>> > term, but the dynamics need not be equivalent.
>>
>>    There's a good question there; but I don't think we're ready for it.
>
>
> At this stage, even we haven't got many answers. So I'm not asking the IE=
TF
> to answer the question right now. I'm merely saying, /if/ our idea works,=
 is
> there at least an /appetite/ in the IETF for reconsidering the definition=
 of
> ECN?

I wouldn't mind it, but frankly my own concern was addressing the
security issue, not the current definition.

> We wanted to make the IETF aware of this research early, because it might
> want to at least hold off on any actions that would otherwise close off t=
his
> option.


>
> And if we find that any change is completely out of the question, we have=
 to
> try a different tack (e.g. ECT(1)).
>
>
>>    I'd really like to discuss the dynamics of responding more quickly
>> but perhaps less drastically for almost any real-time flow.
>>
>>    But proving "equivalence in the long term" seems too hard.
>
>
> This should be the easy part, because the longer that conditions are stab=
le,
> a smoothed signal should tend towards an unsmoothed signal, all other
> factors being equal.
>
> Equivalence during dynamics is the hard part, and I'm suggesting we don't
> sweat too much about that, as long as the performance evaluations are not
> too far apart.
>
>
>> > Much better than the ECN that didn't get deployed
>> > * This is Explicit and Immediate Congestion Notification (EICN?)
>> >   - same wire protocol, much greater benefits
>> > * The advantage of the original ECN (avoiding congestive loss) was
>> > too small to be worth the deployment hassle
>>
>>    Actually, I don't agree that was the problem -- instead I believe
>> the code has been deployed but administratively suppressed because
>> the operators don't trust the transports. There _is_ a significant
>> improvement from one-RTT reaction instead of several (to detect a
>> drop), but the whole process is just too complicated, while the
>> opportunity for abuse remains obvious.
>
>
> I agree. That's the 'deployment hassle' side of my sentence - the extra
> trust-enhancing mechanisms that seemed necessary were too much pain for t=
he
> small gain.

I

>
>
>> > * Predictable ultra-low latency without loss too (similar to
>> > DCTCP-ECN) would be worth deploying
>>
>>    I'm optimistic that latency will become an easier argument.
>>
>> > * But we all thought DCTCP could only be deployed in isolation (e.g.
>> > data centres)
>> >   - we all thought DCTCP traffic would starve alongside today's TCP
>> > traffic
>> >   - because in a DCTCP queue, the ECN threshold is lower than you
>> > would trigger drop
>> >   - and we thought ECN & drop had to be equivalent.

I'm looking forward to being convinced otherwise.

>>    (I'm not sure we'll succeed at breaking that "equivalence"...)
>>
>> > * We believe we've found a way to ensure DCTCP-ECN traffic doesn't
>> > starve
>> >   - we still make DCTCP-ECN equivalent to drop in the long-run, but
>> > not in its dynamics
>>
>>    (I'm still not sure it's worth arguing the "long-run".)
>
>
> I mean competing long-running ECN & non-ECN flows stabilise at predictabl=
e
> rates, rather than one ratchetting itself down to nothing over time
> (starvation).
>
> That's the primary concern of congestion control 'fairness', before anyon=
e
> starts worrying about what the relative rates are. Given apps get differe=
nt
> relative rates with different RTTs, with different size objects or by
> opening multiple flows, we don't need to sweat so much about precisely eq=
ual
> flow rates; but we must sweat about stable convergence.
>
> Results so far show that the proposed idea is at least very robust agains=
t
> starvation.
>
>
> Bob
>
>
>> --
>> John Leslie <john@jlc.net>
>
>
> ________________________________________________________________
> Bob Briscoe,                                                  BT
> _______________________________________________
> aqm mailing list
> aqm@ietf.org
> https://www.ietf.org/mailman/listinfo/aqm



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html

From bob.briscoe@bt.com  Mon Oct 28 00:17:17 2013
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Dave,

At 04:22 28/10/2013, Dave Taht wrote:
>As is probably well known at this point, I make a clear distinction
>between networking problems in the data center and ones in the wild
>and wooly world.

The two are converging. It will gradually become=20
less sensible to think of the two as separate.


>The data center portion of the universe is a couple hundred meters in
>diameter, the other, I dunno, let's say 8x from here to the moon
>(3,075,200,000m).

DCTCP was fixed ages ago for global RTTs.=20
Download the Linux version from Stanford.


>There are all sorts of things that work and are needed in the data
>center that probably won't work outside it. People run straighter
>cables, do microwave, use pause frames at layer 2, etc, etc, in order
>to wring out the last nanosecond of performance, and certainly running
>without loss is important in that world.

Not relevant. Don't be confused by the name=20
DCTCP. It's only a few lines of code different=20
from TCP, and it's not only relevant in data=20
centres. It's for low queuing delay.

It's only confined to a data centre while we=20
can't work out how it co-exists with current=20
Internet traffic - that's what we're doing.


>On Sun, Oct 27, 2013 at 6:36 PM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> > John, inline...
> >
> > At 12:16 26/10/2013, John Leslie wrote:
> >>
> >> Bob Briscoe <bob.briscoe@bt.com> wrote:
> >> >
> >> > Exec summary
> >> > * Early tests show promise that we may have found a way to make the
>
>I'm failing to get excited lacking paper, testable code, and other proofs.

I'll send the paper, but agreed we've only just=20
got started, so you don't have to be excited yet. A lot more testing to do.


> >> > ultra-low queuing delay of data centre TCP incrementally deployable
> >> > on the public Internet
>
>At the moment the edge of the internet is using things like cable and
>gpon which have 2ms or more of inherent latency built into their
>grant/response structures.

You're saying you're not interested in low=20
queuing delay solutions because there's often=20
fluff in badly designed access networks anyway?=20
That sounds like a poor excuse for not aspiring=20
to lower delay. What excuse is there for the=20
100ms of signalling delay on ECN packets in CoDel=20
then? That's 50 times more than this 2ms=20
(currently) inherent access network delay (that=20
some of us are working to remove as well).

If you put an AQM in a host that is the=20
bottleneck for a transfer to your media server in=20
the next room of your house, at the same time as=20
transfers to the wide area, you really don't want=20
to introduce a nominal global RTT into both feedback loops.


> >> > * For rtcweb, we need to address
> >> >   a) cc for r-t media [rmcat w-g in progress]
> >> >   b) Making TCP nicer
> >> >   c) minimise ability of TCP to bloat queues [AQM w-g now in=
 progress]
> >> >   This addresses b) & c)
>
>I do strongly feel that webrtc needs good aqm and packet scheduling in
>order to succeed, and that the webrtc folk should be testing out what
>has already been developed (red, sfqred, codel, fq_codel, pie) and the
>aqm folk testing the webrtc code.

The AQM for DCTCP was developed 2 years before=20
all of those (except RED). Why decide to build on=20
CoDel in 2012, when DCTCP had already been out=20
there and deployed for 2 years with excellent=20
results? And tcp-rcv-cheat had been out there for 5 years.


>I look forward to some interesting conference calls.
>
>I gave the webrtc codebases a whirl against what aqm and packet
>scheduling techniques we have already this past summer, the initial
>results were encouraging, but I was able to easily crash most of the
>browsers sooner than I could take a good, repeatable set of
>measurements. That said, things over there are moving along smartly.
>
> >> >
> >> > The problem
> >> > * All AQMs delay dropping for about one (hard-coded) worst-case RTT,
> >> > in case a burst dissipates (allegedly a 'good queue' according to Van
> >> > Jacobson)
> >>
> >>    This assertion is going to need a lot of support.
> >>
> >>    Bob is a man after my own heart suggesting that an ECN notification
> >> may be sent earlier than a packet drop would be indicated. I don't know
> >> if we can get there; but IMHO that is essential to getting ECN deployed
> >> and used.
> >>
> >>    I don't think I agree with Bob that what's hard-coded is necessarily
> >> a "worst-case" RTT -- and I'm quite sure I'm not willing to make any
> >> pronouncement about "all AQMs".
> >>
> >>    I suggest the talk might be more useful if Bob outlined the AQMs
> >> currently in widespread use and detailed _how_ they delay dropping
> >> for an estimated RTT.
> >
> >
> > You're right. The 'about' in my sentence was meant to indicate some=
 leeway.
> > The specifics depend on each AQM...
> >
> > The only AQM I know of that doesn't smooth over some nominal RTT is=
 DCTCP
> > itself.
>
>DCTCP fits uncomfortably into the aqm catagory.

Oh dear. Is this Dave Taht saying "I don't like=20
it cos it's different to tradition"?

DCTCP shows just how low you can get queuing=20
delay even with a brain-dead simple step AQM as=20
long as you fix the right problem (smooth in TCP,=20
not in the AQM). Doesn't this tell you something=20
about where the problem probably was?


> >
> > * CoDel was designed for 'interval' to be a worst-case (largest) RTT,=
 which
> > it recommends to be set to 100ms. One the queue has exceeded threshold,
>
>Which is 5ms of delay.

105ms of delay on top of RTT delay before it can possibly do anything.

By far the most common case on an access link is=20
a flow arriving when the link's empty.


> > CoDel delays for time 'interval' before starting to signal congestion.
>
>Which it then decreases until it finds something approximating the
>RTT. For a system in a reasonably steady state, it will find a decent
>value and stay there.

Until the queue empties (which it should if the=20
control system is working), then it has to start=20
over, because another flow may have a different RTT.


>For bursty traffic it is not necessarily helpful, and slow start is
>interesting...
>
> >  - I already said how sluggish CoDel would be=20
> for a flow with a much shorter
>
>But you've never published a measurement, unless=20
>there is one in your new paper?

Lots of experiments in the paper, but not the=20
dynamics yet - we deliberately focused on the=20
long-running behaviour first, to check the critical starvation issues first.


> > RTT than CoDel (others have made this point, e.g. for data centres :
> > https://lists.bufferbloat.net/pipermail/codel/2012-August/000448.html).
>
>You might want to read the full thread... a stumbling block on fixing
>the srtt was the stochastic hashing... so now there is a full blown
>non-approximated fq scheduler that implements pacing.
>
>http://www.ietf.org/mail-archive/web/aqm/current/msg00259.html

As I understand it, this only works for an AQM on=20
a host that has the benefit of being told the RTT from higher up the stack.


>And I look forward to a certain upcoming presentation in ICCRG with
>great anticipation as to additional fallout from this....
>
> >  - And in the other direction, we already know that utilisation suffers
> > fairly badly for flows with RTT significantly larger than 100ms.
>
>"target" and "interval" have always been variables in the fq_codel and
>codel codebase

It's not use having variables you don't know how=20
to set. That was the lesson from RED that DCTCP=20
started to solve (concerning variable RTT), and=20
CoDel continued to solve (concerning line=20
variable rate). However, unfortunately CoDel chose the not invented here=
 route.

>and ecn is supported.
>
>tc qdisc add dev your_device root fq_codel target 500us interval 10ms ecn

AFAICT, ECN is only supported in CoDel in the=20
sense that it is treated as equivalent to drop.=20
The factors where ECN is not equivalent to drop have not been catered for.

And ECN certainly hasn't been considered as=20
something to exploit beyond what drop can do. For=20
instance, to take the approach I'm suggesting,=20
but using CoDel not RED, for ECN-capable packets=20
you would use interval as zero. Then the=20
end-system would smooth out bursts of ECN marks instead of CoDel doing it.


>I don't think the pain of adding that one line of configuration would
>affect a data center deployment any. I would certainly like to see
>benchmarks in a data center environment. I personally lack 10Gig hw.
>
>I note that once you get below 1ms on a typical box today, you start
>hitting other bottlenecks in (for example) the cpu scheduler.

As above, not a good reason. We have to knock=20
down each source of delay, one after the other.


> > * PIE suppresses all drops for time max_burst=20
> (set to 100ms by default) from
> > when the drop probability it calculates (but doesn't necessarily use)=
 first
> > rises above zero. This is very similar to CoDel, and similar comments=
 are
> > applicable.
>
>This too has a variable target value, although there are several other
>magic constants that I don't fully understand.

It takes a few minutes of reading to understand=20
all the PIE variables (about the same complexity=20
as CoDel). I think PIE has the following constants (defaults in []):
* target_del (the target queuing delay [20ms] as you say)
* max_burst (has a similar role to interval in CoDel [100ms])
* deq_threshold (there's enough packets in the=20
queue to measure the line-rate [10KB])
* Tupdate, which determines how often the line=20
rate is tested (which shouldn't affect=20
performance unless the machine can't cope with a high enough update rate).

All the other variables are dependents of these,=20
or autotuned (in the case of alpha and beta).

>I would certainly like
>to see some benchmarks in a data center environment.

Just to be clear, for this conversation I'm not=20
focused on a data centre environment - I'm making=20
DCTCP applicable to the public Internet.

>The pie code I
>have for linux shoots for a target of 20ms by default, for some
>reason.
>
> > * RED requires the constant for its exponentially weighted moving=
 average
> > (w_q) to be set taking into account how many=20
> packets are likely to arrive at
> > the link in a 'typical' RTT. Reverse engineering the values recommended=
 by
> > Sally Floyd in the RED paper and in her famous RED parameters Web page
> > <http://www.icir.org/floyd/REDparameters.txt>, she recommended a=
 'typical'
> > RTT of about 130ms.
> >
> > [BTW, I know of people who don't calculate w_q, but just use the value=
 of
> > "0.002" that Sally recommended for her 45Mb/s=20
> link in the original RED paper
> > simulations (and repeated at=20
> <http://www.icir.org/floyd/REDparameters.txt>).
> > This was calculated assuming about 500 packets arrive at a link (from=
 all
> > flows) in a typical RTT. Links have got a lot faster since 1993.
> > Nonetheless, she was considering 45Mb/s for an aggregated link in those
> > days, and it happens to be about right for a single user today.]
> >
> >
> >> > * For a flow with 1/10 or 1/100 of this RTT (e.g. from a CDN or your
> >> > home media server), any congestion signal is delayed tens or hundreds
> >> > of its own RTTs by these AQMs.
>
>Congestion takes time to occur. Consider tcp dynamics as flows also
>only increase relative to RTT, as well.

Er no. As a queue builds, it causes delay.=20
Regular TCP makes this worse than it has to be by=20
using large saw-teeth, so they become a large=20
inherent variable source of delay. VJ calls this=20
a good queue. But if a variant of TCP can remove=20
this variable source of delay, it can't be good.


>A core question is what do you want response time to congestion to be=
 under?
>
>100ms? 5ms? 1ms? 500us?

Wrong question. Answer depends on RTT. That's the=20
whole point. Consider that you may have been brain-washed.


> >>    Clearly, RTTs differing by a factor of ten are quite common at most
> >> nodes traversed in a typical path; and it seems _very_ suboptimal to
>
>2.4 billion folk connect to the internet=20
>generally at RTTs between 20 and 80ms.

The Internet includes your home network,=20
potentially a cache in your home gateway, I/O=20
lanes within your own machine, etc etc. No-one=20
"connects to the Internet at an RTT of" anything.=20
Today, all these 2.4B folk use connections over=20
the Internet with a very wide range of RTTs.


>In a data center environment I have no data.

Typically 100us or 200us base RTT. And all that=20
cheap data centre kit can be and will be re-used in the public Internet too.


> >> have the responsibility for guessing the RTT at the node which must
> >> drop packets.
>
>I certainly favor the ongoing development of end to end cc capable of
>monitoring the RTT and doing the right thing.

Good.


> > For packets that do not support ECN, the dropping node has to make a=
 guess
> > at the RTT, so as not to drop packets unnecessarily, because drop is an
> > impairment as well as a congestion signal. So a transport cannot=
 'undrop'
> > packets.
> >
> > Our point though is that a network node=20
> doesn't have to mimic this behaviour
> > for ECN packets, because ECN is not an impairment. So a transport can
> > un-ECN-mark packets (by smoothing out bursts itself).
> >
> >
> >> > * A TCP flow in slow-start doesn't need the burst smoothed anyway
> >> >   - delaying the signal just makes slow-start overshoot more
> >> >   - a TCP in slow-start knows that it won't allow the burst to
> >> > dissipate anyway
>
>The end of slow start is an ECN notification or a packet drop. What am
>I missing here?

A traditional AQM (RED, CoDel, PIE) suppresses=20
any signal (ECN or drop) for another 100ms or so=20
after slow start has pushed the queue over the=20
delay threshold. That's on top of the feedback=20
delay of 1RTT, because it doesn't know this is=20
slow-start, so it waits to see if the queue goes away.

The transport knows whether it's in slow-start,=20
so it doesn't need to do any smoothing here - it=20
can drop straight out of slow-start without smoothing.

The transport knows when it's in congestion=20
avoidance too, of course, then it can smooth out=20
bursts of ECN by doing its own EWMA over the=20
actual RTT (that it knows properly).

Ie the AQM can't be selective about when it=20
smooths bursts, because it doesn't know what=20
phase each transport flow is in, but each transport does and therefore can.


> >>    A critical point! (It seems obvious to me, but is it obvious to
> >> everyone?)
>
> >>
> >> > The solution: make ECN also mean "Immediate Congestion Notification"?
> >> > * For ECN-capable packets, shift the job of hiding bursts from=
 network
> >> > to
> >> > host:
> >> >   - the network signals ECN with no smoothing delay
> >> >   - then the transport can hide bursts of ECN signals from itself
> >>
> >>    But can we get there from here?
> >>
> >>    The node doing the ECN notification _can't_ know how the transport
> >> will react; and the transport receiving and ECN notification can't know
> >> whether the forwarding node has "smoothed" the signal. (It is truly a
> >> shame we haven't left any bits for signals like this!)
> >
> >
> > Well, we do have ECT(1) still only assigned experimentally and never=
 used,
> > which we could decide to use for this immediate ECN. However, first I=
 want
> > to see whether people think it might be feasible to just redefine the
> > meaning of CE.
>
>A lot of the NONCE logic has been discussed in other rfcs.
>
> >
> > Rationale: So few buffers have ECN support turned on anyway that we=
 should
> > be able to redefine ECN so that many more will want to turn it on.
> >
> > For those AQMs that already support ECN, we believe this retrospective
> > change will make them only a little worse than they are already (and the
> > operator can update them by simple reconfiguration anyway, and is more
> > likely to do so, given these are clearly early-adopter networks).
> >
> >
> >> >   - the transport knows
> >> >     o whether it's TCP or RTP etc,
> >> >     o whether its in congestion avoidance or slow-start,
> >> >     o and it knows its RTT,
> >> >     o so it can know whether to respond immediately or to smooth the
> >> >     signals,
> >> >     o and if so, over what time
> >>
> >>    Yes, but it can't know what smoothing may already have been applied.
> >
> >
> > Yes. If this is a problem, we will have to consider using ECT(1) not CE.
> > But it's pretty academic when so few buffers support ECN.
> >
> > The tiny proportion that do support ECN will already smooth by a=
 'typical
> > RTT' of about 100ms.
> >
> > If a 20ms RTT flow adds smoothing over its own RTT to this, it will be
> > smooth over 120ms.
> > The main problem there is not the extra 20ms,=20
> it's the original 100ms, which
> > we won't lose unless we make this change somehow.
> >
> >
> >> >   - then short RTT flows can smooth the signals with only the delay
> >> > of their /own/ RTT
> >> >     o so they can fill troughs and absorb peaks that longer RTT flows
> >> > cannot
> >> >   - a TCP only needs to smooth the signals if in congestion avoidance
> >> >     o in slow start, it can respond immediately, thus reducing=
 overshoot
> >>
> >>    This would, IMHO, improve "slow start".
> >>
> >> > Incremental Deployment:
> >> > * Immediate congestion notification doesn't need new AQM=
 implementation
> >> >   - it can use the widely implemented WRED algorithm with an
> >> > unexpected configuration
> >>
> >>    Bob is beginning to lose me here. Does he mean that a forwarding=
 node
> >> would apply WRED for both drop and ECN, but with different parameters?
> >>
> >> > * The network classifies packets for this AQM treatment based on
> >> > their ECN-capability
> >> >   - Without ECN, it smoothes the queue before signalling drops
> >>
> >>    Bob has lost me now -- apparently he doesn't mean different
> >> parameters... and I don't recognize this "smoothing" step in WRED.
> >
> >
> > I do mean that a forwarding node would apply=20
> WRED for both drop and ECN, but
> > with different parameters.
> >
> > Each WRED policy-map includes a setting for this smoothing parameter,=
 which
> > Cisco calls the exponential-weighting-constant. Many people don't notice
> > it's there and they just leave it at the=20
> default. For instance, Cisco set it
> > to
> > 2^(-9) ~ 0.002 by default for each of the WRED policy-maps (see
> >=20
>=
 http://www.cisco.com/en/US/docs/ios/12_0s/feature/guide/fswfq26.html#wp1039=
982).
> >
> >
> >> >   - With ECN, it signals immediately, without any smoothing delay
> >> >   - (as today, the operator can still use WRED with the Diffserv=
 field
> >> > too)
> >>
> >>    (Do we need to confuse this discussion by adding diffserv?)
> >
> >
> > A non-Diffserv network still doesn't need to worry about Diffserv.
> >
> > I put this in parentheses because, if WRED is=20
> used today, it is usually used
> > with Diffserv, and I didn't want anyone to worry that they wouldn't be=
 able
> > to continue to do this (e.g. BT use WRED with Diffserv in enterprise
> > networks, as do many other carriers).
>
>Perhaps we need a taxonomy so we can all talk to areas of the network
>we care about? RED as currently defined will not work correctly on
>variable rate networks, like wireless...
>
>I have never thought we'd end up with one aqm or tcp to rule them all..
>
>DC: Data Center
>WI: Wireless
>FL: Fixed Line

Again, this immediate ECN stuff is for public=20
Internet (not just data centres).


> >
> >> > * For TCP apps, the stack will use 'DCTCP' (we've tweaked it), if the
> >> > ends negotiate ECN with the accurate feedback capability.
> >>
> >>    Have we settled on "accurate feedback" already? I thought that was
> >> still under discussion. (I don't follow exactly what it adds...)
> >
> >
> > See response from Richard Scheffenegger. Essentially the TCPM WG has
> > accepted the requirements doc, but not decided between the mechanisms on
> > offer.
> >
> >
> >> > * It should 'just work' if an RTP app or a Reno TCP uses ECN.
> >>
> >>    I don't see any way for a Reno transport using ECN to avoid being
> >> starved if ECN arrives earlier (without notice).
> >
> >
> > We haven't tested legacy Reno with ECN yet (we figured legacy Reno=
 without
> > ECN is a lot more prevalent, so focused on this first). Nonetheless,
> > Reno-ECN is unlikely to starve, because starvation is about long-running
> > behaviour, and once a flow has run for more=20
> than a couple of 100ms RTTs, the
> > immediate ECN signals should be no different from a smoothed ECN. I=
 suspect
> > Reno-ECN might be worse in its short-term=20
> dynamics. But remember Reno-ECN is
> > likely to be a tiny corner-case.
> >
> >
> >> > The request:
> >> > * Much more evaluation to do, but first we want to know:
> >> >   - if the idea works, would the IETF have an appetite for tweaking
> >> > the definition of ECN so it is merely equivalent to drop in the long
> >> > term, but the dynamics need not be equivalent.
> >>
> >>    There's a good question there; but I don't think we're ready for it.
> >
> >
> > At this stage, even we haven't got many answers. So I'm not asking the=
 IETF
> > to answer the question right now. I'm merely=20
> saying, /if/ our idea works, is
> > there at least an /appetite/ in the IETF for=20
> reconsidering the definition of
> > ECN?
>
>I wouldn't mind it, but frankly my own concern was addressing the
>security issue, not the current definition.

It seems strange that you weren't aware of solutions to this then.


> > We wanted to make the IETF aware of this research early, because it=
 might
> > want to at least hold off on any actions that=20
> would otherwise close off this
> > option.
>
>
> >
> > And if we find that any change is completely=20
> out of the question, we have to
> > try a different tack (e.g. ECT(1)).
> >
> >
> >>    I'd really like to discuss the dynamics of responding more quickly
> >> but perhaps less drastically for almost any real-time flow.
> >>
> >>    But proving "equivalence in the long term" seems too hard.
> >
> >
> > This should be the easy part, because the=20
> longer that conditions are stable,
> > a smoothed signal should tend towards an unsmoothed signal, all other
> > factors being equal.
> >
> > Equivalence during dynamics is the hard part, and I'm suggesting we=
 don't
> > sweat too much about that, as long as the performance evaluations are=
 not
> > too far apart.
> >
> >
> >> > Much better than the ECN that didn't get deployed
> >> > * This is Explicit and Immediate Congestion Notification (EICN?)
> >> >   - same wire protocol, much greater benefits
> >> > * The advantage of the original ECN (avoiding congestive loss) was
> >> > too small to be worth the deployment hassle
> >>
> >>    Actually, I don't agree that was the problem -- instead I believe
> >> the code has been deployed but administratively suppressed because
> >> the operators don't trust the transports. There _is_ a significant
> >> improvement from one-RTT reaction instead of several (to detect a
> >> drop), but the whole process is just too complicated, while the
> >> opportunity for abuse remains obvious.
> >
> >
> > I agree. That's the 'deployment hassle' side of my sentence - the extra
> > trust-enhancing mechanisms that seemed necessary were too much pain for=
 the
> > small gain.
>
>I
>
> >
> >
> >> > * Predictable ultra-low latency without loss too (similar to
> >> > DCTCP-ECN) would be worth deploying
> >>
> >>    I'm optimistic that latency will become an easier argument.
> >>
> >> > * But we all thought DCTCP could only be deployed in isolation (e.g.
> >> > data centres)
> >> >   - we all thought DCTCP traffic would starve alongside today's TCP
> >> > traffic
> >> >   - because in a DCTCP queue, the ECN threshold is lower than you
> >> > would trigger drop
> >> >   - and we thought ECN & drop had to be equivalent.
>
>I'm looking forward to being convinced otherwise.

Yup, I don't want to take up too much of anyone's=20
time with this until we've proved it. But we=20
decided a heads up was important. That's all.


Bob


> >>    (I'm not sure we'll succeed at breaking that "equivalence"...)
> >>
> >> > * We believe we've found a way to ensure DCTCP-ECN traffic doesn't
> >> > starve
> >> >   - we still make DCTCP-ECN equivalent to drop in the long-run, but
> >> > not in its dynamics
> >>
> >>    (I'm still not sure it's worth arguing the "long-run".)
> >
> >
> > I mean competing long-running ECN & non-ECN flows stabilise at=
 predictable
> > rates, rather than one ratchetting itself down to nothing over time
> > (starvation).
> >
> > That's the primary concern of congestion control 'fairness', before=
 anyone
> > starts worrying about what the relative rates are. Given apps get=
 different
> > relative rates with different RTTs, with different size objects or by
> > opening multiple flows, we don't need to=20
> sweat so much about precisely equal
> > flow rates; but we must sweat about stable convergence.
> >
> > Results so far show that the proposed idea is at least very robust=
 against
> > starvation.
> >
> >
> > Bob
> >
> >
> >> --
> >> John Leslie <john@jlc.net>
> >
> >
> > ________________________________________________________________
> > Bob Briscoe,                                                  BT
> > _______________________________________________
> > aqm mailing list
> > aqm@ietf.org
> > https://www.ietf.org/mailman/listinfo/aqm
>
>
>
>--
>Dave T=E4ht
>
>Fixing bufferbloat with cerowrt:=20
>http://www.teklibre.com/cerowrt/subscribe.html

________________________________________________________________
Bob Briscoe,                                                  BT=20


From Ruediger.Geib@telekom.de  Mon Oct 28 00:40:24 2013
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Date: Mon, 28 Oct 2013 08:40:05 +0100
Thread-Topic: [tsvwg] I-D Action: draft-geib-tsvwg-diffserv-intercon-04.txt
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David, Brian,

terminology will be changed to "DSCP Aggregation Prefix" in the next versio=
n.

Thanks and regards,

Ruediger

-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of B=
lack, David
Sent: Wednesday, October 23, 2013 4:17 PM
To: Brian E Carpenter; tsvwg@ietf.org
Subject: Re: [tsvwg] I-D Action: draft-geib-tsvwg-diffserv-intercon-04.txt

Changing Precedence to Aggregation makes sense to me.

Thanks,
--David


> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> Brian E Carpenter
> Sent: Tuesday, October 22, 2013 7:57 PM
> To: tsvwg@ietf.org
> Subject: Re: [tsvwg] I-D Action: draft-geib-tsvwg-diffserv-intercon-04.tx=
t
>
> Hi R=FCdiger,
>
> >    DSCP Precedence Prefix  The bits 0-2 of the DSCP (marked "x" in this
> >            generic DSCP field: xxxddd) are called the DSCP Precedence
> >            Prefix [RFC2474] in the following.  By ignoring the value of
> >            bits 3-6 ( d stands for don' care), a simple aggregation of
> >            PHBs differed by DSCP is possible in IP and MPLS backbones,
> >            but also if Ethernet transport is applied.
>
> Thankyou for providing explicit terminology for this. I am still worried
> that by reviving the word "precedence" we would seriously damage outsider=
s'
> understanding of differentiated services. The specific reason we got rid
> of "precedence" and the notion of simple priority (except for implementin=
g
> the legacy code points) was to underline that precedence doesn't provide
> quality of service; if anything, it provides quality of disservice.
>
> Couldn't you call those three bits "DSCP Aggregation Prefix"? Then they
> would refer to 8 traffic classes, and only in a very primitive
> implementation would they be mapped to priority levels.
>
>     Brian
>


From kivinen@iki.fi  Mon Oct 28 07:05:26 2013
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish	reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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Yoav Nir writes:
>  3. While the TCP stack has access to these ICMP packets and the
>  PMTU that they convey, a userspace IKE daemon usually doesn't. See
>  [1] for how people suggest doing it in C#.

Actually for example our code tries to get ICMP messages for IKE
packets, mostly so that we can make timeout shorter if we start to
receive ICMP messages saying the other end is not going to responded.
If we get ICMP port unreachable for every single IKE packet we send,
there is no point of waiting for 2 minutes and 15 retranmissions to
realize that the other end is not responding, 20 seconds and 5 retries
might be better...

> IKE is supposed to be done in two round-trips. It would be better to
> just pick 576 and send way too many IKE fragments then to try PMTU
> discovery.

Yes. We are talking here about sending one 500-300 byte message from
initiator to responder, and getting about same sized message back.

After that we will only use few hundred bytes long exchanges.

Doing PMTU for sending and receiving one packet is bit overkill.
-- 
kivinen@iki.fi

From touch@isi.edu  Mon Oct 28 09:22:57 2013
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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On 10/27/2013 11:20 PM, Yoav Nir wrote:
> I guess the idea is that if the packet is still bigger than the PMTU,
> then the IKE host gets back a "fragmentation needed" ICMP, and the IKE
> daemon learns of that and resends packets with appropriately small size.
>
> There are some issues with this:

You proceed to describe how PMTUD works, but this is intended (AFAICT) 
to be more like PLMUTD - which does not rely on ICMPs.

Joe


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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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On 10/27/2013 10:41 PM, Valery Smyslov wrote:
>>> Always setting DF bit in this case will probably increase the delay
>>> before IKE SA is set up (as more probes will need to be done).
>>
>> Except that if you continue to allow IP fragmentation, you can't claim
>> your solution is robust to IP fragment poisoning.
>
> I think it is.
>
> Consider the situation when IKE responder is under attack
> via IP fragmentation (no matter which - poisoning attack or memory
> exhausting attack). In any case responder will not be able to reply.
> After some (short) timeout initiator will try to apply IKE Fragmentation.
> Then, if those new messages are not fragmented on the path, they
> will bypass reassembly code on responder and the attack will
> be thwarted. If those messages are fragmented, even with their
> smallest allowed size, then it doesn't matter whether DF bit is set or not.
> If it is set, fragmenting device will drop messages, if it is not set,
> than attack will not be thwarted. Nothing can be done.

You're not proposing to use the smallest size, which would be 68 bytes. 
So by not setting DF, you're enabling a fragmentation attack. If you set 
DF, a receiver could drop all fragments from addresses it wants to 
protect against IKE fragment poisoning.

Joe


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From: Matt Mathis <mattmathis@google.com>
To: Valery Smyslov <svanru@gmail.com>
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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--089e0158a9aa24304c04e9d489eb
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Setting DF is supposed to prevent some midpath IPv4 device from silently
"fixing" a MTU problem by doing you a favor that you don't want.

IPv4 fragmentation can be used in a mode that mimics IPv6: Only fragment on
the originating host, always set DF on on all outbound packets, even if
they are already fragmented.     Any protocol or application that does not
work in this configuration will also fail on IPv6.

Thanks,
--MM--
The best way to predict the future is to create it.  - Alan Kay

Privacy matters!  We know from recent events that people are using our
services to speak in defiance of unjust governments.   We treat privacy and
security as matters of life and death, because for some users, they are.


On Sun, Oct 27, 2013 at 10:50 PM, Valery Smyslov <svanru@gmail.com> wrote:

> **
> Hi Matt,
>
> the whole idea of the draft is avoiding IP fragmentation for IKE when
> it prevents IKE to work. What about DF bit - I don't see how setting it
> would help IKE to work.
>
> Regards,
> Valery.
>
>
> ----- Original Message -----
> *From:* Matt Mathis <mattmathis@google.com>
> *To:* Valery Smyslov <svanru@gmail.com>
> *Cc:* tsvwg <tsvwg@ietf.org> ; tsv-ads@tools.ietf.org ; ipsec@ietf.org ;
> draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org ; tsv-dir@ietf.org; Joe
> Touch <touch@isi.edu>
> *Sent:* Saturday, October 26, 2013 12:41 AM
> *Subject:* Re: [tsvwg] [IPsec] TSVDIR-ish
> reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
>
> I concur with Joe: once you have enough machinery work well with IPv6
> fragmentation semantics, you should use it for IPv4 too, and
> unconditionally set DF.   This probably applies to *all* protocols.
>
> IPv4 reassembly is hopelessly out of scale.  IP ID wrap times are likely
> to be sub second for any large CGN connecting to any large service.....
>  They might even be shorter than the queuing times.
>
> I suspect that if you re-review decade old papers on fragmentation, you
> will find some scale assumptions that are no longer correct.  In that time
> the Internet has moved at least another two orders of magnitude in packet
> rates.
>
>  Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>
> Privacy matters!  We know from recent events that people are using our
> services to speak in defiance of unjust governments.   We treat privacy and
> security as matters of life and death, because for some users, they are.
>
>
> On Fri, Oct 25, 2013 at 1:18 PM, Joe Touch <touch@isi.edu> wrote:
>
>>
>>
>> On 10/24/2013 10:45 PM, Valery Smyslov wrote:
>> ...
>>
>>>  You're using existing IKE messages *and* existing timeouts to
>>>> determine when there is a problem. A separate timer would be useful,
>>>> if only to allow you to decouple fragment retransmission from IKE
>>>> transaction retries.
>>>>
>>>
>>> No, the timeouts are different. I should have made it more expplicit in
>>> the draft.
>>>
>>
>> That'd be useful.
>>
>> ...
>>
>>> Always setting DF bit in this case will probably increase the delay
>>> before IKE SA is set up (as more probes will need to be done).
>>>
>>
>> Except that if you continue to allow IP fragmentation, you can't claim
>> your solution is robust to IP fragment poisoning.
>>
>>   Note, that this approach is in line with advices, given for IKE in the
>>>>> paper
>>>>>
>>>>> C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
>>>>>               for UDP-based protocols", ACM Conference on Computer and
>>>>>               Communications Security, October 2003.
>>>>>
>>>>
>>>> That paper doesn't consider IKE-level fragmentation, which you're
>>>> introducing. I agree that DF=0 for IKE without IKE-level fragmentation.
>>>>
>>>
>>> It does, in Section 3.3.
>>>
>>
>> Sorry - I missed that. But that section also gives good reasons why this
>> is a bad idea in IKE too.
>>
>> Joe
>>
>
>

--089e0158a9aa24304c04e9d489eb
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Setting DF is supposed to prevent some midpath IPv4 device=
 from silently &quot;fixing&quot; a MTU problem by doing you a favor that y=
ou don&#39;t want.<div><br></div><div>IPv4 fragmentation can be used in a m=
ode that mimics IPv6: Only fragment on the originating host, always set DF =
on on all outbound packets, even if they are already fragmented. =A0 =A0 An=
y protocol or application that does not work in this configuration will als=
o fail on IPv6.</div>
</div><div class=3D"gmail_extra"><br clear=3D"all"><div><div dir=3D"ltr"><d=
iv>Thanks,</div>--MM--<br>The best way to predict the future is to create i=
t. =A0- Alan Kay<br><br>Privacy matters! =A0We know from recent events that=
 people are using our services to speak in defiance of unjust governments. =
=A0 We treat privacy and security as matters of life and death, because for=
 some users, they are.</div>
</div>
<br><br><div class=3D"gmail_quote">On Sun, Oct 27, 2013 at 10:50 PM, Valery=
 Smyslov <span dir=3D"ltr">&lt;<a href=3D"mailto:svanru@gmail.com" target=
=3D"_blank">svanru@gmail.com</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">
<u></u>





<div bgcolor=3D"#ffffff">
<div><font>Hi Matt,</font></div>
<div><font></font>=A0</div>
<div><font>the whole idea of the draft is avoiding IP fragmentation for=20
IKE when</font></div>
<div><font>it prevents IKE to work. </font><font>What about DF bit=20
- I don&#39;t=A0see how setting it </font></div>
<div><font>would help IKE to work.</font></div>
<div><font></font>=A0</div>
<div><font>Regards,</font></div>
<div><font>Valery.</font></div><div><div class=3D"h5">
<div><font></font>=A0</div>
<blockquote style=3D"BORDER-LEFT:#000000 2px solid;PADDING-LEFT:5px;PADDING=
-RIGHT:0px;MARGIN-LEFT:5px;MARGIN-RIGHT:0px" dir=3D"ltr">
  <div style=3D"FONT:10pt arial">----- Original Message ----- </div>
  <div style=3D"FONT:10pt arial;BACKGROUND:#e4e4e4"><b>From:</b>=20
  <a title=3D"mattmathis@google.com" href=3D"mailto:mattmathis@google.com" =
target=3D"_blank">Matt=20
  Mathis</a> </div>
  <div style=3D"FONT:10pt arial"><b>To:</b> <a title=3D"svanru@gmail.com" h=
ref=3D"mailto:svanru@gmail.com" target=3D"_blank">Valery Smyslov</a> </div>
  <div style=3D"FONT:10pt arial"><b>Cc:</b> <a title=3D"tsvwg@ietf.org" hre=
f=3D"mailto:tsvwg@ietf.org" target=3D"_blank">tsvwg</a> ; <a title=3D"tsv-a=
ds@tools.ietf.org" href=3D"mailto:tsv-ads@tools.ietf.org" target=3D"_blank"=
>tsv-ads@tools.ietf.org</a> ; <a title=3D"ipsec@ietf.org" href=3D"mailto:ip=
sec@ietf.org" target=3D"_blank">ipsec@ietf.org</a> ; <a title=3D"draft-ietf=
-ipsecme-ikev2-fragmentation@tools.ietf.org" href=3D"mailto:draft-ietf-ipse=
cme-ikev2-fragmentation@tools.ietf.org" target=3D"_blank">draft-ietf-ipsecm=
e-ikev2-fragmentation@tools.ietf.org</a>=20
  ; <a title=3D"tsv-dir@ietf.org" href=3D"mailto:tsv-dir@ietf.org" target=
=3D"_blank">tsv-dir@ietf.org</a> ; <a title=3D"touch@isi.edu" href=3D"mailt=
o:touch@isi.edu" target=3D"_blank">Joe Touch</a> </div>
  <div style=3D"FONT:10pt arial"><b>Sent:</b> Saturday, October 26, 2013 12=
:41=20
  AM</div>
  <div style=3D"FONT:10pt arial"><b>Subject:</b> Re: [tsvwg] [IPsec] TSVDIR=
-ish=20
  reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04</div>
  <div><br></div>
  <div dir=3D"ltr">I concur with Joe: once you have enough machinery work w=
ell with=20
  IPv6 fragmentation semantics, you should use it for IPv4 too, and=20
  unconditionally set DF. =A0 This probably applies to *all* protocols.
  <div><br></div>
  <div>IPv4 reassembly is hopelessly out of scale. =A0IP ID wrap times are=
=20
  likely to be sub second for any large CGN connecting to any large service=
.....=20
  =A0They might even be shorter than the queuing times.
  <div><br></div>
  <div>I suspect that if you re-review decade old papers on fragmentation, =
you=20
  will find some scale assumptions that are no longer correct. =A0In that=
=20
  time the Internet has moved at least another two orders of magnitude in p=
acket=20
  rates.</div></div></div>
  <div class=3D"gmail_extra"><br clear=3D"all">
  <div>
  <div dir=3D"ltr">
  <div>Thanks,</div>--MM--<br>The best way to predict the future is to crea=
te=20
  it. =A0- Alan Kay<br><br>Privacy matters! =A0We know from recent events=
=20
  that people are using our services to speak in defiance of unjust governm=
ents.=20
  =A0 We treat privacy and security as matters of life and death, because f=
or=20
  some users, they are.</div></div><br><br>
  <div class=3D"gmail_quote">On Fri, Oct 25, 2013 at 1:18 PM, Joe Touch <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:touch@isi.edu" target=3D"_blank">touch=
@isi.edu</a>&gt;</span> wrote:<br>
  <blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;=
PADDING-LEFT:1ex" class=3D"gmail_quote"><br><br>On 10/24/2013 10:45 PM, Val=
ery Smyslov=20
    wrote:<br>...<br>
    <blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8e=
x;PADDING-LEFT:1ex" class=3D"gmail_quote">
      <div>
      <blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.=
8ex;PADDING-LEFT:1ex" class=3D"gmail_quote">You&#39;re using existing IKE m=
essages *and* existing=20
        timeouts to<br>determine when there is a problem. A separate timer =
would=20
        be useful,<br>if only to allow you to decouple fragment retransmiss=
ion=20
        from IKE<br>transaction retries.<br></blockquote><br></div>No, the=
=20
      timeouts are different. I should have made it more expplicit in<br>th=
e=20
      draft.<br></blockquote><br>That&#39;d be useful.<br><br>...<br>
    <blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8e=
x;PADDING-LEFT:1ex" class=3D"gmail_quote">Always setting DF bit in this cas=
e will probably=20
      increase the delay<br>before IKE SA is set up (as more probes will ne=
ed to=20
      be done).<br></blockquote><br>Except that if you continue to allow IP=
=20
    fragmentation, you can&#39;t claim your solution is robust to IP fragme=
nt=20
    poisoning.<br><br>
    <blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8e=
x;PADDING-LEFT:1ex" class=3D"gmail_quote">
      <div>
      <blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.=
8ex;PADDING-LEFT:1ex" class=3D"gmail_quote">
        <blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px =
0.8ex;PADDING-LEFT:1ex" class=3D"gmail_quote">Note, that this approach is i=
n line with advices,=20
          given for IKE in the<br>paper<br><br>C. Kaufman, R. Perlman, and =
B.=20
          Sommerfeld, &quot;DoS protection<br>=A0 =A0 =A0 =A0 =A0=20
          =A0 =A0 for UDP-based protocols&quot;, ACM Conference on Computer=
=20
          and<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 Communications=20
          Security, October 2003.<br></blockquote><br>That paper doesn&#39;=
t consider=20
        IKE-level fragmentation, which you&#39;re<br>introducing. I agree t=
hat DF=3D0=20
        for IKE without IKE-level fragmentation.<br></blockquote><br></div>=
It=20
      does, in Section 3.3.<br></blockquote><br>Sorry - I missed that. But =
that=20
    section also gives good reasons why this is a bad idea in IKE too.<span=
><font color=3D"#888888"><br><br>Joe<br></font></span></blockquote></div><b=
r></div></blockquote></div></div></div>
</blockquote></div><br></div>

--089e0158a9aa24304c04e9d489eb--

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Changed milestone "Submit 'SCTP SACK Immediately' to IESG for
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URL: http://datatracker.ietf.org/wg/tsvwg/charter/

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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish review ofdraft-ietf-ipsecme-ikev2-fragmentation-04
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Hi Paul,

Sorry for the late reply.

You seem to assume that SPIs are random. This is not mandated by IKEv2 
and in practice it is usually, but not always, the case. See 
http://www.ietf.org/mail-archive/web/ipsec/current/msg06563.html

Thanks,
	Yaron

On 2013-10-22 18:35, Paul Wouters wrote:
> On Tue, 22 Oct 2013, Valery Smyslov wrote:
>
>>> attacks (e.g., overloading the receiver with false fragments).
>>
>> This attack is always possible both with and without fragmentation
>> (just overloading with false full messages) and IKE is designed to
>> withstand it.
>
> Partially. For unfragmented IKE, the cookies/SPI mitigate this if
> you give preference to IKE messages further in the processing queue.
> With this draft, I'm forced to do more crypto on IKE packets before I
> can decide if these need prioritization or not (with the exception of
> the first packet, which cannot get fragmented anyway)
>
>> Handling fragmentation in IKE doesn't make it more vulnerable to this
>> attack.
>
> Handling encrypted fragments does, because you will spend more time
> doing the crypto. Although this still happens after the cookie/spi
> verification. You are doing an additional round of crypto on each fragment
> in addition to the regular IKE crypto you need to perform on the assembled
> packet. Fragments could be made artificially long and the IKE packet
> artificial big, resulting in much more crypto. Now I think this might
> not matter much, because the attacker needs to be local and they will
> only be local to a remote endpoint which has enough CPU, not the ipsec
> core gateway that actually might run out of CPU power when under attack.
>
>> On the other hand, handling fragmentation outside IKE (on IP level)
>> or using unauthenticated fragments in IKE, makes it susceptible to an
>> attack,
>> described in the rationale, i.e. when attacker spending virtually zero
>> resources
>> can stop IKE communication by infrequently injecting bad fragments,
>> which results in "poisoning" of reassembling queue.
>
> But the attack is theoretical and of no practical value.
>
> 1) on-path attacker (remote)
>
> They can just DOS you regardless by withholding packets.
>
> 2) off-path attacker (remote)
>
> These attackers don't know the cookies/spi's so they can't really sent
> a forged fragment you would use for assembly - protected with crypto or
> not. It is also impossible for them to time their attack so that one of
> their fragments intersect with one of your fragments so they would need
> to start sending fragments before the endpoint does but using mostly
> wrong cookies/spis and would swamp an endpoint with easilly detectable
> forged packets.
>
> 3) shared medium like wifi hotspot (local)
>
> These are the only attackers with enough knowledge to sent sufficiently
> advanced bad fragments interfering with incoming fragments. They can
> already do all the damage they want without resorting to IKE fragment
> spoofing.  They can arp spoof the gateway, sent a Wifi de-association
> request, sent enough garbage to fill the air waves, and interfere with
> the IKE in such a way that you think the remote peer does not support
> fragmentation at all. So this draft protects against a local attacker
> that wants to masquerade as a network failure and only prevent you from
> doing IKE, hoping you will fall back to plaintext traffic. Software
> could notify the user that they seem to be under attack, which IMHO
> would be more valuable to the user than quiet resilience against the
> attack.
>
>> Using authenticated fragments prevents this kind of attack.
>
> Yes, it prevents one of the low hanging fruits. Which I don't think is
> worth the complexity of this draft compared to the existing
> fragmentation code out in the wild.
>
>> But using IKE fragmentation in this case might eliminate this
>> vulnerability.
>
> At the price of adding more complexity to IKE. With the attacker not
> stopped from their goal.
>
>>> Additionally, regarding PMTUD, this should be done using PLMTUD (RFC
>>> 4821), and should be described as such.
>>
>> IMHO, PLMTUD doesn't suite well for IKE. First, IKE is UDP-based
>> and RFC4821 has many restrictions when used with connectionless
>> datagram protocols, some of which are often difficult to satisfy in
>> environments,
>> where IKE is used. Second, the goal of PLMTUD is to find _maximum_
>> MTU size that is not fragmented by IP, in other words, to maximize
>> throughput. For IKE the goal of PMTUD is to find _any_reasonable_
>> MTU size that won't be fragmented by IP. Throughput is not important.
>> Third, PMTU discovery in the draft is completely optional and one may
>> use any other mechanisms (including PLMTUD, if it is available)
>> to obtain MTU information.
>
> I agree. There is going to be like 2 to 4 fragments. Size does not
> matter. It's better to cut down on latency/RTT by using an MTU size
> that is for sure to work on the first go.
>
> But that's also why this complexity is not worth it. Fragmentation does
> only mean going from 1 IKE packet to 2-4 IKE packets. The timeframe
> to mess with these are in the 5ms range. If an attacker has that
> accuracy against you, you've lost already regardless. I doubt the
> attacker can even pull it off on wifi.
>
> The original IKE fragmentation is really nice and simple. Copy part of
> the original IKE header, add a single payload with fragment payload
> type and fragment number/max, sent fragments in a burst.
>
> This draft looks like it is fixing a security issue, while it really
> should be just about sending smaller packets as similar as possible
> to the original single packet. Messing with one IKE packet versus 2-4
> IKE packets is not a real problem that I think needs fixing.
>
> Paul
> _______________________________________________
> IPsec mailing list
> IPsec@ietf.org
> https://www.ietf.org/mailman/listinfo/ipsec

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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish	reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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I guess the idea is that if the packet is still bigger than the PMTU, then =
the IKE host gets back a "fragmentation needed" ICMP, and the IKE daemon le=
arns of that and resends packets with appropriately small size.

There are some issues with this:
 1. It requires yet another retransmission
 2. There's a lot of things on the IPv4 Internet that drop ICMP messages. I=
Pv6 simply doesn't work without them, but IPv4 usually manages.
 3. While the TCP stack has access to these ICMP packets and the PMTU that =
they convey, a userspace IKE daemon usually doesn't. See [1] for how people=
 suggest doing it in C#.

IKE is supposed to be done in two round-trips. It would be better to just p=
ick 576 and send way too many IKE fragments then to try PMTU discovery.

Yoav

[1] http://stackoverflow.com/questions/4142080/path-mtu-discovery-in-c-shar=
p

On Oct 28, 2013, at 7:50 AM, Valery Smyslov <svanru@gmail.com<mailto:svanru=
@gmail.com>> wrote:

Hi Matt,

the whole idea of the draft is avoiding IP fragmentation for IKE when
it prevents IKE to work. What about DF bit - I don't see how setting it
would help IKE to work.

Regards,
Valery.

----- Original Message -----
From: Matt Mathis<mailto:mattmathis@google.com>
To: Valery Smyslov<mailto:svanru@gmail.com>
Cc: tsvwg<mailto:tsvwg@ietf.org> ; tsv-ads@tools.ietf.org<mailto:tsv-ads@to=
ols.ietf.org> ; ipsec@ietf.org<mailto:ipsec@ietf.org> ; draft-ietf-ipsecme-=
ikev2-fragmentation@tools.ietf.org<mailto:draft-ietf-ipsecme-ikev2-fragment=
ation@tools.ietf.org> ; tsv-dir@ietf.org<mailto:tsv-dir@ietf.org> ; Joe Tou=
ch<mailto:touch@isi.edu>
Sent: Saturday, October 26, 2013 12:41 AM
Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fr=
agmentation-04

I concur with Joe: once you have enough machinery work well with IPv6 fragm=
entation semantics, you should use it for IPv4 too, and unconditionally set=
 DF.   This probably applies to *all* protocols.

IPv4 reassembly is hopelessly out of scale.  IP ID wrap times are likely to=
 be sub second for any large CGN connecting to any large service.....  They=
 might even be shorter than the queuing times.

I suspect that if you re-review decade old papers on fragmentation, you wil=
l find some scale assumptions that are no longer correct.  In that time the=
 Internet has moved at least another two orders of magnitude in packet rate=
s.

Thanks,
--MM--
The best way to predict the future is to create it.  - Alan Kay

Privacy matters!  We know from recent events that people are using our serv=
ices to speak in defiance of unjust governments.   We treat privacy and sec=
urity as matters of life and death, because for some users, they are.


On Fri, Oct 25, 2013 at 1:18 PM, Joe Touch <touch@isi.edu<mailto:touch@isi.=
edu>> wrote:


On 10/24/2013 10:45 PM, Valery Smyslov wrote:
...
You're using existing IKE messages *and* existing timeouts to
determine when there is a problem. A separate timer would be useful,
if only to allow you to decouple fragment retransmission from IKE
transaction retries.

No, the timeouts are different. I should have made it more expplicit in
the draft.

That'd be useful.

...
Always setting DF bit in this case will probably increase the delay
before IKE SA is set up (as more probes will need to be done).

Except that if you continue to allow IP fragmentation, you can't claim your=
 solution is robust to IP fragment poisoning.

Note, that this approach is in line with advices, given for IKE in the
paper

C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
              for UDP-based protocols", ACM Conference on Computer and
              Communications Security, October 2003.

That paper doesn't consider IKE-level fragmentation, which you're
introducing. I agree that DF=3D0 for IKE without IKE-level fragmentation.

It does, in Section 3.3.

Sorry - I missed that. But that section also gives good reasons why this is=
 a bad idea in IKE too.

Joe

_______________________________________________
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IPsec@ietf.org<mailto:IPsec@ietf.org>
https://www.ietf.org/mailman/listinfo/ipsec


--_000_0E72C76582CF4A2CBE6A31735954A9BBcheckpointcom_
Content-Type: text/html; charset="iso-8859-1"
Content-ID: <0F175F27AD02A84080DA1735FD1304D7@ad.checkpoint.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<base href=3D"x-msg://657/">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
I guess the idea is that if the packet is still bigger than the PMTU, then =
the IKE host gets back a &quot;fragmentation needed&quot; ICMP, and the IKE=
 daemon learns of that and resends packets with appropriately small size.
<div><br>
</div>
<div>There are some issues with this:</div>
<div>&nbsp;1. It requires yet another retransmission</div>
<div>&nbsp;2. There's a lot of things on the IPv4 Internet that drop ICMP m=
essages. IPv6 simply doesn't work without them, but IPv4 usually manages.</=
div>
<div>&nbsp;3. While the TCP stack has access to these ICMP packets and the =
PMTU that they convey, a userspace IKE daemon usually doesn't. See [1] for =
how people suggest doing it in C#.</div>
<div><br>
</div>
<div>IKE is supposed to be done in two round-trips. It would be better to j=
ust pick 576 and send way too many IKE fragments then to try PMTU discovery=
.</div>
<div><br>
</div>
<div>Yoav</div>
<div><br>
</div>
<div>[1]&nbsp;<a href=3D"http://stackoverflow.com/questions/4142080/path-mt=
u-discovery-in-c-sharp">http://stackoverflow.com/questions/4142080/path-mtu=
-discovery-in-c-sharp</a></div>
<div><br>
<div>
<div>On Oct 28, 2013, at 7:50 AM, Valery Smyslov &lt;<a href=3D"mailto:svan=
ru@gmail.com">svanru@gmail.com</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div bgcolor=3D"#ffffff" style=3D"font-family: Tahoma; font-size: medium; f=
ont-style: normal; font-variant: normal; font-weight: normal; letter-spacin=
g: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-=
indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spa=
cing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
">
<div><font size=3D"2">Hi Matt,</font></div>
<div><font size=3D"2"></font>&nbsp;</div>
<div><font size=3D"2">the whole idea of the draft is avoiding IP fragmentat=
ion for IKE when</font></div>
<div><font size=3D"2">it prevents IKE to work.<span class=3D"Apple-converte=
d-space">&nbsp;</span></font><font size=3D"2">What about DF bit - I don't&n=
bsp;see how setting it</font></div>
<div><font size=3D"2">would help IKE to work.</font></div>
<div><font size=3D"2"></font>&nbsp;</div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">Valery.</font></div>
<div><font size=3D"2"></font>&nbsp;</div>
<blockquote dir=3D"ltr" style=3D"border-left-color: rgb(0, 0, 0); border-le=
ft-width: 2px; border-left-style: solid; padding-left: 5px; padding-right: =
0px; margin-left: 5px; margin-right: 0px; ">
<div style=3D"font-style: normal; font-variant: normal; font-weight: normal=
; font-size: 10pt; line-height: normal; font-family: arial; ">
----- Original Message -----</div>
<div style=3D"font-style: normal; font-variant: normal; font-weight: normal=
; font-size: 10pt; line-height: normal; font-family: arial; background-colo=
r: rgb(228, 228, 228); background-position: initial initial; background-rep=
eat: initial initial; ">
<b>From:</b><span class=3D"Apple-converted-space">&nbsp;</span><a title=3D"=
mattmathis@google.com" href=3D"mailto:mattmathis@google.com">Matt Mathis</a=
></div>
<div style=3D"font-style: normal; font-variant: normal; font-weight: normal=
; font-size: 10pt; line-height: normal; font-family: arial; ">
<b>To:</b><span class=3D"Apple-converted-space">&nbsp;</span><a title=3D"sv=
anru@gmail.com" href=3D"mailto:svanru@gmail.com">Valery Smyslov</a></div>
<div style=3D"font-style: normal; font-variant: normal; font-weight: normal=
; font-size: 10pt; line-height: normal; font-family: arial; ">
<b>Cc:</b><span class=3D"Apple-converted-space">&nbsp;</span><a title=3D"ts=
vwg@ietf.org" href=3D"mailto:tsvwg@ietf.org">tsvwg</a><span class=3D"Apple-=
converted-space">&nbsp;</span>;<span class=3D"Apple-converted-space">&nbsp;=
</span><a title=3D"tsv-ads@tools.ietf.org" href=3D"mailto:tsv-ads@tools.iet=
f.org">tsv-ads@tools.ietf.org</a><span class=3D"Apple-converted-space">&nbs=
p;</span>;<span class=3D"Apple-converted-space">&nbsp;</span><a title=3D"ip=
sec@ietf.org" href=3D"mailto:ipsec@ietf.org">ipsec@ietf.org</a><span class=
=3D"Apple-converted-space">&nbsp;</span>;<span class=3D"Apple-converted-spa=
ce">&nbsp;</span><a title=3D"draft-ietf-ipsecme-ikev2-fragmentation@tools.i=
etf.org" href=3D"mailto:draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.o=
rg">draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org</a><span class=3D=
"Apple-converted-space">&nbsp;</span>;<span class=3D"Apple-converted-space"=
>&nbsp;</span><a title=3D"tsv-dir@ietf.org" href=3D"mailto:tsv-dir@ietf.org=
">tsv-dir@ietf.org</a><span class=3D"Apple-converted-space">&nbsp;</span>;<=
span class=3D"Apple-converted-space">&nbsp;</span><a title=3D"touch@isi.edu=
" href=3D"mailto:touch@isi.edu">Joe
 Touch</a></div>
<div style=3D"font-style: normal; font-variant: normal; font-weight: normal=
; font-size: 10pt; line-height: normal; font-family: arial; ">
<b>Sent:</b><span class=3D"Apple-converted-space">&nbsp;</span>Saturday, Oc=
tober 26, 2013 12:41 AM</div>
<div style=3D"font-style: normal; font-variant: normal; font-weight: normal=
; font-size: 10pt; line-height: normal; font-family: arial; ">
<b>Subject:</b><span class=3D"Apple-converted-space">&nbsp;</span>Re: [tsvw=
g] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04</di=
v>
<div><br>
</div>
<div dir=3D"ltr">I concur with Joe: once you have enough machinery work wel=
l with IPv6 fragmentation semantics, you should use it for IPv4 too, and un=
conditionally set DF. &nbsp; This probably applies to *all* protocols.
<div><br>
</div>
<div>IPv4 reassembly is hopelessly out of scale. &nbsp;IP ID wrap times are=
 likely to be sub second for any large CGN connecting to any large service.=
.... &nbsp;They might even be shorter than the queuing times.
<div><br>
</div>
<div>I suspect that if you re-review decade old papers on fragmentation, yo=
u will find some scale assumptions that are no longer correct. &nbsp;In tha=
t time the Internet has moved at least another two orders of magnitude in p=
acket rates.</div>
</div>
</div>
<div class=3D"gmail_extra"><br clear=3D"all">
<div>
<div dir=3D"ltr">
<div>Thanks,</div>
--MM--<br>
The best way to predict the future is to create it. &nbsp;- Alan Kay<br>
<br>
Privacy matters! &nbsp;We know from recent events that people are using our=
 services to speak in defiance of unjust governments. &nbsp; We treat priva=
cy and security as matters of life and death, because for some users, they =
are.</div>
</div>
<br>
<br>
<div class=3D"gmail_quote">On Fri, Oct 25, 2013 at 1:18 PM, Joe Touch<span =
class=3D"Apple-converted-space">&nbsp;</span><span dir=3D"ltr">&lt;<a href=
=3D"mailto:touch@isi.edu" target=3D"_blank">touch@isi.edu</a>&gt;</span><sp=
an class=3D"Apple-converted-space">&nbsp;</span>wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left-color: rgb(204, 204,=
 204); border-left-width: 1px; border-left-style: solid; margin: 0px 0px 0p=
x 0.8ex; padding-left: 1ex; ">
<br>
<br>
On 10/24/2013 10:45 PM, Valery Smyslov wrote:<br>
...<br>
<blockquote class=3D"gmail_quote" style=3D"border-left-color: rgb(204, 204,=
 204); border-left-width: 1px; border-left-style: solid; margin: 0px 0px 0p=
x 0.8ex; padding-left: 1ex; ">
<div class=3D"im">
<blockquote class=3D"gmail_quote" style=3D"border-left-color: rgb(204, 204,=
 204); border-left-width: 1px; border-left-style: solid; margin: 0px 0px 0p=
x 0.8ex; padding-left: 1ex; ">
You're using existing IKE messages *and* existing timeouts to<br>
determine when there is a problem. A separate timer would be useful,<br>
if only to allow you to decouple fragment retransmission from IKE<br>
transaction retries.<br>
</blockquote>
<br>
</div>
No, the timeouts are different. I should have made it more expplicit in<br>
the draft.<br>
</blockquote>
<br>
That'd be useful.<br>
<br>
...<br>
<blockquote class=3D"gmail_quote" style=3D"border-left-color: rgb(204, 204,=
 204); border-left-width: 1px; border-left-style: solid; margin: 0px 0px 0p=
x 0.8ex; padding-left: 1ex; ">
Always setting DF bit in this case will probably increase the delay<br>
before IKE SA is set up (as more probes will need to be done).<br>
</blockquote>
<br>
Except that if you continue to allow IP fragmentation, you can't claim your=
 solution is robust to IP fragment poisoning.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left-color: rgb(204, 204,=
 204); border-left-width: 1px; border-left-style: solid; margin: 0px 0px 0p=
x 0.8ex; padding-left: 1ex; ">
<div class=3D"im">
<blockquote class=3D"gmail_quote" style=3D"border-left-color: rgb(204, 204,=
 204); border-left-width: 1px; border-left-style: solid; margin: 0px 0px 0p=
x 0.8ex; padding-left: 1ex; ">
<blockquote class=3D"gmail_quote" style=3D"border-left-color: rgb(204, 204,=
 204); border-left-width: 1px; border-left-style: solid; margin: 0px 0px 0p=
x 0.8ex; padding-left: 1ex; ">
Note, that this approach is in line with advices, given for IKE in the<br>
paper<br>
<br>
C. Kaufman, R. Perlman, and B. Sommerfeld, &quot;DoS protection<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; for UDP-based protocols&qu=
ot;, ACM Conference on Computer and<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Communications Security, O=
ctober 2003.<br>
</blockquote>
<br>
That paper doesn't consider IKE-level fragmentation, which you're<br>
introducing. I agree that DF=3D0 for IKE without IKE-level fragmentation.<b=
r>
</blockquote>
<br>
</div>
It does, in Section 3.3.<br>
</blockquote>
<br>
Sorry - I missed that. But that section also gives good reasons why this is=
 a bad idea in IKE too.<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
Joe<br>
</font></span></blockquote>
</div>
<br>
</div>
</blockquote>
_______________________________________________<br>
IPsec mailing list<br>
<a href=3D"mailto:IPsec@ietf.org">IPsec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/ipsec">https://www.ietf.or=
g/mailman/listinfo/ipsec</a><br>
</div>
</blockquote>
</div>
<br>
</div>
</body>
</html>

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Thread-Topic: [IPsec] [tsvwg]	TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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OK, you convinced me. 576 is the right value to recommend for initiator. No=
t sure if for responder it's also preferable to use 576, or to go with the =
idea in 2.5.1 of using the fragment size from the request.

Yoav

-----Original Message-----
From: Tero Kivinen [mailto:kivinen@iki.fi]=20
Sent: Monday, October 28, 2013 4:05 PM
To: Yoav Nir
Cc: Valery Smyslov; tsvwg; Joe Touch; <tsv-ads@tools.ietf.org>; <ipsec@ietf=
.org>; <draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org>; Matt Mathis=
; <tsv-dir@ietf.org>
Subject: Re: [IPsec] [tsvwg] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fr=
agmentation-04

Yoav Nir writes:
>  3. While the TCP stack has access to these ICMP packets and the  PMTU=20
> that they convey, a userspace IKE daemon usually doesn't. See  [1] for=20
> how people suggest doing it in C#.

Actually for example our code tries to get ICMP messages for IKE packets, m=
ostly so that we can make timeout shorter if we start to receive ICMP messa=
ges saying the other end is not going to responded.
If we get ICMP port unreachable for every single IKE packet we send, there =
is no point of waiting for 2 minutes and 15 retranmissions to realize that =
the other end is not responding, 20 seconds and 5 retries might be better..=
.

> IKE is supposed to be done in two round-trips. It would be better to=20
> just pick 576 and send way too many IKE fragments then to try PMTU=20
> discovery.

Yes. We are talking here about sending one 500-300 byte message from initia=
tor to responder, and getting about same sized message back.

After that we will only use few hundred bytes long exchanges.

Doing PMTU for sending and receiving one packet is bit overkill.
--
kivinen@iki.fi

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From: "Valery Smyslov" <svanru@gmail.com>
To: "Joe Touch" <touch@isi.edu>
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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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>> That's true. I don't see any requirements in section 10.4. that the 
>> probes
>> must be distinct. Request/reply nature of IKE's messages make them
>> well suitable for probes and I see no need here to introduce
>> separate mechanism, adding complexity, consuming bandwith and adding 
>> delay.
>> Moreover, section 10.4 explicitely allows using application data for
>> probes.
>
> Yes, but there are two components:
> - the probes
> - the timeouts
>
> You're using existing IKE messages *and* existing timeouts to determine 
> when there is a problem. A separate timer would be useful, if only to 
> allow you to decouple fragment retransmission from IKE transaction 
> retries.

No, the timeouts are different. I should have made it more expplicit in the 
draft.
When doing PMTU discovery host needs to try several MTU sizes
before considering peer to be dead, so exchange timeout must be
greater then the sum of all probes timeouts. Probe timeout should
be relatively short, just less then 10 seconds before going to next size,
while the whole exchange timeout is pretty long, about one minute or more.

>>> From Section 9: You don't talk about trying to make sure that IPv4 DF
>>> is set, as per Section 9, which means that even the conservative
>>> values you pick might continue to generate fragments downstream.
>>
>> With all due respect, I have to disagree that setting DF bit is needed
>> in case of IKE.
>> With IKE the goal is not to make sure that _no_ IP fragmentation takes
>> place
>> on the path, but to make sure that IP fragmentation issues, if any,
>> do not prevent IKE communications. So, IP fragmentation is present on
>> the path,
>> but it doesn't bother us - let it be.
>
> Agreed. But that's the case when IKE works with whatever message size you 
> start with.
>
>> If it doesn't allow IKE to work
>> (either by dropping IP fragments or by attacking destination host with
>> bogus fragments) - we will try to avoid it. So, in your example, even if
>> IP fragments are generated downstream, but those fragments are passed -
>> let them be.
>
> Your mechanism kicks in (presumably) when IKE fails, and then you retry 
> with application-layer fragments. App-layer fragments are useless if the 
> network level then fragments the messages. That's why you should set DF=1.

Again, there's no need for IKE to absolutely disallow IP fragmentation 
unless
it prevents IKE to communicate. If we have two routers down the
path, first fragmenting IP packets into size S1 and the second fragmenting
them into size S2, so that S2 < S1, and some device between them
that drops IP fragments, then it is enough for IKE to make so that
application fragments will have size < S1, but probably > S2.
Then we will avoid first fragmentation, but will still allow second one,
as it doesn't bother us. In this case application fragments are
useful, even with the presence of IP fragmentation.

Always setting DF bit in this case will probably increase the delay
before IKE SA is set up (as more probes will need to be done).

>> Note, that this approach is in line with advices, given for IKE in the
>> paper
>>
>> C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
>>               for UDP-based protocols", ACM Conference on Computer and
>>               Communications Security, October 2003.
>
> That paper doesn't consider IKE-level fragmentation, which you're 
> introducing. I agree that DF=0 for IKE without IKE-level fragmentation.

It does, in Section 3.3.

>> that is reffered by RFC5996 as recommendations to protect against IP
>> fragmentation issues. Section 3.3 of this paper discusses the ways PMTUD
>> should be done in IKE and finds it more advantageous not to set DF bit.
>
> That too refers to IKE without IKE-level fragmentation (see Sec 2). Again, 
> once you introduce fragmentation at the IKE-level to avoid IP 
> fragmentation, it makes no sense to continue to allow it at the IP level.

Please, see above.

> Joe

Regards,
Valery. 


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>> Always setting DF bit in this case will probably increase the delay
>> before IKE SA is set up (as more probes will need to be done).
>
> Except that if you continue to allow IP fragmentation, you can't claim 
> your solution is robust to IP fragment poisoning.

I think it is.

Consider the situation when IKE responder is under attack
via IP fragmentation (no matter which - poisoning attack or memory
exhausting attack). In any case responder will not be able to reply.
After some (short) timeout initiator will try to apply IKE Fragmentation.
Then, if those new messages are not fragmented on the path, they
will bypass reassembly code on responder and the attack will
be thwarted. If those messages are fragmented, even with their
smallest allowed size, then it doesn't matter whether DF bit is set or not.
If it is set, fragmenting device will drop messages, if it is not set,
than attack will not be thwarted. Nothing can be done.

Resume: setting DF bit in situation when even the smallest IKE Fragment
messages are still fragmented by IP will definitely prevent IKE to work.
Unsetting it will leave a chance (as with any DoS attack).

>> It does, in Section 3.3.
>
> Sorry - I missed that. But that section also gives good reasons why this 
> is a bad idea in IKE too.

It lists some drawbacks: added complexity and lack of individual ACKs.
They were discussed by WG and found acceptable.
It suggests using TCP instead. That was considered by WG,
but rejected due to numerous issues.

Valery. 


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Hi Matt,

the whole idea of the draft is avoiding IP fragmentation for IKE when
it prevents IKE to work. What about DF bit - I don't see how setting it=20
would help IKE to work.

Regards,
Valery.

  ----- Original Message -----=20
  From: Matt Mathis=20
  To: Valery Smyslov=20
  Cc: tsvwg ; tsv-ads@tools.ietf.org ; ipsec@ietf.org ; =
draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org ; tsv-dir@ietf.org =
; Joe Touch=20
  Sent: Saturday, October 26, 2013 12:41 AM
  Subject: Re: [tsvwg] [IPsec] TSVDIR-ish =
reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04


  I concur with Joe: once you have enough machinery work well with IPv6 =
fragmentation semantics, you should use it for IPv4 too, and =
unconditionally set DF.   This probably applies to *all* protocols.


  IPv4 reassembly is hopelessly out of scale.  IP ID wrap times are =
likely to be sub second for any large CGN connecting to any large =
service.....  They might even be shorter than the queuing times.


  I suspect that if you re-review decade old papers on fragmentation, =
you will find some scale assumptions that are no longer correct.  In =
that time the Internet has moved at least another two orders of =
magnitude in packet rates.


  Thanks,
  --MM--
  The best way to predict the future is to create it.  - Alan Kay

  Privacy matters!  We know from recent events that people are using our =
services to speak in defiance of unjust governments.   We treat privacy =
and security as matters of life and death, because for some users, they =
are.



  On Fri, Oct 25, 2013 at 1:18 PM, Joe Touch <touch@isi.edu> wrote:



    On 10/24/2013 10:45 PM, Valery Smyslov wrote:
    ...

        You're using existing IKE messages *and* existing timeouts to
        determine when there is a problem. A separate timer would be =
useful,
        if only to allow you to decouple fragment retransmission from =
IKE
        transaction retries.



      No, the timeouts are different. I should have made it more =
expplicit in
      the draft.


    That'd be useful.

    ...

      Always setting DF bit in this case will probably increase the =
delay
      before IKE SA is set up (as more probes will need to be done).


    Except that if you continue to allow IP fragmentation, you can't =
claim your solution is robust to IP fragment poisoning.


          Note, that this approach is in line with advices, given for =
IKE in the
          paper

          C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
                        for UDP-based protocols", ACM Conference on =
Computer and
                        Communications Security, October 2003.


        That paper doesn't consider IKE-level fragmentation, which =
you're
        introducing. I agree that DF=3D0 for IKE without IKE-level =
fragmentation.



      It does, in Section 3.3.


    Sorry - I missed that. But that section also gives good reasons why =
this is a bad idea in IKE too.

    Joe



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<DIV><FONT size=3D2>Hi Matt,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>the whole idea of the draft is avoiding IP =
fragmentation for=20
IKE when</FONT></DIV>
<DIV><FONT size=3D2>it prevents IKE to work. </FONT><FONT size=3D2>What =
about DF bit=20
- I don't&nbsp;see how setting it </FONT></DIV>
<DIV><FONT size=3D2>would help IKE to work.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Regards,</FONT></DIV>
<DIV><FONT size=3D2>Valery.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; =
PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px"=20
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  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: =
black"><B>From:</B>=20
  <A title=3Dmattmathis@google.com =
href=3D"mailto:mattmathis@google.com">Matt=20
  Mathis</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A title=3Dsvanru@gmail.com =

  href=3D"mailto:svanru@gmail.com">Valery Smyslov</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A title=3Dtsvwg@ietf.org=20
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  title=3Dipsec@ietf.org =
href=3D"mailto:ipsec@ietf.org">ipsec@ietf.org</A> ; <A=20
  title=3Ddraft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org=20
  =
href=3D"mailto:draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org">dra=
ft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org</A>=20
  ; <A title=3Dtsv-dir@ietf.org=20
  href=3D"mailto:tsv-dir@ietf.org">tsv-dir@ietf.org</A> ; <A =
title=3Dtouch@isi.edu=20
  href=3D"mailto:touch@isi.edu">Joe Touch</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Saturday, October 26, =
2013 12:41=20
  AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [tsvwg] [IPsec] =
TSVDIR-ish=20
  reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04</DIV>
  <DIV><BR></DIV>
  <DIV dir=3Dltr>I concur with Joe: once you have enough machinery work =
well with=20
  IPv6 fragmentation semantics, you should use it for IPv4 too, and=20
  unconditionally set DF. &nbsp; This probably applies to *all* =
protocols.
  <DIV><BR></DIV>
  <DIV>IPv4 reassembly is hopelessly out of scale. &nbsp;IP ID wrap =
times are=20
  likely to be sub second for any large CGN connecting to any large =
service.....=20
  &nbsp;They might even be shorter than the queuing times.
  <DIV><BR></DIV>
  <DIV>I suspect that if you re-review decade old papers on =
fragmentation, you=20
  will find some scale assumptions that are no longer correct. &nbsp;In =
that=20
  time the Internet has moved at least another two orders of magnitude =
in packet=20
  rates.</DIV></DIV></DIV>
  <DIV class=3Dgmail_extra><BR clear=3Dall>
  <DIV>
  <DIV dir=3Dltr>
  <DIV>Thanks,</DIV>--MM--<BR>The best way to predict the future is to =
create=20
  it. &nbsp;- Alan Kay<BR><BR>Privacy matters! &nbsp;We know from recent =
events=20
  that people are using our services to speak in defiance of unjust =
governments.=20
  &nbsp; We treat privacy and security as matters of life and death, =
because for=20
  some users, they are.</DIV></DIV><BR><BR>
  <DIV class=3Dgmail_quote>On Fri, Oct 25, 2013 at 1:18 PM, Joe Touch =
<SPAN=20
  dir=3Dltr>&lt;<A href=3D"mailto:touch@isi.edu"=20
  target=3D_blank>touch@isi.edu</A>&gt;</SPAN> wrote:<BR>
  <BLOCKQUOTE=20
  style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
  class=3Dgmail_quote><BR><BR>On 10/24/2013 10:45 PM, Valery Smyslov=20
    wrote:<BR>...<BR>
    <BLOCKQUOTE=20
    style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
    class=3Dgmail_quote>
      <DIV class=3Dim>
      <BLOCKQUOTE=20
      style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
      class=3Dgmail_quote>You're using existing IKE messages *and* =
existing=20
        timeouts to<BR>determine when there is a problem. A separate =
timer would=20
        be useful,<BR>if only to allow you to decouple fragment =
retransmission=20
        from IKE<BR>transaction retries.<BR></BLOCKQUOTE><BR></DIV>No, =
the=20
      timeouts are different. I should have made it more expplicit =
in<BR>the=20
      draft.<BR></BLOCKQUOTE><BR>That'd be useful.<BR><BR>...<BR>
    <BLOCKQUOTE=20
    style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
    class=3Dgmail_quote>Always setting DF bit in this case will probably =

      increase the delay<BR>before IKE SA is set up (as more probes will =
need to=20
      be done).<BR></BLOCKQUOTE><BR>Except that if you continue to allow =
IP=20
    fragmentation, you can't claim your solution is robust to IP =
fragment=20
    poisoning.<BR><BR>
    <BLOCKQUOTE=20
    style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
    class=3Dgmail_quote>
      <DIV class=3Dim>
      <BLOCKQUOTE=20
      style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
      class=3Dgmail_quote>
        <BLOCKQUOTE=20
        style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
        class=3Dgmail_quote>Note, that this approach is in line with =
advices,=20
          given for IKE in the<BR>paper<BR><BR>C. Kaufman, R. Perlman, =
and B.=20
          Sommerfeld, "DoS protection<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
          &nbsp; &nbsp; for UDP-based protocols", ACM Conference on =
Computer=20
          and<BR>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
Communications=20
          Security, October 2003.<BR></BLOCKQUOTE><BR>That paper doesn't =
consider=20
        IKE-level fragmentation, which you're<BR>introducing. I agree =
that DF=3D0=20
        for IKE without IKE-level =
fragmentation.<BR></BLOCKQUOTE><BR></DIV>It=20
      does, in Section 3.3.<BR></BLOCKQUOTE><BR>Sorry - I missed that. =
But that=20
    section also gives good reasons why this is a bad idea in IKE =
too.<SPAN=20
    class=3DHOEnZb><FONT=20
  =
color=3D#888888><BR><BR>Joe<BR></FONT></SPAN></BLOCKQUOTE></DIV><BR></DIV=
></BLOCKQUOTE></BODY></HTML>

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From: "Valery Smyslov" <svanru@gmail.com>
To: "Joe Touch" <touch@isi.edu>
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>> Consider the situation when IKE responder is under attack
>> via IP fragmentation (no matter which - poisoning attack or memory
>> exhausting attack). In any case responder will not be able to reply.
>> After some (short) timeout initiator will try to apply IKE Fragmentation.
>> Then, if those new messages are not fragmented on the path, they
>> will bypass reassembly code on responder and the attack will
>> be thwarted. If those messages are fragmented, even with their
>> smallest allowed size, then it doesn't matter whether DF bit is set or 
>> not.
>> If it is set, fragmenting device will drop messages, if it is not set,
>> than attack will not be thwarted. Nothing can be done.
>
> You're not proposing to use the smallest size, which would be 68 bytes.

No, as it is not possible with this solution. Depending on algorithms 
involved
the smallest IP packet containing IKE message is about
100 bytes, and it is almost useless as in this case it  contains only
a few bytes of useful data. The minimum IP packet size still useful
would be about 256 bytes.

> So by not setting DF, you're enabling a fragmentation attack. If you set 
> DF, a receiver could drop all fragments from addresses it wants to protect 
> against IKE fragment poisoning.

Yes. But if even the smallest IKE messages are to be fragmented
by intermediate node, then setting DF bit (or dropping all IP fragments)
will just not enable IKE to succeed. By not setting it we give IKE a chance.


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Subject: Re: [tsvwg] [IPsec] TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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While I agree with you in general, I think that IKE has its own =
peculiarity,
that makes things a bit different.

There is a difference between IPv6 and IPv4. In IPv6 one can assume,
that packets below 1280 bytes will not be fragmented en route. This =
leaves enough room
to make IKE level fragmentation and be sure, that no IP fragmentation =
will take place.
With IPv4 any IP packet greater 68 bytes is fragmentable. Due to IKE =
message
construction restrictions you cannot make IKE fragments that small,
so you always have a possibility, that packets will still be fragmented =
by IP level.
If you send them with DF=3D1 and intermediate device still wants to =
fragment them,
then you will just prevent IKE to succeed with 100% probability. If in =
this case DF=3D0,
then it might work.

If, as you suggest, originating host performs IP fragmentation and sends =
IP fragments
with DF=3D1, then we have exactly the problem the draft is intended to =
solve -=20
the presence of IP fragments on the whole path. Draft introduces =
IKE-level fragmentation=20
to avoid IP fragmentation whenever possible and to only let it be if it =
is impossible. But if it=20
is unavoidable, then there are more chances for IKE to work if IP =
fragments exist
only on the part of the path, i.e. when fragmentation is done by =
intermediate device.

  ----- Original Message -----=20
  From: Matt Mathis=20
  To: Valery Smyslov=20
  Cc: tsvwg ; tsv-ads@tools.ietf.org ; ipsec@ietf.org ; =
draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org ; tsv-dir@ietf.org =
; Joe Touch=20
  Sent: Tuesday, October 29, 2013 2:22 AM
  Subject: Re: [tsvwg] [IPsec] TSVDIR-ish =
reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04


  Setting DF is supposed to prevent some midpath IPv4 device from =
silently "fixing" a MTU problem by doing you a favor that you don't =
want.


  IPv4 fragmentation can be used in a mode that mimics IPv6: Only =
fragment on the originating host, always set DF on on all outbound =
packets, even if they are already fragmented.     Any protocol or =
application that does not work in this configuration will also fail on =
IPv6.


  Thanks,
  --MM--
  The best way to predict the future is to create it.  - Alan Kay

  Privacy matters!  We know from recent events that people are using our =
services to speak in defiance of unjust governments.   We treat privacy =
and security as matters of life and death, because for some users, they =
are.



  On Sun, Oct 27, 2013 at 10:50 PM, Valery Smyslov <svanru@gmail.com> =
wrote:

    Hi Matt,

    the whole idea of the draft is avoiding IP fragmentation for IKE =
when
    it prevents IKE to work. What about DF bit - I don't see how setting =
it=20
    would help IKE to work.

    Regards,
    Valery.

      ----- Original Message -----=20
      From: Matt Mathis=20
      To: Valery Smyslov=20
      Cc: tsvwg ; tsv-ads@tools.ietf.org ; ipsec@ietf.org ; =
draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org ; tsv-dir@ietf.org =
; Joe Touch=20
      Sent: Saturday, October 26, 2013 12:41 AM
      Subject: Re: [tsvwg] [IPsec] TSVDIR-ish =
reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04


      I concur with Joe: once you have enough machinery work well with =
IPv6 fragmentation semantics, you should use it for IPv4 too, and =
unconditionally set DF.   This probably applies to *all* protocols.=20


      IPv4 reassembly is hopelessly out of scale.  IP ID wrap times are =
likely to be sub second for any large CGN connecting to any large =
service.....  They might even be shorter than the queuing times.=20


      I suspect that if you re-review decade old papers on =
fragmentation, you will find some scale assumptions that are no longer =
correct.  In that time the Internet has moved at least another two =
orders of magnitude in packet rates.


      Thanks,
      --MM--
      The best way to predict the future is to create it.  - Alan Kay

      Privacy matters!  We know from recent events that people are using =
our services to speak in defiance of unjust governments.   We treat =
privacy and security as matters of life and death, because for some =
users, they are.



      On Fri, Oct 25, 2013 at 1:18 PM, Joe Touch <touch@isi.edu> wrote:



        On 10/24/2013 10:45 PM, Valery Smyslov wrote:
        ...

            You're using existing IKE messages *and* existing timeouts =
to
            determine when there is a problem. A separate timer would be =
useful,
            if only to allow you to decouple fragment retransmission =
from IKE
            transaction retries.



          No, the timeouts are different. I should have made it more =
expplicit in
          the draft.


        That'd be useful.

        ...

          Always setting DF bit in this case will probably increase the =
delay
          before IKE SA is set up (as more probes will need to be done).


        Except that if you continue to allow IP fragmentation, you can't =
claim your solution is robust to IP fragment poisoning.


              Note, that this approach is in line with advices, given =
for IKE in the
              paper

              C. Kaufman, R. Perlman, and B. Sommerfeld, "DoS protection
                            for UDP-based protocols", ACM Conference on =
Computer and
                            Communications Security, October 2003.


            That paper doesn't consider IKE-level fragmentation, which =
you're
            introducing. I agree that DF=3D0 for IKE without IKE-level =
fragmentation.



          It does, in Section 3.3.


        Sorry - I missed that. But that section also gives good reasons =
why this is a bad idea in IKE too.

        Joe





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<DIV><FONT size=3D2>While I agree with you in general, I think that IKE =
has its=20
own peculiarity,</FONT></DIV>
<DIV><FONT size=3D2>that makes things a bit different.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>There is a difference between IPv6 and IPv4. In IPv6 =
one can=20
assume,</FONT></DIV>
<DIV><FONT size=3D2>that packets below 1280 bytes will not be fragmented =
en route.=20
This leaves enough room</FONT></DIV>
<DIV><FONT size=3D2>to make IKE level fragmentation and be sure, that no =
IP=20
fragmentation will take place.</FONT></DIV>
<DIV><FONT size=3D2>With IPv4 any IP packet greater 68 bytes is =
fragmentable. Due=20
to IKE message</FONT></DIV>
<DIV><FONT size=3D2>construction restrictions you cannot make IKE =
fragments that=20
small,</FONT></DIV>
<DIV><FONT size=3D2>so you always have a possibility, that packets will =
still be=20
fragmented by IP level.</FONT></DIV>
<DIV><FONT size=3D2>If you send them with DF=3D1 and intermediate device =
still wants=20
to fragment them,</FONT></DIV>
<DIV><FONT size=3D2>then you will just prevent IKE to succeed with 100%=20
probability. If in this case DF=3D0,</FONT></DIV>
<DIV><FONT size=3D2>then it might work.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>If, as you suggest, originating host performs IP =
fragmentation=20
and sends&nbsp;IP fragments</FONT></DIV>
<DIV><FONT size=3D2>with DF=3D1, then we have exactly the problem the =
draft is=20
intended to solve - </FONT></DIV>
<DIV><FONT size=3D2>the presence of IP fragments on the whole path. =
Draft=20
introduces IKE-level fragmentation </FONT></DIV>
<DIV><FONT size=3D2>to avoid IP fragmentation </FONT><FONT =
size=3D2>whenever=20
possible and to only let it be if it is impossible. But if it =
</FONT></DIV>
<DIV><FONT size=3D2>is unavoidable, then there are more chances for IKE =
to work if=20
IP fragments exist</FONT></DIV>
<DIV><FONT size=3D2>only on the part of the path, i.e.&nbsp;when=20
fragmentation&nbsp;is done by intermediate device.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; =
PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px"=20
dir=3Dltr>
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: =
black"><B>From:</B>=20
  <A title=3Dmattmathis@google.com =
href=3D"mailto:mattmathis@google.com">Matt=20
  Mathis</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A title=3Dsvanru@gmail.com =

  href=3D"mailto:svanru@gmail.com">Valery Smyslov</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A title=3Dtsvwg@ietf.org=20
  href=3D"mailto:tsvwg@ietf.org">tsvwg</A> ; <A =
title=3Dtsv-ads@tools.ietf.org=20
  href=3D"mailto:tsv-ads@tools.ietf.org">tsv-ads@tools.ietf.org</A> ; <A =

  title=3Dipsec@ietf.org =
href=3D"mailto:ipsec@ietf.org">ipsec@ietf.org</A> ; <A=20
  title=3Ddraft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org=20
  =
href=3D"mailto:draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org">dra=
ft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org</A>=20
  ; <A title=3Dtsv-dir@ietf.org=20
  href=3D"mailto:tsv-dir@ietf.org">tsv-dir@ietf.org</A> ; <A =
title=3Dtouch@isi.edu=20
  href=3D"mailto:touch@isi.edu">Joe Touch</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, October 29, 2013 =
2:22=20
  AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [tsvwg] [IPsec] =
TSVDIR-ish=20
  reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04</DIV>
  <DIV><BR></DIV>
  <DIV dir=3Dltr>Setting DF is supposed to prevent some midpath IPv4 =
device from=20
  silently "fixing" a MTU problem by doing you a favor that you don't =
want.
  <DIV><BR></DIV>
  <DIV>IPv4 fragmentation can be used in a mode that mimics IPv6: Only =
fragment=20
  on the originating host, always set DF on on all outbound packets, =
even if=20
  they are already fragmented. &nbsp; &nbsp; Any protocol or application =
that=20
  does not work in this configuration will also fail on =
IPv6.</DIV></DIV>
  <DIV class=3Dgmail_extra><BR clear=3Dall>
  <DIV>
  <DIV dir=3Dltr>
  <DIV>Thanks,</DIV>--MM--<BR>The best way to predict the future is to =
create=20
  it. &nbsp;- Alan Kay<BR><BR>Privacy matters! &nbsp;We know from recent =
events=20
  that people are using our services to speak in defiance of unjust =
governments.=20
  &nbsp; We treat privacy and security as matters of life and death, =
because for=20
  some users, they are.</DIV></DIV><BR><BR>
  <DIV class=3Dgmail_quote>On Sun, Oct 27, 2013 at 10:50 PM, Valery =
Smyslov <SPAN=20
  dir=3Dltr>&lt;<A href=3D"mailto:svanru@gmail.com"=20
  target=3D_blank>svanru@gmail.com</A>&gt;</SPAN> wrote:<BR>
  <BLOCKQUOTE=20
  style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
  class=3Dgmail_quote><U></U>
    <DIV bgcolor=3D"#ffffff">
    <DIV><FONT size=3D+0>Hi Matt,</FONT></DIV>
    <DIV><FONT size=3D+0></FONT>&nbsp;</DIV>
    <DIV><FONT size=3D+0>the whole idea of the draft is avoiding IP =
fragmentation=20
    for IKE when</FONT></DIV>
    <DIV><FONT size=3D+0>it prevents IKE to work. </FONT><FONT =
size=3D+0>What about=20
    DF bit - I don't&nbsp;see how setting it </FONT></DIV>
    <DIV><FONT size=3D+0>would help IKE to work.</FONT></DIV>
    <DIV><FONT size=3D+0></FONT>&nbsp;</DIV>
    <DIV><FONT size=3D+0>Regards,</FONT></DIV>
    <DIV><FONT size=3D+0>Valery.</FONT></DIV>
    <DIV>
    <DIV class=3Dh5>
    <DIV><FONT size=3D+0></FONT>&nbsp;</DIV>
    <BLOCKQUOTE=20
    style=3D"BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; =
PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px"=20
    dir=3Dltr>
      <DIV style=3D"FONT: 10pt arial">----- Original Message ----- =
</DIV>
      <DIV style=3D"FONT: 10pt arial; BACKGROUND: #e4e4e4"><B>From:</B> =
<A=20
      title=3Dmattmathis@google.com =
href=3D"mailto:mattmathis@google.com"=20
      target=3D_blank>Matt Mathis</A> </DIV>
      <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Dsvanru@gmail.com=20
      href=3D"mailto:svanru@gmail.com" target=3D_blank>Valery =
Smyslov</A> </DIV>
      <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A =
title=3Dtsvwg@ietf.org=20
      href=3D"mailto:tsvwg@ietf.org" target=3D_blank>tsvwg</A> ; <A=20
      title=3Dtsv-ads@tools.ietf.org =
href=3D"mailto:tsv-ads@tools.ietf.org"=20
      target=3D_blank>tsv-ads@tools.ietf.org</A> ; <A =
title=3Dipsec@ietf.org=20
      href=3D"mailto:ipsec@ietf.org" target=3D_blank>ipsec@ietf.org</A> =
; <A=20
      title=3Ddraft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org=20
      =
href=3D"mailto:draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org"=20
      =
target=3D_blank>draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org</A>=
 ;=20
      <A title=3Dtsv-dir@ietf.org href=3D"mailto:tsv-dir@ietf.org"=20
      target=3D_blank>tsv-dir@ietf.org</A> ; <A title=3Dtouch@isi.edu=20
      href=3D"mailto:touch@isi.edu" target=3D_blank>Joe Touch</A> </DIV>
      <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Saturday, October 26, =
2013=20
      12:41 AM</DIV>
      <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [tsvwg] =
[IPsec]=20
      TSVDIR-ish reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04</DIV>
      <DIV><BR></DIV>
      <DIV dir=3Dltr>I concur with Joe: once you have enough machinery =
work well=20
      with IPv6 fragmentation semantics, you should use it for IPv4 too, =
and=20
      unconditionally set DF. &nbsp; This probably applies to *all* =
protocols.=20
      <DIV><BR></DIV>
      <DIV>IPv4 reassembly is hopelessly out of scale. &nbsp;IP ID wrap =
times=20
      are likely to be sub second for any large CGN connecting to any =
large=20
      service..... &nbsp;They might even be shorter than the queuing =
times.=20
      <DIV><BR></DIV>
      <DIV>I suspect that if you re-review decade old papers on =
fragmentation,=20
      you will find some scale assumptions that are no longer correct. =
&nbsp;In=20
      that time the Internet has moved at least another two orders of =
magnitude=20
      in packet rates.</DIV></DIV></DIV>
      <DIV class=3Dgmail_extra><BR clear=3Dall>
      <DIV>
      <DIV dir=3Dltr>
      <DIV>Thanks,</DIV>--MM--<BR>The best way to predict the future is =
to=20
      create it. &nbsp;- Alan Kay<BR><BR>Privacy matters! &nbsp;We know =
from=20
      recent events that people are using our services to speak in =
defiance of=20
      unjust governments. &nbsp; We treat privacy and security as =
matters of=20
      life and death, because for some users, they =
are.</DIV></DIV><BR><BR>
      <DIV class=3Dgmail_quote>On Fri, Oct 25, 2013 at 1:18 PM, Joe =
Touch <SPAN=20
      dir=3Dltr>&lt;<A href=3D"mailto:touch@isi.edu"=20
      target=3D_blank>touch@isi.edu</A>&gt;</SPAN> wrote:<BR>
      <BLOCKQUOTE=20
      style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
      class=3Dgmail_quote><BR><BR>On 10/24/2013 10:45 PM, Valery Smyslov =

        wrote:<BR>...<BR>
        <BLOCKQUOTE=20
        style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
        class=3Dgmail_quote>
          <DIV>
          <BLOCKQUOTE=20
          style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px =
0.8ex; PADDING-LEFT: 1ex"=20
          class=3Dgmail_quote>You're using existing IKE messages *and* =
existing=20
            timeouts to<BR>determine when there is a problem. A separate =
timer=20
            would be useful,<BR>if only to allow you to decouple =
fragment=20
            retransmission from IKE<BR>transaction=20
          retries.<BR></BLOCKQUOTE><BR></DIV>No, the timeouts are =
different. I=20
          should have made it more expplicit in<BR>the=20
        draft.<BR></BLOCKQUOTE><BR>That'd be useful.<BR><BR>...<BR>
        <BLOCKQUOTE=20
        style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
        class=3Dgmail_quote>Always setting DF bit in this case will =
probably=20
          increase the delay<BR>before IKE SA is set up (as more probes =
will=20
          need to be done).<BR></BLOCKQUOTE><BR>Except that if you =
continue to=20
        allow IP fragmentation, you can't claim your solution is robust =
to IP=20
        fragment poisoning.<BR><BR>
        <BLOCKQUOTE=20
        style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; =
PADDING-LEFT: 1ex"=20
        class=3Dgmail_quote>
          <DIV>
          <BLOCKQUOTE=20
          style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px =
0.8ex; PADDING-LEFT: 1ex"=20
          class=3Dgmail_quote>
            <BLOCKQUOTE=20
            style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px =
0.8ex; PADDING-LEFT: 1ex"=20
            class=3Dgmail_quote>Note, that this approach is in line with =

              advices, given for IKE in the<BR>paper<BR><BR>C. Kaufman, =
R.=20
              Perlman, and B. Sommerfeld, "DoS protection<BR>&nbsp; =
&nbsp;=20
              &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; for UDP-based =
protocols", ACM=20
              Conference on Computer and<BR>&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
              &nbsp; &nbsp; Communications Security, October=20
            2003.<BR></BLOCKQUOTE><BR>That paper doesn't consider =
IKE-level=20
            fragmentation, which you're<BR>introducing. I agree that =
DF=3D0 for=20
            IKE without IKE-level =
fragmentation.<BR></BLOCKQUOTE><BR></DIV>It=20
          does, in Section 3.3.<BR></BLOCKQUOTE><BR>Sorry - I missed =
that. But=20
        that section also gives good reasons why this is a bad idea in =
IKE=20
        too.<SPAN><FONT=20
      =
color=3D#888888><BR><BR>Joe<BR></FONT></SPAN></BLOCKQUOTE></DIV><BR></DIV=
></BLOCKQUOTE></DIV></DIV></DIV></BLOCKQUOTE></DIV><BR></DIV></BLOCKQUOTE=
></BODY></HTML>

------=_NextPart_000_056F_01CED491.732A1FD0--


From kivinen@iki.fi  Tue Oct 29 07:03:20 2013
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To: "Valery Smyslov" <svanru@gmail.com>
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Cc: tsvwg <tsvwg@ietf.org>, Joe Touch <touch@isi.edu>, tsv-ads@tools.ietf.org, ipsec@ietf.org, draft-ietf-ipsecme-ikev2-fragmentation@tools.ietf.org, Matt Mathis <mattmathis@google.com>, tsv-dir@ietf.org
Subject: Re: [tsvwg] [IPsec] TSVDIR-ish	reviewofdraft-ietf-ipsecme-ikev2-fragmentation-04
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Valery Smyslov writes:
> Draft introduces IKE-level fragmentation to avoid IP fragmentation
> whenever possible and to only let it be if it is impossible.

Actually the first preference is to use IP fragmentation, and if that
works, it is what we use, and we do not try IKE fragmentation at all.
Only if the large packets do not go through then we try to enable IKE
fragmentation and so on...

So all of this draft is used only if there is some device on the path
that will drop at least some fragments somewhere. And yes, we might
miscategorize some network errors as IP fragments getting lost, as we
have no indication from the network whether the packet was lost
because it got fragmented on the IP level, or for some reasons. There
is no way IKEv2 could even get that information, so we assume that if
multiple full size retransmission of IKEv2 packets are lost, then
there might be fragmentation issue, and we enable IKE fragmentation.

> But if it is unavoidable, then there are more chances for IKE to
> work if IP fragments exist only on the part of the path, i.e. when
> fragmentation is done by intermediate device.

Yep.
-- 
kivinen@iki.fi

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Title: Liaison reply to "Development of Informative Codepoint mapping in IT=
U-T Study Group 12" [to IETF TSV]
Submission Date: 2013-10-25
URL of the IETF Web page: http://datatracker.ietf.org/liaison/1287/

From: ITU-T Study Group 12, Q17/12 (Hiroshi Ota <hiroshi.ota@itu.int>)
To: Transport Area Working Group (Gorry Fairhurst <gorry@erg.abdn.ac.uk>, D=
avid Black <david.black@emc.com>, James Polk <jmpolk@cisco.com>)
Cc: Martin Stiemerling <mls.ietf@gmail.com>,Spencer Dawkins <spencerdawkins=
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Reponse Contact: =

Technical Contact: =

Purpose: For information

Body: =

Attachments:

    Liaison reply to "Development of Informative Codepoint mapping in ITU-T=
 Study Group 12" [to IETF TSV]
    https://datatracker.ietf.org/documents/LIAISON/liaison-2013-10-25-itu-t=
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Date: Thu, 31 Oct 2013 08:06:54 -0000
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Subject: [tsvwg] Presenters: Copies of slides needed for tsvwg at ietf-88
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If you are listed in the Agenda for tsvwg (see below), please send slides
to the Chair by Sunday noon, so that these may be uploaded for the
meeting:

http://www.ietf.org/proceedings/88/agenda/agenda-88-tsvwg

Uploading draft slides ahead of the actual tsvwg meeting is really helpful
for our remote participants - even if the slide deck may need minor
adjustment prior to the meeting.

Gorry

