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Subject: Re: [tcpm] Four-way Handshake justification
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> On Oct 31, 2014, at 3:20 PM, Joe Touch <touch@ISI.EDU> wrote:
> On 10/31/2014 11:01 AM, John Leslie wrote:
>>> (Yes, there are other timers, such as for delayed ACKs, but that is
>>>> an efficiency enhancement, TCP can functionally operate without =
them).
>=20
> I'm not sure I agree with this.
>=20
> It seems like TCP would lockup and never make progress if packets were =
lost.

The sentence right before that was:
	Timers ensure that control and data packets are retransmitted
	if they are not acknowledged.

Those timers ensure that TCP doesn=E2=80=99t lock up, by retransmitting =
un-ACKed
data and control(SYN,FIN) packets.  ACK only packets are never =
retransmitted.

The delayed ACK timer isn=E2=80=99t part of the reliability of TCP, it =
is a
performance improvement that allows consolidation of ACKs.  ACKs could
always be sent immediately in response to the incoming packet instead
of being delayed by a timer.

		-David

>=20
> Joe
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] Four-way Handshake justification
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> On Nov 1, 2014, at 10:07 AM, David Borman <dab@weston.borman.com> wrote:
>=20
>=20
>> On Oct 31, 2014, at 3:20 PM, Joe Touch <touch@ISI.EDU> wrote:
>> On 10/31/2014 11:01 AM, John Leslie wrote:
>>>> (Yes, there are other timers, such as for delayed ACKs, but that is
>>>>> an efficiency enhancement, TCP can functionally operate without them).=

>>=20
>> I'm not sure I agree with this.
>>=20
>> It seems like TCP would lockup and never make progress if packets were lo=
st.
>=20
> The sentence right before that was:
>    Timers ensure that control and data packets are retransmitted
>    if they are not acknowledged.

I'm not sure how you can claim tcp can functionally operate without them. Ip=
 isn't lossless even over a lossless link layer.=20

Many argue that timers are the only fundamental part of any protocol (esp. W=
atson, Day).=20

> Those timers ensure that TCP doesn=E2=80=99t lock up, by retransmitting un=
-ACKed
> data and control(SYN,FIN) packets.  ACK only packets are never retransmitt=
ed.

What about keepalives?

> The delayed ACK timer isn=E2=80=99t part of the reliability of TCP, it is a=

> performance improvement that allows consolidation of ACKs.  ACKs could
> always be sent immediately in response to the incoming packet instead
> of being delayed by a timer.
>=20
>        -David
>=20
>>=20
>> Joe
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20


From nobody Sun Nov  2 15:33:39 2014
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Per Hurtig <per.hurtig@kau.se>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: draft-ietf-tcpm-rtorestart-04
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Hi,

I've read -04 and I have a couple of questions and suggestions:

* Section 4: The "total number of outstanding and previously unsent segment=
s" in sum of two variables, right? If so, why not use the term "sum"?

* Section 4: I also wonder if the text should define "previously unsent seg=
ments". The term "previously unsent data" is very well-known in the RFC ser=
ies. A quick search revealed that "previously unsent segments" is usually o=
nly used to refer to situations when segments with this property have indee=
d been transmitted. Is my understanding correct that draft-ietf-tcpm-rtores=
tart-04 uses this term differently? I think it here refers to data that wil=
l be sent *in future*. In that case, I wonder if it is really clear how the=
 sender can determine the number of "previously unsent segments". For insta=
nce, that number may depend on the segmentation strategy of the sender, in =
particular for applications that use many small write() calls. In this case=
 a TCP stack may e.g. decide to send partial segments. I am not sure if the=
 actual number of segments created from the "unsent data" is really known i=
n advance always. Or do I miss something?

* Section 5.1: "Given RTOR's ability to only work when it is beneficial for=
 the loss recovery process, it is suitable as a system-wide default mechani=
sm for TCP traffic." I think other text in the draft asks for further exper=
imentation regarding the actual trade-off between benefit and risk. Given t=
hat, I think another wording should be used (e.g., "Given that RTOR is a mo=
stly conservative algorithm", ...). Given that this is not a PS document, I=
'd also prefer a more careful phrasing regarding the recommendation as syst=
em-wide default (e.g., "it is suitable for experimentation as system-wide d=
efault").

* Section 5.2: As mentioned in the last face-to-face meeting, spurious time=
out negatively affect the network by needlessly sending data. This negative=
 effect is not mentioned in this section. Why?

* Section 5.2: Mobile networks have a variable RTT, including e.g. longer d=
elays during handovers (either without or with transient loss during the ha=
ndover). A lot of past work on spurious timeout recovery has been motivated=
 by these delay spikes. Reducing the RTO almost certainly has an impact in =
such environments; I think performance can both increase or decrease. If no=
 data on that is available, mobile networks seem to me an area where more e=
xperimentation would be particularly useful, right?

* Section 5.2: Is there any data on the risk of spurious timeouts in networ=
ks where RTT and the minimum RTO are of the same order of magnitude? For in=
stance, in satellite networks the RTT can be huge.

* Section 5.2: "However, with respect to RTOR spurious timeouts are only a =
problem for applications transmitting multiple bursts of data within a sing=
le flow." I think this section should be reworded and it should more explic=
itly discuss the impact of RTOR on HTTP/1.1 and HTTP/2.0 traffic, including=
 e.g. adaptive video streaming. They all transmit "multiple bursts of data =
within a single flow". To me, the current sentence could imply that RTOR wo=
uld be a "problem" for a vast majority of Internet traffic. A more explicit=
 discussion on how RTOR affects HTTP/1.1 and HTTP/2.0 would IMHO make a lot=
 of sense.

* Section 5.3: Given that not all stacks are segment-based, this section se=
ems really relevant. However, the description how to determine outstanding =
segments is vague ("it is possible to exactly determine..."). I think this =
could be reworded to provide more guidance to implementers. The same applie=
s to the calculation of "previously unsent segments"; for instance, the tex=
t does not explain what happens if the "number of bytes in the send queue" =
is not a multiple of the SMSS (see also above regarding the definition of t=
his variable).

* Section 6: "In contrast, RTOR is trying to make the RTO more appropriate =
in cases where there is no need to be overly cautious." I think this senten=
ce should use a more neutral language. I think whether an RTO value is "app=
ropriate" is impossible to know (in advance), and whether a more aggressive=
 timeout is "cautious" or not may depend on the network under consideration=
. For instance, an alternative would be to compare the complexity of RTOR a=
nd TLP.

* Section 9: I wonder if an attacker could try to send certain ACK patterns=
 to increase the risk of timeouts e.g. at a Web Server, leveraging the smal=
ler RTO value. If successful, the RTOs could significantly reduce applicati=
on performance. Probably, such an attacker could cause much more harm by ot=
her means, i.e., this may not be a significant security risk. But the curre=
nt security analysis is very short.

Thanks

Michael


> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Per Hurtig
> Sent: Tuesday, October 28, 2014 12:18 AM
> To: tcpm@ietf.org
> Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rtorestart-04.txt
>=20
> Hi all,
>=20
> we have now updated the draft to address the outstanding comments. The
> main changes from last version are:
>=20
>    o  Changed the algorithm to allow RTOR when there is unsent data
> available, but the cwnd does not allow transmission.
>    o  Changed the algorithm to not trigger if "RTO - T_earliest" <=3D 0,
> to avoid that ACKs to previous retransmissions trigger premature
> timeouts.
>    o  Made minor adjustments throughout the document to adjust for the
> algorithmic changes.
>    o  Improved the wording throughout the document.
>=20
>=20
> for experimental results and a Linux implementation, please visit:
> http://riteproject.eu/resources/rto-restart/
>=20
>=20
>=20
> Cheers,
> Per
>=20
> On 2014-10-28 00:11, internet-drafts@ietf.org wrote:
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >   This draft is a work item of the TCP Maintenance and Minor
> Extensions Working Group of the IETF.
> >
> >          Title           : TCP and SCTP RTO Restart
> >          Authors         : Per Hurtig
> >                            Anna Brunstrom
> >                            Andreas Petlund
> >                            Michael Welzl
> > 	Filename        : draft-ietf-tcpm-rtorestart-04.txt
> > 	Pages           : 13
> > 	Date            : 2014-10-27
> >
> > Abstract:
> >     This document describes a modified algorithm for managing the TCP
> and
> >     SCTP retransmission timers that provides faster loss recovery
> when
> >     there is a small amount of outstanding data for a connection.
> The
> >     modification, RTO Restart (RTOR), allows the transport to restart
> its
> >     retransmission timer more aggressively in situations where fast
> >     retransmit cannot be used.  This enables faster loss detection
> and
> >     recovery for connections that are short-lived or application-
> limited.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-tcpm-rtorestart/
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-tcpm-rtorestart-04
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rtorestart-04
> >
> >
> > 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/
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> >
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon Nov  3 15:12:38 2014
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Date: Mon, 3 Nov 2014 23:12:14 +0000
To: Joe Touch <touch@isi.edu>
From: Bob Briscoe <bob.briscoe@bt.com>
In-Reply-To: <5446C9FB.7050006@isi.edu>
References: <F67E5C77-60CC-4617-A8C1-BF4263111CBF@weston.borman.com> <544047E3.8080900@isi.edu> <6A5AD7CB-EA2B-4FCD-B162-4D2929BC7B85@weston.borman.com> <544156C7.5040906@isi.edu> <21EB0FC2-EDF7-41CB-A0B5-A632FA78C7E9@weston.borman.com> <54419853.7040705@isi.edu> <08B20814-7C96-469F-AE0D-1B3E7C964322@weston.borman.com> <5441A65F.3080508@isi.edu> <C42C64B2-007E-4DFA-87D6-2C682617FFA3@weston.borman.com> <544593A4.2090108@isi.edu> <96F21A43-3505-4A89-A976-10AF531C32C2@weston.borman.com> <54469C30.9070906@isi.edu> <2F22DAD4-109C-4BF7-9B59-51BBE503F5BC@weston.borman.com> <5446BD67.5030701@isi.edu> <8366E48E-8AE4-424F-A9DB-972D5A37CC5C@weston.borman.com> <5446C9FB.7050006@isi.edu>
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Subject: Re: [tcpm] TCP Four-Way Handshake
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Joe,

A 4WHS RFC could specify precisely how any=20
existing TCP option would be negotiated within a=20
4WHS (ie it would update a number of TCP option RFCs all at once).

Then, if/when 4WHS was implemented, any necessary=20
changes to the negotiation code for each TCP=20
option could be included in the same release as the update to the 4WHS.

You both stated that each TCP option spec defines=20
its own negotiation logic, and that there is no=20
general negotiation rule. But the talking at=20
cross-purposes seemed to go on for dozens of=20
rounds despite this. It seemed to end without any=20
apparent reconciliation - perhaps your keyboards=20
burned out? So I thought the above points might still be useful.

In David's latest update=20
<draft-borman-tcpm-tcp4way-00> he has added a=20
roster of all the existing TCP options and how=20
they are negotiated. This could easily be updated=20
to say how any that need to be changed MUST be negotiated if a 4WHS were=
 used.



Bob

At 21:02 21/10/2014, Joe Touch wrote:


>On 10/21/2014 1:55 PM, David Borman wrote:
> >
> >> On Oct 21, 2014, at 3:09 PM, Joe Touch <touch@isi.edu> wrote:
> >> On 10/21/2014 11:50 AM, David Borman wrote:
> >>>
> >>>> On Oct 21, 2014, at 12:47 PM, Joe Touch <touch@isi.edu> wrote:
> >> ...
> >>>> You're thus proposing NEW requirements on option negotiation, and=
 that
> >>>> ought to be clear in your document as well as considered in=
 comparison
> >>>> to all existing and pending options.
> >>>
> >>> What new requirements?
> >>
> >> That options be negotiated exclusively by appearance in the SYN in each
> >> direction.
> >
> > You're reading things that aren=E2=80=99t there.  There is no one=
 mechanism for
> > enabling use of TCP options.
>
>That is my primary point.
>
> >  Each option defines for itself how it is
> > to be used, including any enablement.  For=20
> many existing options, enablement
> > is defined as the option being present in both a packet sent and=
 received
> > with the SYN bit.  Examples include Window=20
> Scale, Timestamps, and SACK Permitted.
> >
> > That is nothing new, and is not a requirement for any other TCP option.
>
>4WHS won't work when options are not defined as being enabled by the
>option being present in both SYN segments of a 3WHS.
>
> > ...
> >>>> But it has to include *anything* that the client might want to ACK.
> >>>
> >>> No it doesn=E2=80=99t.  It would only include options that it would=
 still
> >>> like to be enabled, but weren=E2=80=99t included in the inbound SYN. =
 Why
> >>> on earth would it offer up to enable options it doesn=E2=80=99t want=
 to use?
> >>
> >> It has to send all the options that might be coming in the client's
> >> SYN/ACK. It has to read minds or be exhaustive.
> > ...
> >
> > Absolutely not.
> >
> > In the 4WHS, for options that are only enabled by their presence in the
> > SYN exchange, the servers SYN/ACK can contain additional options to=
 allow
> > the client to agree to enable them by including them in it=E2=80=99s=
 SYN/ACK.
> > That=E2=80=99s it.  End of story.
>
>Again:
>
>#1      Client sends SYN with A, B ->
>
>#2                              <- Server sends SYN/ACK
>                                 with what? it can send A,B
>                                 what else?
>
>#3      Client sends SYN/ACK with X,Y->
>
>#4                              <- server, never having seen X or Y
>                                 MUST have had to send X AND Y in
>                                 its SYN/ACK
>
>If all options are enabled by occurring in a SYN in both directions, we
>have to also note that #2 is the only SYN(ACK) the server sends.
>
>#2 SYN/ACK has to anticipate the options sent in #3 client SYN/ACK.
>
>How exactly does it do that? Doesn't it have to send ANY option that
>MIGHT be arriving in the client SYN/ACK - but it has to do that BEFORE
>the client sends it.
>
>Joe
>
>_______________________________________________
>tcpm mailing list
>tcpm@ietf.org
>https://www.ietf.org/mailman/listinfo/tcpm

________________________________________________________________
Bob Briscoe,                                                  BT =20


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Joe,

If the OOB segment arrives at a firewall before the SYN, the firewall 
is nearly certain to drop the OOB segment as a non-SYN packet that is 
not part of a connection opened earlier by a SYN.

Even if OOB arrives second, the following conversation implies some 
firewall admins will have included a rule to drop it:
<http://www.linuxquestions.org/questions/linux-security-4/tcp-packet-flags-syn-fin-ack-etc-and-firewall-rules-317389/>

Search for:
"ACK must exist in existing/related connections."
"At least one flag must be set (cannot have none set)."
"A NULL packet is something you should never see"

I know we have to do traversal experiments, but I found this forum 
while looking for something else.



Bob

________________________________________________________________
Bob Briscoe,                                                  BT  


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David,

Thx for updating the draft & filename, and=20
particularly for the adding the roster of=20
existing TCP options. However, to be honest, I=20
think tcp4way has got caught up in following a=20
false trail. Let me explain, and I'll add other critique afterwards too.

1) Following a false trail?

Tcp4way aims to give the client a second chance=20
to negotiate some more TCP options on a=20
subsequent SYN/ACK. The original reason you=20
wanted this (from emails back in the summer) was=20
because it would be really complex to allow a=20
second segment from the client to negotiate more=20
options if the hosts had already made an interim=20
decision after the first round. They might need=20
to revoke some of their earlier decisions, which=20
would be extremely complex and often impossible.

However, all that tcp4way has achieved is to send=20
more segments before making a decision. That=20
means more segments on which TCP options cannot be /used/ yet.

I've calculated how many segments in both=20
directions before a 4WHS server enters the established state:
* Without TFO: 3 segments, including 2 that could carry data
* A resumed connection with TFO & IW=3D3: 7 segments, all potentially with=
 data.
* A resumed connection with TFO & IW=3D10: 21=20
segments, all potentially with data.

So all tcp4way achieves is to prevent TCP options=20
being included on more segments.

I believe it was wrong to conclude (back in the=20
summer) that the problem is all about holding=20
back a move to the established state. Most=20
existing TCP options apply to segments of any=20
kind, irrespective of whether the host is in the=20
established state when it sends them, and=20
irrespective of whether they even carry data.

In all the following examples, if the client=20
wants to support option X and the server can=20
support X, the server would ideally like to start=20
using X on the first SYN/ACK, whether or not the=20
client is offering to delay moving into the established state for a round:
* TCP-AO
* The following sub-options of the tcpcrypt CRYPT=20
option: DECLINE, PKCONF*, NEXTK1*, NEXTK2*,=20
INIT1, INIT2, UNKNOWN, SYNCOOOKIE, ACKCOOKIE, REKEY, REKEYSTREAM.
* the tcpcrypt MAC option
* MPTCP_JOIN, ADD_ADDR(2), DSS, MP_PRIO, MP_FAIL
* SACK permitted
* MSS
* TFO
* TS (if used for RTT calculation)
* UTO

The only ones that are meaningless on the first SYN/ACK are:
* SACK (because the SACK block for zero previous=20
ACKs would be pretty uninteresting. However SACK=20
would be useful on ACKs from the client while still in SYN-RCVD)
* TS (if used for PAWS, 'cos there's no risk of wrap on the first few=
 segments)
* WS (The client can't use a large RWND until it=20
has opened up cwnd thru a few rounds of=20
slow-start. However, this is arguable, because=20
there might be occasions where a connection can safely inherit a large=
 cwnd).

Some option are only meaningful once data has=20
started (e.g. UTO, MPTCP-DSS, WS), but even those=20
can be used prior to connection establishment.

We can't know which of these two lists=20
yet-to-be-defined new options will be in. If the=20
past is a good predictor of the future, the first=20
list is more likely (ie. the ones tcp4way doesn't help much).

2) Latency

Tcp4way always introduces latency. How much depends on which scenario:
* TCP client sends request first (by far the majority of connections): 1 RTT
* TCP server sends data first (e.g. resuming a push connection): 0.5 RTT

As JohnL said, "any added latency is nearly=20
fatal". I strongly agree with every word, except "nearly".

3) Firewalls & Intrusion Detection Systems

Outgoing SYN/ACK:
To a firewall protecting a private network, an=20
outgoing SYN/ACK is as abnormal as an incoming=20
SYN. I guess most firewalls will not have a rule=20
blocking an outgoing SYN/ACK. But given it's=20
abnormal, I'm sure some IDSs will block it.

4W Reserved flag:
I suspect some firewalls / IDSs might drop=20
packets with a Reserved flag set. I suspect we'll=20
see a fair amount of 'correction' of Reserved flags back to zero too.

I appreciate that the 4W flag isn't crucial to your scheme.
But the second SYN/ACK is fundamental.

Even if you don't use a Reserved flag, you've=20
still got to find something 'abnormal' to signal support for the 4WHS.

I don't have data for Reserved flag discard or=20
normalisation. If tcp4way used a TCP option,=20
[Honda] reported %stripped options to the following port numbers:
  port 80: 14%
  port 443: 6%
  port 34343: 4%.
I suspect a lot of the option stripping is due to=20
connection splitters, which won't set options=20
that they don't understand for the ongoing=20
connection. I'm sure they won't set unrecognised=20
Reserved flags either. So I suspect traversal=20
will be just as bad for Reserved flags as for options.

It is particularly important for TCP options that=20
provide security (e.g. TCP-AO, tcpcrypt) to aim=20
for 100% middlebox traversal. If attempts to use=20
security fail so often they become normal, it is=20
too easy for an attacker to target any connection=20
it wants to snoop or alter by stripping options=20
or flags, which just looks like yet another middlebox problem.

I.e. when middlebox interference looks just like=20
a downgrade attack, it's really easy to make a=20
downgrade attack look like middlebox interference.

__________________
They are my 3 main criticisms. The rest are nits...

4) Nit: Reflection of Reserved Flags

Simple reflection of Reserved flags cannot be=20
used to confirm that the server agrees with the=20
client. There are TCP implementations that just=20
reflect whatever Reserved flags the client sends,=20
whether or not they support them.

This is fairly easy to work-round. Similar to ECN=20
negotiation [RFC3168, Sec 6.1.1.2] you just=20
define the confirmation as a different pattern of=20
flags (but this burns more flags).

Richard Scheffenegger was testing the Alexa top=20
1M Web servers in Apr with flags on SYN as follows:
         > CWR ECE SYN
The correct response of ECN-capable servers was:
         <     ECE SYN ACK
I know he found a fair number (sorry I don't have=20
the %) that just dumbly reflected:
         < CWR ECE SYN ACK
And when he set the nonce flag on a SYN sent to=20
these same broken Web servers, all of them just=20
dumbly reflected back the new pattern:
         > NS CWR ECE SYN
         < NS CWR ECE SYN ACK

5) Nit: Remembering EDO support across connections

S.7.3.2.1 says that server EDO support can be=20
saved alongside a TFO cookie. I don't believe=20
this is safe. During an upgrade, there might be=20
some server replicas that support EDO and others=20
that don't. The legacy servers would then ignore=20
the EDO option, but still pass TCP options as a corrupt payload to the app.

[BTW, it is safe for a client to hold solely the=20
TFO capability of a server in a cookie, because=20
(unlike EDO) TFO doesn't put data that is not=20
intended for the app in the payload. So, if a=20
replica server that picks up the resumed=20
connection does not support TFO, it will simply=20
ignore the cookie and just hold the data back until the 3WHS completes.]

6) Nit: Unknown Option

Altho it would be nice to think that this might=20
help, there doesn't seem to be a case where it=20
will. Can you give an example where the Unknown Options is useful?

7) Nit: T/TCP is obsolete.

T/TCP had vulnerabilities and was obsoleted by RFC6247.



Bob



At 16:21 24/10/2014, David Borman wrote:
>I=E2=80=99ve just posted an update to the TCP Four-Way handshake document.
>
>1) I changed the name of the document to include=20
>tcpm: draft-dborman-tcpm-tcp4way-00.txt
>
>2) I=E2=80=99ve added text that any implementation=20
>that indicates support for the Four-Way=20
>handshake is also indicating that it properly=20
>handles unknown TCP options, as per RFC=20
>1122.  We keep dancing around that issue, and=20
>this gives us a way to stop worrying about it anymore, and
>a way for an implementation to know for sure=20
>that it isn=E2=80=99t talking to a legacy TCP implementation.
>
>3) I=E2=80=99ve added a preliminary section to discuss specific TCP options
>
>4) I=E2=80=99ve added text about the TS option to=20
>clarify RFC 7323, that it is the presence of=20
>TSopt in both a sent and received packet with=20
>the SYN bit that enables the option, i.e. it is=20
>not limited to the specific instance of the=20
><SYN> and <SYN,ACK> packets of the three-way handshake.
>
>
>And something entirely new:
>
>5) I=E2=80=99ve added a new TCP option, the Unknown=20
>Option option.  Currently, the primary way to=20
>determine that the other side of the connection=20
>doesn=E2=80=99t support a specific option is by lack=20
>of response to that option, because it silently=20
>ignores it.  The Unknown Option option provides=20
>a mechanism where a TCP can inform the other=20
>side that it does not understand or support any=20
>given option for a connection.  This can be very=20
>useful for sending TCP options in non-SYN=20
>packets, especially if the option is not=20
>explicitly enabled through the SYN exchange, or=20
>it is a one-way option.  UTO comes to mind.
>
>The document needs more work, but I wanted to=20
>get an update out today, before Mondays submission cutoff.
>
>                         -David Borman
>
> > On Oct 24, 2014, at 11:01 AM, Internet-Drafts@ietf.org wrote:
> >
> >
> > A New Internet-Draft is available from the=20
> on-line Internet-Drafts directories.
> >
> >
> >        Title           : TCP Four-Way Handshake
> >        Author          : David Borman
> >       Filename        : draft-borman-tcpm-tcp4way-00.txt
> >       Pages           : 21
> >       Date            : 2014-10-24
> >
> > Abstract:
> >   One of the limitations of TCP is that it has limited space for TCP
> >   options, only 40 bytes.  Many mechanisms have been proposed for for
> >   extending the TCP option space, but the biggest challenge has been to
> >   get additional option space in the initial SYN packet.
> >
> >   This memo presents a optional four-way TCP handshake to allow
> >   extended option space to be used in SYN packets in both directions.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-borman-tcpm-tcp4way/
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-borman-tcpm-tcp4way-00
> >
> >
> > 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.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>_______________________________________________
>tcpm mailing list
>tcpm@ietf.org
>https://www.ietf.org/mailman/listinfo/tcpm

________________________________________________________________
Bob Briscoe,                                                  BT=20
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<body>
David, <br><br>
Thx for updating the draft &amp; filename, and particularly for the
adding the roster of existing TCP options. However, to be honest, I think
tcp4way has got caught up in following a false trail. Let me explain, and
I'll add other critique afterwards too.<br><br>
<b><u>1) Following a false trail?<br><br>
</u></b>Tcp4way aims to give the client a second chance to negotiate some
more TCP options on a subsequent SYN/ACK. The original reason you wanted
this (from emails back in the summer) was because it would be really
complex to allow a second segment from the client to negotiate more
options if the hosts had already made an interim decision after the first
round. They might need to revoke some of their earlier decisions, which
would be extremely complex and often impossible.<br><br>
However, all that tcp4way has achieved is to send more segments before
making a decision. That means more segments on which TCP options cannot
be /used/ yet. <br><br>
I've calculated how many segments in both directions before a 4WHS server
enters the established state:<br>
* Without TFO: 3 segments, including 2 that could carry data<br>
* A resumed connection with TFO &amp; IW=3D3: 7 segments, all potentially
with data.<br>
* A resumed connection with TFO &amp; IW=3D10: 21 segments, all potentially
with data.<br><br>
So all tcp4way achieves is to prevent TCP options being included on more
segments. <br><br>
I believe it was wrong to conclude (back in the summer) that the problem
is all about holding back a move to the established state. Most existing
TCP options apply to segments of any kind, irrespective of whether the
host is in the established state when it sends them, and irrespective of
whether they even carry data. <br><br>
In all the following examples, if the client wants to support option X
and the server can support X, the server would ideally like to start
using X on the first SYN/ACK, whether or not the client is offering to
delay moving into the established state for a round:<br>
* TCP-AO<br>
* The following sub-options of the tcpcrypt CRYPT option: DECLINE,
PKCONF*, NEXTK1*, NEXTK2*, INIT1, INIT2, UNKNOWN, SYNCOOOKIE, ACKCOOKIE,
REKEY, REKEYSTREAM.<br>
* the tcpcrypt MAC option<br>
* MPTCP_JOIN, ADD_ADDR(2), DSS, MP_PRIO, MP_FAIL<br>
* SACK permitted<br>
* MSS<br>
* TFO <br>
* TS (if used for RTT calculation)<br>
* UTO<br><br>
The only ones that are meaningless on the first SYN/ACK are:<br>
* SACK (because the SACK block for zero previous ACKs would be pretty
uninteresting. However SACK would be useful on ACKs from the client while
still in SYN-RCVD) <br>
* TS (if used for PAWS, 'cos there's no risk of wrap on the first few
segments)<br>
* WS (The client can't use a large RWND until it has opened up cwnd thru
a few rounds of slow-start. However, this is arguable, because there
might be occasions where a connection can safely inherit a large
cwnd).<br><br>
Some option are only meaningful once data has started (e.g. UTO,
MPTCP-DSS, WS), but even those can be used prior to connection
establishment.<br><br>
We can't know which of these two lists yet-to-be-defined new options will
be in. If the past is a good predictor of the future, the first list is
more likely (ie. the ones tcp4way doesn't help much).<br><br>
<b><u>2) Latency<br><br>
</u></b>Tcp4way always introduces latency. How much depends on which
scenario:<br>
* TCP client sends request first (by far the majority of connections): 1
RTT<br>
* TCP server sends data first (e.g. resuming a push connection): 0.5
RTT<br><br>
As JohnL said, &quot;any added latency is nearly fatal&quot;. I strongly
agree with every word, except &quot;nearly&quot;.<br><br>
<b><u>3) Firewalls &amp; Intrusion Detection Systems<br><br>
</u></b>Outgoing SYN/ACK: <br>
To a firewall protecting a private network, an outgoing SYN/ACK is as
abnormal as an incoming SYN. I guess most firewalls will not have a rule
blocking an outgoing SYN/ACK. But given it's abnormal, I'm sure some IDSs
will block it.<br><br>
4W Reserved flag:<br>
I suspect some firewalls / IDSs might drop packets with a Reserved flag
set. I suspect we'll see a fair amount of 'correction' of Reserved flags
back to zero too.<br><br>
I appreciate that the 4W flag isn't crucial to your scheme.<br>
But the second SYN/ACK is fundamental. <br><br>
Even if you don't use a Reserved flag, you've still got to find something
'abnormal' to signal support for the 4WHS.<br><br>
I don't have data for Reserved flag discard or normalisation. If tcp4way
used a TCP option, [Honda] reported %stripped options to the following
port numbers:<br>
&nbsp;port 80: 14% <br>
&nbsp;port 443: 6% <br>
&nbsp;port 34343: 4%.<br>
I suspect a lot of the option stripping is due to connection splitters,
which won't set options that they don't understand for the ongoing
connection. I'm sure they won't set unrecognised Reserved flags either.
So I suspect traversal will be just as bad for Reserved flags as for
options.<br><br>
It is particularly important for TCP options that provide security (e.g.
TCP-AO, tcpcrypt) to aim for 100% middlebox traversal. If attempts to use
security fail so often they become normal, it is too easy for an attacker
to target any connection it wants to snoop or alter by stripping options
or flags, which just looks like yet another middlebox problem. <br><br>
I.e. when middlebox interference looks just like a downgrade attack, it's
really easy to make a downgrade attack look like middlebox
interference.<br><br>
__________________<br>
They are my 3 main criticisms. The rest are nits...<br><br>
<b><u>4) Nit: Reflection of Reserved Flags<br><br>
</u></b>Simple reflection of Reserved flags cannot be used to confirm
that the server agrees with the client. There are TCP implementations
that just reflect whatever Reserved flags the client sends, whether or
not they support them.<br><br>
This is fairly easy to work-round. Similar to ECN negotiation [RFC3168,
Sec 6.1.1.2] you just define the confirmation as a different pattern of
flags (but this burns more flags).<br><br>
Richard Scheffenegger was testing the Alexa top 1M Web servers in Apr
with flags on SYN as follows:<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>&gt; CWR
ECE SYN <br>
The correct response of ECN-capable servers was:<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>
&lt;&nbsp;&nbsp;&nbsp;&nbsp; ECE SYN ACK<br>
I know he found a fair number (sorry I don't have the %) that just dumbly
reflected:<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>&lt; CWR
ECE SYN ACK<br>
And when he set the nonce flag on a SYN sent to these same broken Web
servers, all of them just dumbly reflected back the new pattern:<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>&gt; NS
CWR ECE SYN <br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>&lt; NS
CWR ECE SYN ACK<br><br>
<b><u>5) Nit: Remembering EDO support across connections<br><br>
</u></b>S.7.3.2.1 says that server EDO support can be saved alongside a
TFO cookie. I don't believe this is safe. During an upgrade, there might
be some server replicas that support EDO and others that don't. The
legacy servers would then ignore the EDO option, but still pass TCP
options as a corrupt payload to the app.<br><br>
[BTW, it is safe for a client to hold solely the TFO capability of a
server in a cookie, because (unlike EDO) TFO doesn't put data that is not
intended for the app in the payload. So, if a replica server that picks
up the resumed connection does not support TFO, it will simply ignore the
cookie and just hold the data back until the 3WHS completes.]<br><br>
<b><u>6) Nit: Unknown Option<br><br>
</u></b>Altho it would be nice to think that this might help, there
doesn't seem to be a case where it will. Can you give an example where
the Unknown Options is useful?<br><br>
<b><u>7) Nit: T/TCP is obsolete.<br><br>
</u></b>T/TCP had vulnerabilities and was obsoleted by RFC6247.<br><br>
<br><br>
Bob<br><br>
<br><br>
At 16:21 24/10/2014, David Borman wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">I=E2=80=99ve just posted an=
 update to
the TCP Four-Way handshake document.<br><br>
1) I changed the name of the document to include tcpm:
draft-dborman-tcpm-tcp4way-00.txt<br><br>
2) I=E2=80=99ve added text that any implementation that indicates support fo=
r
the Four-Way handshake is also indicating that it properly handles
unknown TCP options, as per RFC 1122.&nbsp; We keep dancing around that
issue, and this gives us a way to stop worrying about it anymore,
and<br>
a way for an implementation to know for sure that it isn=E2=80=99t talking=
 to a
legacy TCP implementation.<br><br>
3) I=E2=80=99ve added a preliminary section to discuss specific TCP
options<br><br>
4) I=E2=80=99ve added text about the TS option to clarify RFC 7323, that it =
is
the presence of TSopt in both a sent and received packet with the SYN bit
that enables the option, i.e. it is not limited to the specific instance
of the &lt;SYN&gt; and &lt;SYN,ACK&gt; packets of the three-way
handshake.<br><br>
<br>
And something entirely new:<br><br>
5) I=E2=80=99ve added a new TCP option, the Unknown Option option.&nbsp;
Currently, the primary way to determine that the other side of the
connection doesn=E2=80=99t support a specific option is by lack of response =
to
that option, because it silently ignores it.&nbsp; The Unknown Option
option provides a mechanism where a TCP can inform the other side that it
does not understand or support any given option for a connection.&nbsp;
This can be very useful for sending TCP options in non-SYN packets,
especially if the option is not explicitly enabled through the SYN
exchange, or it is a one-way option.&nbsp; UTO comes to mind.<br><br>
The document needs more work, but I wanted to get an update out today,
before Mondays submission cutoff.<br><br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>-David
Borman<br><br>
&gt; On Oct 24, 2014, at 11:01 AM, Internet-Drafts@ietf.org wrote:<br>
&gt; <br>
&gt; <br>
&gt; A New Internet-Draft is available from the on-line Internet-Drafts
directories.<br>
&gt; <br>
&gt; <br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : TCP
Four-Way Handshake<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Author&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : David
Borman<br>
&gt;
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
draft-borman-tcpm-tcp4way-00.txt<br>
&gt;
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>
Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
21<br>
&gt;
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>
Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
2014-10-24<br>
&gt; <br>
&gt; Abstract:<br>
&gt;&nbsp;&nbsp; One of the limitations of TCP is that it has limited
space for TCP<br>
&gt;&nbsp;&nbsp; options, only 40 bytes.&nbsp; Many mechanisms have been
proposed for for<br>
&gt;&nbsp;&nbsp; extending the TCP option space, but the biggest
challenge has been to<br>
&gt;&nbsp;&nbsp; get additional option space in the initial SYN
packet.<br>
&gt; <br>
&gt;&nbsp;&nbsp; This memo presents a optional four-way TCP handshake to
allow<br>
&gt;&nbsp;&nbsp; extended option space to be used in SYN packets in both
directions.<br>
&gt; <br>
&gt; <br>
&gt; The IETF datatracker status page for this draft is:<br>
&gt;
<a href=3D"https://datatracker.ietf.org/doc/draft-borman-tcpm-tcp4way/" eudo=
ra=3D"autourl">
https://datatracker.ietf.org/doc/draft-borman-tcpm-tcp4way/</a><br>
&gt; <br>
&gt; There's also a htmlized version available at:<br>
&gt;
<a href=3D"http://tools.ietf.org/html/draft-borman-tcpm-tcp4way-00" eudora=
=3D"autourl">
http://tools.ietf.org/html/draft-borman-tcpm-tcp4way-00</a><br>
&gt; <br>
&gt; <br>
&gt; Please note that it may take a couple of minutes from the time of
submission<br>
&gt; until the htmlized version and diff are available at
tools.ietf.org.<br>
&gt; <br>
&gt; Internet-Drafts are also available by anonymous FTP at:<br>
&gt;
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" eudora=3D"autourl">
ftp://ftp.ietf.org/internet-drafts/</a><br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; I-D-Announce mailing list<br>
&gt; I-D-Announce@ietf.org<br>
&gt;
<a href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce" eudora=3D"aut=
ourl">
https://www.ietf.org/mailman/listinfo/i-d-announce</a><br>
&gt; Internet-Draft directories:
<a href=3D"http://www.ietf.org/shadow.html" eudora=3D"autourl">
http://www.ietf.org/shadow.html</a><br>
&gt; or
<a href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" eudora=3D"autourl">
ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><br><br>
_______________________________________________<br>
tcpm mailing list<br>
tcpm@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" eudora=3D"autourl">
https://www.ietf.org/mailman/listinfo/tcpm</a></blockquote>
<x-sigsep><p></x-sigsep>
________________________________________________________________<br>
Bob
Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
BT</body>
</html>

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Yoshifumi,

I've just sent a critique of tcp4way to David &=20
the list. I've pointed out problems with many of=20
the things you say you like about it inline below...

At 04:34 22/10/2014, Yoshifumi Nishida wrote:
>Hi David,
>
>(as an individual of course..)
>I personally like this idea from the following reasons.
>
>1: use SYN-ACK to initiate feature negotiation=20
>(as discussed in the previous thread)

tcp4way just seems to prevent use of TCP options=20
on more segments (and for more rounds).


>2: it can expand SYN options space to 80 octets=20
>without overriding DO field. It might be more middlebox friendly.

I agree that if options are going to be placed=20
beyond DO, and /if/ the signal that says they are=20
there could be stripped or normalised by=20
middleboxes, it is important that the signal that=20
says they are there is ultra-safe - checked and=20
double-checked for middlebox interference.

However, deferring the point at which a=20
connection moves to the established state doesn't=20
suddenly mean that the segments sent before this didn't need any TCP=
 options.

Also:
* Some IDSs and perhaps some stateful firewalls=20
might not like the abnormal extra SYN/ACK.
* The segments have to be abnormal (e.g. 4W flag=20
or a TCP option) in order to signal 4WHS support.=20
Your [Honda] study found option stripping was far=20
from a minor effect, probably due to split=20
connections - a split connection would also strip a Reserved flag.


>3: we might be able to re-design syn cookie=20
>solution that can address the previous issues.

What previous issues?


>4: even if an option in SYN/ACK has been removed=20
>by middleboxes, it can reliably find this situation.

So can any solution that is prepared to add as much latency as the 4WHS.


>5: in mptcp, initiating subflow requires 4=20
>packet exchanges. With 4-way handshake=20
>mechanism, we might be able to do this with non-mptcp-specific way.=C2

There seems little point in doing a 4-way=20
exchange in a non-mptcp-specific way if it still=20
has the same latency problem. There is certainly=20
little point if another solution is available=20
that adds no latency and traverses middleboxes.


>6: similar to 5, it might be useful some secure=20
>negotiations where we don't want to put=20
>something that requires computations in SYN, but=20
>want to perform some cryptic algorithms in later packet exchanges.

Any specific TCP security option can always=20
require that computation is deferred, without=20
making a 4WHS the /only/ way to extend option space for all TCP options.


Nonetheless, I do appreciate you posting your=20
views as an individual, so that we can have a discussion about these points.

Cheers




Bob



>(as one of co-chairs..)
>
>When you submit new version, please include=20
>-tcpm- in the draft's name so that we can track it easily.
>
>Regards,
>--
>Yoshi

At 04:34 22/10/2014, Yoshifumi Nishida wrote:
>Hi David,
>
>(as an individual of course..)
>I personally like this idea from the following reasons.
>
>1: use SYN-ACK to initiate feature negotiation=20
>(as discussed in the previous thread)
>
>2: it can expand SYN options space to 80 octets=20
>without overriding DO field. It might be more middlebox friendly.
>
>3: we might be able to re-design syn cookie=20
>solution that can address the previous issues.
>
>4: even if an option in SYN/ACK has been removed=20
>by middleboxes, it can reliably find this situation.
>
>5: in mptcp, initiating subflow requires 4=20
>packet exchanges. With 4-way handshake=20
>mechanism, we might be able to do this with non-mptcp-specific way.=C2
>
>6: similar to 5, it might be useful some secure=20
>negotiations where we don't want to put=20
>something that requires computations in SYN, but=20
>want to perform some cryptic algorithms in later packet exchanges.
>
>
>(as one of co-chairs..)
>
>When you submit new version, please include=20
>-tcpm- in the draft's name so that we can track it easily.
>
>Regards,
>--
>Yoshi
>
>
>
>On Tue, Oct 14, 2014 at 2:44 PM, David Borman=20
><<mailto:dab@weston.borman.com>dab@weston.borman.com> wrote:
>Hi,
>
>I=E2=80=99ve just posted draft-borman-tcp4way-00.txt,=20
>which describes a 4-Way TCP handshake that is=20
>intended to allow the TCP EDO option to be used=20
>during the initial SYN exchange.=C2  This was=20
>mentioned during the TCPM WG at the last IETF=20
>meeting.=C2  Please see the draft for more details.
>
>=C2  =C2  =C2  =C2  =C2  =C2  =C2  =C2  =C2  =C2  =C2  =C2  -David Borman
>_______________________________________________
>tcpm mailing list
><mailto:tcpm@ietf.org>tcpm@ietf.org
>https://www.ietf.org/mailman/listinfo/tcpm
>
>
>_______________________________________________
>tcpm mailing list
>tcpm@ietf.org
>https://www.ietf.org/mailman/listinfo/tcpm

________________________________________________________________
Bob Briscoe,                                                  BT=20
--=====================_878517087==.ALT
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<body>
Yoshifumi,<br><br>
I've just sent a critique of tcp4way to David &amp; the list. I've
pointed out problems with many of the things you say you like about it
inline below...<br><br>
At 04:34 22/10/2014, Yoshifumi Nishida wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">Hi David,<br><br>
(as an individual of course..)<br>
I personally like this idea from the following reasons.<br><br>
1: use SYN-ACK to initiate feature negotiation (as discussed in the
previous thread)</blockquote><br>
tcp4way just seems to prevent use of TCP options on more segments (and
for more rounds).<br><br>
<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">2: it can expand SYN options
space to 80 octets without overriding DO field. It might be more
middlebox friendly.</blockquote><br>
I agree that if options are going to be placed beyond DO, and /if/ the
signal that says they are there could be stripped or normalised by
middleboxes, it is important that the signal that says they are there is
ultra-safe - checked and double-checked for middlebox interference.
<br><br>
However, deferring the point at which a connection moves to the
established state doesn't suddenly mean that the segments sent before
this didn't need any TCP options.<br><br>
Also:<br>
* Some IDSs and perhaps some stateful firewalls might not like the
abnormal extra SYN/ACK.<br>
* The segments have to be abnormal (e.g. 4W flag or a TCP option) in
order to signal 4WHS support. Your [Honda] study found option stripping
was far from a minor effect, probably due to split connections - a split
connection would also strip a Reserved flag.<br><br>
<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">3: we might be able to re-des=
ign
syn cookie solution that can address the previous
issues.</blockquote><br>
What previous issues?<br><br>
<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">4: even if an option in SYN/A=
CK
has been removed by middleboxes, it can reliably find this
situation.</blockquote><br>
So can any solution that is prepared to add as much latency as the
4WHS.<br><br>
<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">5: in mptcp, initiating subfl=
ow
requires 4 packet exchanges. With 4-way handshake mechanism, we might be
able to do this with non-mptcp-specific way.=C2 </blockquote><br>
There seems little point in doing a 4-way exchange in a
non-mptcp-specific way if it still has the same latency problem. There is
certainly little point if another solution is available that adds no
latency and traverses middleboxes.<br><br>
<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">6: similar to 5, it might be
useful some secure negotiations where we don't want to put something that
requires computations in SYN, but want to perform some cryptic algorithms
in later packet exchanges.</blockquote><br>
Any specific TCP security option can always require that computation is
deferred, without making a 4WHS the /only/ way to extend option space for
all TCP options.<br><br>
<br>
Nonetheless, I do appreciate you posting your views as an individual, so
that we can have a discussion about these points.<br><br>
Cheers<br><br>
<br><br>
<br>
Bob<br><br>
<br><br>
<blockquote type=3Dcite class=3Dcite cite=3D"">(as one of=
 co-chairs..)<br><br>
When you submit new version, please include -tcpm- in the draft's name so
that we can track it easily.<br><br>
Regards,<br>
--<br>
Yoshi</blockquote><br>
At 04:34 22/10/2014, Yoshifumi Nishida wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">Hi David,<br><br>
(as an individual of course..)<br>
I personally like this idea from the following reasons.<br><br>
1: use SYN-ACK to initiate feature negotiation (as discussed in the
previous thread)<br><br>
2: it can expand SYN options space to 80 octets without overriding DO
field. It might be more middlebox friendly.<br><br>
3: we might be able to re-design syn cookie solution that can address the
previous issues.<br><br>
4: even if an option in SYN/ACK has been removed by middleboxes, it can
reliably find this situation.<br><br>
5: in mptcp, initiating subflow requires 4 packet exchanges. With 4-way
handshake mechanism, we might be able to do this with non-mptcp-specific
way.=C2 <br><br>
6: similar to 5, it might be useful some secure negotiations where we
don't want to put something that requires computations in SYN, but want
to perform some cryptic algorithms in later packet exchanges.<br><br>
<br>
(as one of co-chairs..)<br><br>
When you submit new version, please include -tcpm- in the draft's name so
that we can track it easily.<br><br>
Regards,<br>
--<br>
Yoshi<br><br>
<br><br>
On Tue, Oct 14, 2014 at 2:44 PM, David Borman
&lt;<a href=3D"mailto:dab@weston.borman.com">dab@weston.borman.com</a>&gt;
wrote:<br>

<dl>
<dd>Hi,<br><br>

<dd>I=E2=80=99ve just posted draft-borman-tcp4way-00.txt, which describes a
4-Way TCP handshake that is intended to allow the TCP EDO option to be
used during the initial SYN exchange.=C2&nbsp; This was mentioned during
the TCPM WG at the last IETF meeting.=C2&nbsp; Please see the draft for
more details.<br>
<font color=3D"#888888"><br>

<dd>=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; =
=C2&nbsp;
=C2&nbsp; =C2&nbsp; =C2&nbsp; =C2&nbsp; -David Borman<br>

<dd>_______________________________________________<br>

<dd>tcpm mailing list<br>

<dd><a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>

<dd><a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" eudora=3D"autourl=
">
https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</font><br>

</dl><br>
_______________________________________________<br>
tcpm mailing list<br>
tcpm@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" eudora=3D"autourl">
https://www.ietf.org/mailman/listinfo/tcpm</a></blockquote>
<x-sigsep><p></x-sigsep>
________________________________________________________________<br>
Bob
Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
BT</body>
</html>

--=====================_878517087==.ALT--


From nobody Mon Nov  3 16:22:53 2014
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Date: Mon, 03 Nov 2014 16:22:29 -0800
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Firewall rules and OOB
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On 11/3/2014 3:15 PM, Bob Briscoe wrote:
> Joe,
> 
> If the OOB segment arrives at a firewall before the SYN, the firewall is
> nearly certain to drop the OOB segment as a non-SYN packet that is not
> part of a connection opened earlier by a SYN.

Yes, as with any such segment.

> Even if OOB arrives second, the following conversation implies some
> firewall admins will have included a rule to drop it:
> <http://www.linuxquestions.org/questions/linux-security-4/tcp-packet-flags-syn-fin-ack-etc-and-firewall-rules-317389/>

That depends on whether they implement those rules, or that's just "how
they understand it".

And, if those rules are accurate, there are lots of other valid packets
that are not permitted either (e.g., SYN with URG, PSH).

> Search for:
> "ACK must exist in existing/related connections."
> "At least one flag must be set (cannot have none set)."
> "A NULL packet is something you should never see"
> 
> I know we have to do traversal experiments, but I found this forum while
> looking for something else.

Agreed, but as you note it's just one forum.

This is a great example of why NAT behavior must be modeled correctly,
not just derived from blind rules.

Joe


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Subject: Re: [tcpm] TCP Four-Way Handshake
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On 11/3/2014 3:12 PM, Bob Briscoe wrote:
> Joe,
> 
> A 4WHS RFC could specify precisely how any existing TCP option would be
> negotiated within a 4WHS (ie it would update a number of TCP option RFCs
> all at once).
> 
> Then, if/when 4WHS was implemented, any necessary changes to the
> negotiation code for each TCP option could be included in the same
> release as the update to the 4WHS.

Yes, but it might limit how such options can be negotiated. That's the
sense in which neither I nor Dave wants to have 4WHS include such limits.

Joe


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Hi Bob,

On Mon, Nov 3, 2014 at 3:43 PM, Bob Briscoe <bob.briscoe@bt.com> wrote:

>  Yoshifumi,
>
> I've just sent a critique of tcp4way to David & the list. I've pointed ou=
t
> problems with many of the things you say you like about it inline below..=
.
>
> At 04:34 22/10/2014, Yoshifumi Nishida wrote:
>
> Hi David,
>
> (as an individual of course..)
> I personally like this idea from the following reasons.
>
> 1: use SYN-ACK to initiate feature negotiation (as discussed in the
> previous thread)
>
>
> tcp4way just seems to prevent use of TCP options on more segments (and fo=
r
> more rounds).
>

Hmm. Sorry, I'm not very sure if I understand this.

> 2: it can expand SYN options space to 80 octets without overriding DO
> field. It might be more middlebox friendly.
>
>
> I agree that if options are going to be placed beyond DO, and /if/ the
> signal that says they are there could be stripped or normalised by
> middleboxes, it is important that the signal that says they are there is
> ultra-safe - checked and double-checked for middlebox interference.
>
> However, deferring the point at which a connection moves to the
> established state doesn't suddenly mean that the segments sent before thi=
s
> didn't need any TCP options.
>
> Also:
> * Some IDSs and perhaps some stateful firewalls might not like the
> abnormal extra SYN/ACK.
> * The segments have to be abnormal (e.g. 4W flag or a TCP option) in orde=
r
> to signal 4WHS support. Your [Honda] study found option stripping was far
> from a minor effect, probably due to split connections - a split connecti=
on
> would also strip a Reserved flag.
>

Yep. You got a point and we know the discussions for middlebox is always
tricky.
I know this is not a very strong argument, but if we use payload as option
space, we'll need extra mechanisms to avoid being parsed wrongly, which
will increase complexity.


> 3: we might be able to re-design syn cookie solution that can address the
> previous issues.
>
> What previous issues?
>

In SYN cookie, tcp cannot use TCP options other than mss. Also, if the
third ack is lost, the connection might be frozen because the server cannot
retransmit SYN/ACK.


> 4: even if an option in SYN/ACK has been removed by middleboxes, it can
> reliably find this situation.
>
>
> So can any solution that is prepared to add as much latency as the 4WHS.
>

Yep. maybe. but, not sure about complexity.

> 5: in mptcp, initiating subflow requires 4 packet exchanges. With 4-way
> handshake mechanism, we might be able to do this with non-mptcp-specific
> way.=C3=82
>
>
> There seems little point in doing a 4-way exchange in a non-mptcp-specifi=
c
> way if it still has the same latency problem. There is certainly little
> point if another solution is available that adds no latency and traverses
> middleboxes.
>

Well, it may be a very tiny point, but, the third ACK can be pure ACK.
So, in order to perform 4-way exchange,  mptcp may need to violate a basic
rule in TCP, which is "don't send ack for ack"
I personally don't want to see this kind of adhoc logic if possible.

> 6: similar to 5, it might be useful some secure negotiations where we
> don't want to put something that requires computations in SYN, but want t=
o
> perform some cryptic algorithms in later packet exchanges.
>
>
> Any specific TCP security option can always require that computation is
> deferred, without making a 4WHS the /only/ way to extend option space for
> all TCP options.
>

Yep. again, you maybe right, but not sure about complexity.
A problem in 3WHS is there's no good way to retransmit the third ACK.
We might be able to solve it with additional logic, but at the same time, I
would like to see something simpler.

I think I'm focusing on complexity rather than latency.
I'm not very sure if this is right or not, but I believe complexity is also
an important factor.

However, I cannot and don't want to say 4WHS is less complex than others
for now, it will be required more detailed analysis and comparisons.
I just intuitively thought it looks like that. I might be wrong. I just
would like to say we need more analysis.

Thanks,
--
Yoshi

--001a11c123dec8dac6050704d325
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Bob,<div><div class=3D"gmail_extra"><br><div class=3D"g=
mail_quote">On Mon, Nov 3, 2014 at 3:43 PM, Bob Briscoe <span dir=3D"ltr">&=
lt;<a href=3D"mailto:bob.briscoe@bt.com" target=3D"_blank">bob.briscoe@bt.c=
om</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
Yoshifumi,<br><br>
I&#39;ve just sent a critique of tcp4way to David &amp; the list. I&#39;ve
pointed out problems with many of the things you say you like about it
inline below...<span><br><br>
At 04:34 22/10/2014, Yoshifumi Nishida wrote:<br>
<blockquote type=3D"cite">Hi David,<br><br>
(as an individual of course..)<br>
I personally like this idea from the following reasons.<br><br>
1: use SYN-ACK to initiate feature negotiation (as discussed in the
previous thread)</blockquote><br></span>
tcp4way just seems to prevent use of TCP options on more segments (and
for more rounds).</div></blockquote><div><br></div><div>Hmm. Sorry, I&#39;m=
 not very sure if I understand this.=C2=A0</div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div><span>
<blockquote type=3D"cite">2: it can expand SYN options
space to 80 octets without overriding DO field. It might be more
middlebox friendly.</blockquote><br></span>
I agree that if options are going to be placed beyond DO, and /if/ the
signal that says they are there could be stripped or normalised by
middleboxes, it is important that the signal that says they are there is
ultra-safe - checked and double-checked for middlebox interference.
<br><br>
However, deferring the point at which a connection moves to the
established state doesn&#39;t suddenly mean that the segments sent before
this didn&#39;t need any TCP options.<br><br>
Also:<br>
* Some IDSs and perhaps some stateful firewalls might not like the
abnormal extra SYN/ACK.<br>
* The segments have to be abnormal (e.g. 4W flag or a TCP option) in
order to signal 4WHS support. Your [Honda] study found option stripping
was far from a minor effect, probably due to split connections - a split
connection would also strip a Reserved flag.</div></blockquote><div><br></d=
iv><div>Yep. You got a point and we know the discussions for middlebox is a=
lways tricky.</div><div>I know this is not a very strong argument, but if w=
e use payload as option space, we&#39;ll need extra mechanisms to avoid bei=
ng parsed wrongly, which will increase complexity.=C2=A0</div><div>=C2=A0</=
div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-lef=
t:1px #ccc solid;padding-left:1ex"><div><span>
<blockquote type=3D"cite">3: we might be able to re-design
syn cookie solution that can address the previous
issues.</blockquote></span>
What previous issues?</div></blockquote><div><br></div><div>In SYN cookie, =
tcp cannot use TCP options other than mss. Also, if the third ack is lost, =
the connection might be frozen because the server cannot retransmit SYN/ACK=
.</div><div>=C2=A0<br></div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><span><bloc=
kquote type=3D"cite">4: even if an option in SYN/ACK
has been removed by middleboxes, it can reliably find this
situation.</blockquote><br></span>
So can any solution that is prepared to add as much latency as the
4WHS.<br></div></blockquote><div><br></div><div>Yep. maybe. but, not sure a=
bout complexity.</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><blockquote type=
=3D"cite">5: in mptcp, initiating subflow
requires 4 packet exchanges. With 4-way handshake mechanism, we might be
able to do this with non-mptcp-specific way.=C3=82 </blockquote><br>
There seems little point in doing a 4-way exchange in a
non-mptcp-specific way if it still has the same latency problem. There is
certainly little point if another solution is available that adds no
latency and traverses middleboxes.</div></blockquote><div><br></div><div>We=
ll, it may be a very tiny point, but, the third ACK can be pure ACK.=C2=A0<=
/div><div>So, in order to perform 4-way exchange, =C2=A0mptcp may need to v=
iolate a basic rule in TCP, which is &quot;don&#39;t send ack for ack&quot;=
</div><div>I personally don&#39;t want to see this kind of adhoc logic if p=
ossible.</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div><span>
<blockquote type=3D"cite">6: similar to 5, it might be
useful some secure negotiations where we don&#39;t want to put something th=
at
requires computations in SYN, but want to perform some cryptic algorithms
in later packet exchanges.</blockquote><br></span>
Any specific TCP security option can always require that computation is
deferred, without making a 4WHS the /only/ way to extend option space for
all TCP options.<br></div></blockquote><div><br></div><div>Yep. again, you =
maybe right, but not sure about complexity.</div><div>A problem in 3WHS is =
there&#39;s no good way to retransmit the third ACK.=C2=A0</div><div>We mig=
ht be able to solve it with additional logic, but at the same time, I would=
 like to see something simpler.</div><div><br></div><div>I think I&#39;m fo=
cusing on complexity rather than latency.=C2=A0</div><div>I&#39;m not very =
sure if this is right or not, but I believe complexity is also an important=
 factor.=C2=A0</div><div><br></div><div>However, I cannot and don&#39;t wan=
t to say 4WHS is less complex than others for now, it will be required more=
 detailed analysis and comparisons.</div><div>I just intuitively thought it=
 looks like that. I might be wrong. I just would like to say we need more a=
nalysis.</div><div><br></div><div>Thanks,</div><div>--</div><div>Yoshi</div=
></div></div></div></div>

--001a11c123dec8dac6050704d325--


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Yoshifumi,

At 09:09 04/11/2014, Yoshifumi Nishida wrote:
>Hi Bob,
>
>On Mon, Nov 3, 2014 at 3:43 PM, Bob Briscoe=20
><<mailto:bob.briscoe@bt.com>bob.briscoe@bt.com> wrote:
>Yoshifumi,
>
>I've just sent a critique of tcp4way to David &=20
>the list. I've pointed out problems with many of=20
>the things you say you like about it inline below...
>
>At 04:34 22/10/2014, Yoshifumi Nishida wrote:
>>Hi David,
>>
>>(as an individual of course..)
>>I personally like this idea from the following reasons.
>>
>>1: use SYN-ACK to initiate feature negotiation=20
>>(as discussed in the previous thread)
>
>tcp4way just seems to prevent use of TCP options=20
>on more segments (and for more rounds).
>
>
>Hmm. Sorry, I'm not very sure if I understand this.=C2

See email to Dave Borman yesterday.
<http://www.ietf.org/mail-archive/web/tcpm/current/msg09322.html>
Look for my first point: "Following a false trail?"
When you step back and think about it, tcp4way is=20
just confusing us into thinking it is doing=20
something useful, but I don't believe it is.

>>2: it can expand SYN options space to 80 octets=20
>>without overriding DO field. It might be more middlebox friendly.
>
>I agree that if options are going to be placed=20
>beyond DO, and /if/ the signal that says they=20
>are there could be stripped or normalised by=20
>middleboxes, it is important that the signal=20
>that says they are there is ultra-safe - checked=20
>and double-checked for middlebox interference.
>
>However, deferring the point at which a=20
>connection moves to the established state=20
>doesn't suddenly mean that the segments sent=20
>before this didn't need any TCP options.
>
>Also:
>* Some IDSs and perhaps some stateful firewalls=20
>might not like the abnormal extra SYN/ACK.
>* The segments have to be abnormal (e.g. 4W flag=20
>or a TCP option) in order to signal 4WHS=20
>support. Your [Honda] study found option=20
>stripping was far from a minor effect, probably=20
>due to split connections - a split connection would also strip a Reserved=
 flag.
>
>
>Yep. You got a point and we know the discussions=20
>for middlebox is always tricky.
>I know this is not a very strong argument, but=20
>if we use payload as option space, we'll need=20
>extra mechanisms to avoid being parsed wrongly,=20
>which will increase complexity.=C2

EDO, tcp4way, EOS, Inner Space /all/ use payload=20
as option space, because if an option like EDO=20
that extends the header is stripped, the data=20
receiver thinks the data offset has reverted to=20
DO, so the place where the options had been placed becomes payload.

I explained to Joe in the posting pasted below,=20
why there is a safety issue when header and=20
payload do not necessarily share the same fate.

At 21:36 14/10/2014, Bob Briscoe wrote:
>>[Inner Space] does not involve any additional=20
>>TCP options in the TCP option space, ie the=20
>>space between the end of the 20B base TCP=20
>>header and the max 60B Data Offset. This makes=20
>>each [Inner Space] segment /inherently/ robust against option stripping.
>>
>>All the additional options of [Inner Space] are=20
>>beyond the Data Offset, including the option=20
>>that creates the additional space for the other=20
>>options. So they all share the same fate.
>>
>>Whereas, in EDO the option that creates the=20
>>space beyond DO is not itself beyond DO. So the=20
>>option that creates the space is at risk of=20
>>being stripped without stripping the space it=20
>>created. The two do not /inherently/ share the same fate.
>>
>>Yes, checks can be added to EDO to check for=20
>>option stripping (e.g. deferring use of extra=20
>>space until after the SYN/ACK). But this has to=20
>>assume that stripping behaviour remains=20
>>consistent through the flow. Whereas, with=20
>>[Inner Space], each individual segment is=20
>>robust against option stripping, all by itself.

This is why Joe has updated EDO to now require an=20
EDO option on every segment, even if no extra=20
option space is needed. Then if the receiver sees=20
the EDO option disappear at a certain point in=20
the connection, it knows that the payload could be corrupt.

This is still not as inherently safe as Inner=20
Space. For instance segmentation offload might be=20
programmed to forward unrecognised options but it=20
still resegments the data. So if it blindly=20
forwarded EDO with resegmented data, the app would receive corrupt payload.

>=C2
>>3: we might be able to re-design syn cookie=20
>>solution that can address the previous issues.
>What previous issues?
>
>
>In SYN cookie, tcp cannot use TCP options other=20
>than mss. Also, if the third ack is lost, the=20
>connection might be frozen because the server cannot retransmit SYN/ACK.

OK. This is why I have published the EchoCookie=20
option separately. I originally had it within the=20
Inner Space draft, but it should be usable by any of the approaches.

>=C2
>>4: even if an option in SYN/ACK has been=20
>>removed by middleboxes, it can reliably find this situation.
>
>So can any solution that is prepared to add as much latency as the 4WHS.
>
>
>Yep. maybe. but, not sure about complexity.

Lots of people can design simple solutions that=20
do not work. The task is surely to design the=20
simplest solution that does work. We need to=20
agree whether latency is part of the design brief.

>>5: in mptcp, initiating subflow requires 4=20
>>packet exchanges. With 4-way handshake=20
>>mechanism, we might be able to do this with non-mptcp-specific way.=C3=82
>
>There seems little point in doing a 4-way=20
>exchange in a non-mptcp-specific way if it still=20
>has the same latency problem. There is certainly=20
>little point if another solution is available=20
>that adds no latency and traverses middleboxes.
>
>
>Well, it may be a very tiny point, but, the third ACK can be pure ACK.=C2
>So, in order to perform 4-way exchange, =C2 mptcp=20
>may need to violate a basic rule in TCP, which is "don't send ack for ack"
>I personally don't want to see this kind of adhoc logic if possible.

This is not a tiny point. It is important to try=20
to avoid ad hoc exceptions. I am having that=20
discussion with the tcpcrypt authors, concerning a MAC on a pure ACK.

Because we /are/ changing basic stuff about TCP=20
here, we have to be prepared to change even=20
'basic' rules. However, if we do, we have to=20
replace them with equally universal new rules, not just ad hoc hacks.

>>6: similar to 5, it might be useful some secure=20
>>negotiations where we don't want to put=20
>>something that requires computations in SYN,=20
>>but want to perform some cryptic algorithms in later packet exchanges.
>
>Any specific TCP security option can always=20
>require that computation is deferred, without=20
>making a 4WHS the /only/ way to extend option space for all TCP options.
>
>
>Yep. again, you maybe right, but not sure about complexity.
>A problem in 3WHS is there's no good way to retransmit the third ACK.=C2
>We might be able to solve it with additional=20
>logic, but at the same time, I would like to see something simpler.
>
>I think I'm focusing on complexity rather than latency.=C2
>I'm not very sure if this is right or not, but I=20
>believe complexity is also an important factor.=C2
>
>However, I cannot and don't want to say 4WHS is=20
>less complex than others for now, it will be=20
>required more detailed analysis and comparisons.
>I just intuitively thought it looks like that. I=20
>might be wrong. I just would like to say we need more analysis.

Simplicity is very important, because otherwise bugs will appear.
However, as I said, the task is to design the=20
simplest solution that works, not a simple solution that doesn't work.

I am certain that a solution that adds latency=20
will not see any deployment - ie. it will not 'work'.

I also want /one/ solution that /completely/ gets=20
rid of the complexity of middlebox traversal for=20
every other TCP option. It is false simplicity if=20
the middlebox problem is not solved when=20
providing extra option space. Then every TCP=20
option still has all the traversal complexity.

Finally, I would caution against measuring the=20
complexity of a scheme by the number of pages of=20
the draft. For instance, the protocol spec in=20
inner-space-01 is only 12pp, because I have spent=20
a lot of effort simplifying the protocol design.=20
But the draft is 49pp, much of which is=20
justifying all the subtle aspects of what appears to be a simple design.

Most authors don't explain their rationale in a=20
draft. They don't realise that other people might=20
not infer their intent from the design, then they=20
have to write the rationale repeatedly in=20
response to questions on mailing lists.

I have posted a presentation about Inner Space that is a quick way to get=
 it:
<http://www.bobbriscoe.net/present.html#1411inner-space>


>Thanks,

This is useful conversation - thank you



Bob
>--
>Yoshi

________________________________________________________________
Bob Briscoe,                                                  BT=20
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<html>
<body>
Yoshifumi,<br><br>
At 09:09 04/11/2014, Yoshifumi Nishida wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">Hi Bob,<br><br>
On Mon, Nov 3, 2014 at 3:43 PM, Bob Briscoe
&lt;<a href=3D"mailto:bob.briscoe@bt.com">bob.briscoe@bt.com</a>&gt;
wrote:<br>

<dl>
<dd>Yoshifumi,<br><br>

<dd>I've just sent a critique of tcp4way to David &amp; the list. I've
pointed out problems with many of the things you say you like about it
inline below...<br><br>

<dd>At 04:34 22/10/2014, Yoshifumi Nishida wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">
<dd>Hi David,<br><br>

<dd>(as an individual of course..)<br>

<dd>I personally like this idea from the following reasons.<br><br>

<dd>1: use SYN-ACK to initiate feature negotiation (as discussed in the
previous thread)</blockquote><br>

<dd>tcp4way just seems to prevent use of TCP options on more segments
(and for more rounds).<br><br>

</dl><br>
Hmm. Sorry, I'm not very sure if I understand this.=C2 </blockquote><br>
See email to Dave Borman yesterday.<br>
&lt;<a=
 href=3D"http://www.ietf.org/mail-archive/web/tcpm/current/msg09322.html" eu=
dora=3D"autourl">
http://www.ietf.org/mail-archive/web/tcpm/current/msg09322.html</a>
&gt;<br>
Look for my first point: &quot;Following a false trail?&quot;<br>
When you step back and think about it, tcp4way is just confusing us into
thinking it is doing something useful, but I don't believe it
is.<br><br>
<blockquote type=3Dcite class=3Dcite cite=3D"">
<blockquote type=3Dcite class=3Dcite cite=3D"">
<dl>
<dd>2: it can expand SYN options space to 80 octets without overriding DO
field. It might be more middlebox friendly.</blockquote><br>

<dd>I agree that if options are going to be placed beyond DO, and /if/
the signal that says they are there could be stripped or normalised by
middleboxes, it is important that the signal that says they are there is
ultra-safe - checked and double-checked for middlebox interference.
<br><br>

<dd>However, deferring the point at which a connection moves to the
established state doesn't suddenly mean that the segments sent before
this didn't need any TCP options.<br><br>

<dd>Also:<br>

<dd>* Some IDSs and perhaps some stateful firewalls might not like the
abnormal extra SYN/ACK.<br>

<dd>* The segments have to be abnormal (e.g. 4W flag or a TCP option) in
order to signal 4WHS support. Your [Honda] study found option stripping
was far from a minor effect, probably due to split connections - a split
connection would also strip a Reserved flag.<br><br>

</dl><br>
Yep. You got a point and we know the discussions for middlebox is always
tricky.<br>
I know this is not a very strong argument, but if we use payload as
option space, we'll need extra mechanisms to avoid being parsed wrongly,
which will increase complexity.=C2 </blockquote><br>
EDO, tcp4way, EOS, Inner Space /all/ use payload as option space, because
if an option like EDO that extends the header is stripped, the data
receiver thinks the data offset has reverted to DO, so the place where
the options had been placed becomes payload. <br><br>
I explained to Joe in the posting pasted below, why there is a safety
issue when header and payload do not necessarily share the same fate.
<br><br>
At 21:36 14/10/2014, Bob Briscoe wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">
<blockquote type=3Dcite class=3Dcite cite=3D"">[Inner Space] does not involv=
e
any additional TCP options in the TCP option space, ie the space between
the end of the 20B base TCP header and the max 60B Data Offset. This
makes each [Inner Space] segment /inherently/ robust against option
stripping. <br><br>
All the additional options of [Inner Space] are beyond the Data Offset,
including the option that creates the additional space for the other
options. So they all share the same fate. <br><br>
Whereas, in EDO the option that creates the space beyond DO is not itself
beyond DO. So the option that creates the space is at risk of being
stripped without stripping the space it created. The two do not
/inherently/ share the same fate.<br>
<br>
Yes, checks can be added to EDO to check for option stripping (e.g.
deferring use of extra space until after the SYN/ACK). But this has to
assume that stripping behaviour remains consistent through the flow.
Whereas, with [Inner Space], each individual segment is robust against
option stripping, all by itself.</blockquote></blockquote><br>
This is why Joe has updated EDO to now require an EDO option on every
segment, even if no extra option space is needed. Then if the receiver
sees the EDO option disappear at a certain point in the connection, it
knows that the payload could be corrupt.<br><br>
This is still not as inherently safe as Inner Space. For instance
segmentation offload might be programmed to forward unrecognised options
but it still resegments the data. So if it blindly forwarded EDO with
resegmented data, the app would receive corrupt payload.<br><br>
<blockquote type=3Dcite class=3Dcite cite=3D"">=C2 <br>
<blockquote type=3Dcite class=3Dcite cite=3D"">
<dl>
<dd>3: we might be able to re-design syn cookie solution that can address
the previous issues.</blockquote>
<dd>What previous issues?<br><br>

</dl><br>
In SYN cookie, tcp cannot use TCP options other than mss. Also, if the
third ack is lost, the connection might be frozen because the server
cannot retransmit SYN/ACK.</blockquote><br>
OK. This is why I have published the EchoCookie option separately. I
originally had it within the Inner Space draft, but it should be usable
by any of the approaches.<br><br>
<blockquote type=3Dcite class=3Dcite cite=3D"">=C2 <br>
<blockquote type=3Dcite class=3Dcite cite=3D"">
<dl>
<dd>4: even if an option in SYN/ACK has been removed by middleboxes, it
can reliably find this situation.</blockquote><br>

<dd>So can any solution that is prepared to add as much latency as the
4WHS.<br><br>

</dl><br>
Yep. maybe. but, not sure about complexity.</blockquote><br>
Lots of people can design simple solutions that do not work. The task is
surely to design the simplest solution that does work. We need to agree
whether latency is part of the design brief.<br><br>
<blockquote type=3Dcite class=3Dcite cite=3D"">
<blockquote type=3Dcite class=3Dcite cite=3D"">
<dl>
<dd>5: in mptcp, initiating subflow requires 4 packet exchanges. With
4-way handshake mechanism, we might be able to do this with
non-mptcp-specific way.=C3=82 </blockquote><br>

<dd>There seems little point in doing a 4-way exchange in a
non-mptcp-specific way if it still has the same latency problem. There is
certainly little point if another solution is available that adds no
latency and traverses middleboxes.<br><br>

</dl><br>
Well, it may be a very tiny point, but, the third ACK can be pure ACK.=C2
<br>
So, in order to perform 4-way exchange, =C2 mptcp may need to violate a
basic rule in TCP, which is &quot;don't send ack for ack&quot;<br>
I personally don't want to see this kind of adhoc logic if
possible.</blockquote><br>
This is not a tiny point. It is important to try to avoid ad hoc
exceptions. I am having that discussion with the tcpcrypt authors,
concerning a MAC on a pure ACK. <br><br>
Because we /are/ changing basic stuff about TCP here, we have to be
prepared to change even 'basic' rules. However, if we do, we have to
replace them with equally universal new rules, not just ad hoc hacks.
<br><br>
<blockquote type=3Dcite class=3Dcite cite=3D"">
<blockquote type=3Dcite class=3Dcite cite=3D"">
<dl>
<dd>6: similar to 5, it might be useful some secure negotiations where we
don't want to put something that requires computations in SYN, but want
to perform some cryptic algorithms in later packet
exchanges.</blockquote><br>

<dd>Any specific TCP security option can always require that computation
is deferred, without making a 4WHS the /only/ way to extend option space
for all TCP options.<br><br>

</dl><br>
Yep. again, you maybe right, but not sure about complexity.<br>
A problem in 3WHS is there's no good way to retransmit the third ACK.=C2
<br>
We might be able to solve it with additional logic, but at the same time,
I would like to see something simpler.<br><br>
I think I'm focusing on complexity rather than latency.=C2 <br>
I'm not very sure if this is right or not, but I believe complexity is
also an important factor.=C2 <br><br>
However, I cannot and don't want to say 4WHS is less complex than others
for now, it will be required more detailed analysis and comparisons.<br>
I just intuitively thought it looks like that. I might be wrong. I just
would like to say we need more analysis.</blockquote><br>
Simplicity is very important, because otherwise bugs will appear.<br>
However, as I said, the task is to design the simplest solution that
works, not a simple solution that doesn't work.<br><br>
I am certain that a solution that adds latency will not see any
deployment - ie. it will not 'work'.<br><br>
I also want /one/ solution that /completely/ gets rid of the complexity
of middlebox traversal for every other TCP option. It is false simplicity
if the middlebox problem is not solved when providing extra option space.
Then every TCP option still has all the traversal complexity.<br><br>
Finally, I would caution against measuring the complexity of a scheme by
the number of pages of the draft. For instance, the protocol spec in
inner-space-01 is only 12pp, because I have spent a lot of effort
simplifying the protocol design. But the draft is 49pp, much of which is
justifying all the subtle aspects of what appears to be a simple design.
<br><br>
Most authors don't explain their rationale in a draft. They don't realise
that other people might not infer their intent from the design, then they
have to write the rationale repeatedly in response to questions on
mailing lists.<br><br>
I have posted a presentation about Inner Space that is a quick way to get
it:<br>
&lt;<a href=3D"http://www.bobbriscoe.net/present.html#1411inner-space" eudor=
a=3D"autourl">
http://www.bobbriscoe.net/present.html#1411inner-space</a>&gt;<br><br>
<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">Thanks,</blockquote><br>
This is useful conversation - thank you<br><br>
<br><br>
Bob<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">--<br>
Yoshi</blockquote>
<x-sigsep><p></x-sigsep>
________________________________________________________________<br>
Bob
Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
BT</body>
</html>

--=====================_941999697==.ALT--


From nobody Tue Nov  4 09:47:06 2014
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On 11/4/2014 8:52 AM, Bob Briscoe wrote:
...
> EDO, tcp4way, EOS, Inner Space /all/ use payload as option space,
> because if an option like EDO that extends the header is stripped, the
> data receiver thinks the data offset has reverted to DO, so the place
> where the options had been placed becomes payload.
> 
> I explained to Joe in the posting pasted below, why there is a safety
> issue when header and payload do not necessarily share the same fate.

That's also true for DO, FWIW.

However, the issue of how the option is encoded is orthogonal to these
solutions. Bob's in-band encoding would work in any of them, but also
ends up "encoding TCP inside TCP", which can easily lead to escalation
(as I've pointed out on the list already).

For that reason, I don't support the idea of in-band encoding, but
that's independent of the way in which the space itself is indicated and
coordinated.

...
> This is why Joe has updated EDO to now require an EDO option on every
> segment, even if no extra option space is needed. Then if the receiver
> sees the EDO option disappear at a certain point in the connection, it
> knows that the payload could be corrupt.
> 
> This is still not as inherently safe as Inner Space.

And neither would be as safe as running encrypted TCP inside TCP.

The question is whether we need to deal with this issue, i.e., whether
such middlebox asymmetries can occur in practice.

> ...We need to agree
> whether latency is part of the design brief.

+1

Joe


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> On Nov 3, 2014, at 5:21 PM, Bob Briscoe <bob.briscoe@bt.com> wrote:
>=20
> David,=20
>=20
> Thx for updating the draft & filename, and particularly for the adding =
the roster of existing TCP options. However, to be honest, I think =
tcp4way has got caught up in following a false trail. Let me explain, =
and I'll add other critique afterwards too.
>=20
> 1) Following a false trail?
>=20
> Tcp4way aims to give the client a second chance to negotiate some more =
TCP options on a subsequent SYN/ACK. The original reason you wanted this =
(from emails back in the summer) was because it would be really complex =
to allow a second segment from the client to negotiate more options if =
the hosts had already made an interim decision after the first round. =
They might need to revoke some of their earlier decisions, which would =
be extremely complex and often impossible.

No.  My original point is that negotiating TCP options in packets =
without the SYN bit requires additional complexity because ACK-only =
packets are not reliably transmitted.  The 4WHS adds a client SYN/ACK =
packet that is reliably transmitted, and is distinct from the original =
client SYN packet.

For a non-directional option, it doesn=92t matter which side sends it =
first, hence the server SYN/ACK can propose using additional =
non-directional options, which can then be confirmed in the client =
SYN/ACK of the 4WHS.

Because both sides know that the other side supports extended option =
space, both of the SYN/ACK packets can take advantage of it.

> However, all that tcp4way has achieved is to send more segments before =
making a decision. That means more segments on which TCP options cannot =
be /used/ yet.

Again, no.  Options in the initial SYN and SYN/ACK are not delayed.  =
Nothing from the client side is delayed, because it decides the full set =
of what is being used when it gets the servers SYN/ACK, because it knows =
which options from the SYN were accepted by the server, and which =
options proposed by the server it is willing to acknowledge when it =
sends the SYN/ACK.  It=92s the same point as with the 3WHS.

The server won=92t know if the client supports the additional options =
that the server proposed in the SYN/ACK until it gets the client=92s =
SYN/ACK, so from the server standpoint, there=92s one more RTT for those =
additional options.


>=20
> I've calculated how many segments in both directions before a 4WHS =
server enters the established state:
> * Without TFO: 3 segments, including 2 that could carry data
> * A resumed connection with TFO & IW=3D3: 7 segments, all potentially =
with data.
> * A resumed connection with TFO & IW=3D10: 21 segments, all =
potentially with data.
>=20
> So all tcp4way achieves is to prevent TCP options being included on =
more segments.

I don=92t follow this logic at all.  With the 4WHS the server enters =
established state at the exact same time as it does with the 3WHS, when =
its SYN is acknowledged.  Whether that is an ACK-only packet with a =
3WHS, or a SYN/ACK with a 4WHS, from the server side it still happens at =
the same point in the exchange.  =46rom the servers perspective, the =
only difference is that for the 4WHS it sends a final ACK and does final =
option enablement, and for the 3WHS it doesn=92t.=20


>=20
> I believe it was wrong to conclude (back in the summer) that the =
problem is all about holding back a move to the established state. Most =
existing TCP options apply to segments of any kind, irrespective of =
whether the host is in the established state when it sends them, and =
irrespective of whether they even carry data.

I never said the problem was about holding back established state.

I said that if the servers SYN/ACK contained additional options, with =
the 4WHS the client can respond with a SYN/ACK to complete negotiation =
of those additional options, whereas the 3WHS doesn=92t have that =
possibility.  And the only reason those options wouldn=92t have been in =
the SYN is because there wasn=92t enough space, but with EDO there is =
now space in the SYN/ACK.

>=20
> In all the following examples, if the client wants to support option X =
and the server can support X, the server would ideally like to start =
using X on the first SYN/ACK, whether or not the client is offering to =
delay moving into the established state for a round:

If X was in the SYN, it can.  If X wasn=92t in the SYN, it can=92t, =
because it doesn=92t know that the client supports X, it=92ll have to =
wait for the SYN/ACK to know if the client supports X.

> * TCP-AO
TCP-AO is not negotiated.

> * The following sub-options of the tcpcrypt CRYPT option: DECLINE, =
PKCONF*, NEXTK1*, NEXTK2*, INIT1, INIT2, UNKNOWN, SYNCOOOKIE, ACKCOOKIE, =
REKEY, REKEYSTREAM.
> * the tcpcrypt MAC option
> * MPTCP_JOIN, ADD_ADDR(2), DSS, MP_PRIO, MP_FAIL
> * SACK permitted
> * MSS
> * TFO=20
> * TS (if used for RTT calculation)
> * UTO
>=20
> The only ones that are meaningless on the first SYN/ACK are:
> * SACK (because the SACK block for zero previous ACKs would be pretty =
uninteresting. However SACK would be useful on ACKs from the client =
while still in SYN-RCVD)=20
> * TS (if used for PAWS, 'cos there's no risk of wrap on the first few =
segments)
> * WS (The client can't use a large RWND until it has opened up cwnd =
thru a few rounds of slow-start. However, this is arguable, because =
there might be occasions where a connection can safely inherit a large =
cwnd).
>=20
> Some option are only meaningful once data has started (e.g. UTO, =
MPTCP-DSS, WS), but even those can be used prior to connection =
establishment.

>=20
> We can't know which of these two lists yet-to-be-defined new options =
will be in. If the past is a good predictor of the future, the first =
list is more likely (ie. the ones tcp4way doesn't help much).

I=92m not quite sure what your point is with just an enumerated list.  =
For each of them, the question is, if the option is not included in the =
initial client SYN, can it make sense in the servers SYN/ACK?  For =
example, if a server doesn=92t receive TFO in the SYN, it could still =
send TFO in the SYN/ACK, which would just be ignored by non-TFO =
implementations.  Directional options, where the servers option is =
dependent on first knowing the contents of the clients option, are not =
candidates.

The 4WHS does not give the client a second chance to initiate option =
negotiation, that would require a 5WHS.

>=20
> 2) Latency
>=20
> Tcp4way always introduces latency. How much depends on which scenario:
> * TCP client sends request first (by far the majority of connections): =
1 RTT
> * TCP server sends data first (e.g. resuming a push connection): 0.5 =
RTT
>=20
> As JohnL said, "any added latency is nearly fatal". I strongly agree =
with every word, except "nearly=94.

If you are talking about latency of delivering user data, there isn=92t =
any.  See my message entitled "We already support multiple versions of =
the TCP header=94, and look for the section entitled =93Delivering User =
Data=94.  Those changes will be in the next revision of the 4WHS draft.


> 3) Firewalls & Intrusion Detection Systems
...

I don=92t subscribe to the notion that today=92s deployed middle boxes =
get to dictate all future development of the TCP protocol.  Yes, =
ignoring them would be silly, but kowtowing to them is equally silly.

We need to first decide on the destination, and then find a path to that =
destination.  If we just look at the paths through the middle boxes, =
we=92ll still get somewhere, but will it be where we want to go?

> __________________
> They are my 3 main criticisms. The rest are nits...
>=20
> 4) Nit: Reflection of Reserved Flags
...

Did you miss sections 4.3.2 and 4.3.3 where I discussed why this is not =
an issue?

> 5) Nit: Remembering EDO support across connections
>=20
> S.7.3.2.1 says that server EDO support can be saved alongside a TFO =
cookie. I don't believe this is safe. During an upgrade, there might be =
some server replicas that support EDO and others that don't. The legacy =
servers would then ignore the EDO option, but still pass TCP options as =
a corrupt payload to the app.
>=20
> [BTW, it is safe for a client to hold solely the TFO capability of a =
server in a cookie, because (unlike EDO) TFO doesn't put data that is =
not intended for the app in the payload. So, if a replica server that =
picks up the resumed connection does not support TFO, it will simply =
ignore the cookie and just hold the data back until the 3WHS completes.]

Good points, I=92ll take that out.

>=20
> 6) Nit: Unknown Option
>=20
> Altho it would be nice to think that this might help, there doesn't =
seem to be a case where it will. Can you give an example where the =
Unknown Options is useful?

I did, UTO.  It would also be useful for late enablement of TS.  =
Basically, it is useful for using any option which does not require that =
use of that option be negotiated in the initial SYN exchange.

The main purpose of the Unknown Option option is to provide a generic =
way to inform the other side that you don=92t understand a particular =
option, when that option first appears mid-stream, rather than support =
for that option being negotiated in the initial SYN exchange.  =
Otherwise, if you send an option for the first time mid-stream, the only =
way to know that the other side doesn=92t support it is by lack of any =
response to that option.  That requires tracking what sequence number =
you first sent it with, and waiting for the ACK for that sequence number =
to decide on a non-response.  But sequence number tracking doesn=92t =
work if you don=92t have any new data to send, so you need another =
mechanism.  If you know the other side support the Unknown Option =
option, then you know it=92ll tell you if you send any other options =
that it doesn=92t understand.=20

> 7) Nit: T/TCP is obsolete.
>=20
> T/TCP had vulnerabilities and was obsoleted by RFC6247.

Yes, that=92s why the document says:

      Though now obsolete [RFC6247], this is a good
      example of a directional TCP option.

> Bob

		-David=


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From: David Borman <dab@weston.borman.com>
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> On Nov 4, 2014, at 3:01 PM, David Borman <dab@weston.borman.com> =
wrote:
...
> I don=92t subscribe to the notion that today=92s deployed middle boxes =
get to dictate all future development of the TCP protocol.  Yes, =
ignoring them would be silly, but kowtowing to them is equally silly.
>=20
> We need to first decide on the destination, and then find a path to =
that destination.  If we just look at the paths through the middle =
boxes, we=92ll still get somewhere, but will it be where we want to go?
...
I re-read that, and I could be clearer.

We need to first decide what we want TCP to look like, and then find a =
way to transition from current implementations to the new design, which =
might include middle boxes having to change, rather than letting current =
middle box implementations dictate how we can or can=92t change TCP.

	-David


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+1


> On Nov 4, 2014, at 5:15 PM, David Borman <dab@weston.borman.com> wrote:
>=20
>=20
>>> On Nov 4, 2014, at 3:01 PM, David Borman <dab@weston.borman.com> wrote:
>> ...
>> I don=E2=80=99t subscribe to the notion that today=E2=80=99s deployed mid=
dle boxes get to dictate all future development of the TCP protocol.  Yes, i=
gnoring them would be silly, but kowtowing to them is equally silly.
>>=20
>> We need to first decide on the destination, and then find a path to that d=
estination.  If we just look at the paths through the middle boxes, we=E2=80=
=99ll still get somewhere, but will it be where we want to go?
> ...
> I re-read that, and I could be clearer.
>=20
> We need to first decide what we want TCP to look like, and then find a way=
 to transition from current implementations to the new design, which might i=
nclude middle boxes having to change, rather than letting current middle box=
 implementations dictate how we can or can=E2=80=99t change TCP.
>=20
>    -David
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Wed Nov  5 05:42:16 2014
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Hi,

We have updated the TCPM agenda a bit. The current agenda is at =
http://www.ietf.org/proceedings/91/agenda/agenda-91-tcpm

Speakers: please send your slides by Sunday 15:00 Hawaii time.

- Pasi


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Catching up...

On 10/27/2014 9:42 PM, Wesley Eddy wrote:
...
> For what it's worth, I also think the OOB approach is more desirable
> than the "dual-stack" type, however, am more uncertain about firewall
> behavior towards the segments with both SYN & ACK bits clear. 

Let's be clear - we're ALL in the dark about any of these, both as to
whether they will end up working or to what extent they would be
over-engineered to get around phantoms that don't exist.

> That's
> definitely a good experiment to get widespread data on.  I also don't
> really understand specifically how one decides how long to wait for
> the OOB segment before assuming it isn't coming,

You treat the OOB and SYN as a pair, and treat it with the same time-out
on both ends as far as giving up or retransmitting. How long you keep
queued copies to deal with reordering or other arrival-timing issues is
up to the receiver.

> but that could be
> tweakable or algorithmically determined, so I don't think it's a major
> issue.

Yes, esp. for the caching part.

Joe


From nobody Wed Nov  5 23:24:59 2014
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hi Joe, all,

I'm *almost* caught up on this thread, but I did want jump in to =
underline something that Joe said here:
=20
> On 06 Nov 2014, at 01:55, Joe Touch <touch@isi.edu> wrote:
>=20
> Catching up...
>=20
> On 10/27/2014 9:42 PM, Wesley Eddy wrote:
> ...
>> For what it's worth, I also think the OOB approach is more desirable
>> than the "dual-stack" type, however, am more uncertain about firewall
>> behavior towards the segments with both SYN & ACK bits clear.=20
>=20
> Let's be clear - we're ALL in the dark about any of these, both as to
> whether they will end up working or to what extent they would be
> over-engineered to get around phantoms that don't exist.

+many.

One of the reasons it's been difficult to follow this whole discussion =
(beyond "I did not read email during the month of October") is that the =
model of the middle/meddle/muddleboxes we're trying to defend against is =
often implicit in the descriptions of the proposed mechanisms.

Further, it's not clear to me the extent to which these implicit models =
correspond to real-world devices, or to the extent that they do how =
often they are deployed in such a way that they would lead to =
reachability failure in the case of certain proposed extensions, or =
header/payload invariance failure that would lead to failure modes in =
those extensions. A middlebox that breaks a proposed extension mechanism =
in impossible to debug ways but that appears on 1e-7 of the paths on the =
Internet and is involved in 1e-14 flows falls into the category of =
"acceptable risk".

Further, even if we were arguing from explicit models backed up by =
openly available data, it would be unclear how the data from study A =
applies to situation B. Published network measurements relevant to these =
questions is subject to fairly large systematic biases, or put another =
way, we know much more about academic networks than we do about mobile =
access networks.

I see two ways forward here, which are probably complementary.

(1) Measure path impairment on the Internet

We pull together all the data from various studies of broken paths on =
the Internet, build from it a set of observations (i.e. at time t the =
path from x to y had these characteristics), look for correlations among =
those characteristics to allow the amplification of observations when =
possible, then correlate this "path impairment data" with estimates of =
traffic and topology information to allow us to quantify the risk of =
extension failure or extension-linked connectivity failure, and =
prioritize design effort based on this estimated risk.=20

This is not a new idea. There is already a fair amount of work in this =
space, and probably a bit of infrastructure we could reuse, but to be =
useful for protocol design I think it would need to be placed into a =
single database, or at least various path impairment databases would =
need to share a common schema.

I recognize that this proposal has all the same problems with =
measurement data access that all other open measurement platforms do, =
with the added bonus that active measurements in this space will be =
interpreted as malicious by many of the same people who configure their =
firewalls to do end-to-end-hostile things. And given the people reading =
this message, even if we pulled all available data and designed lots of =
active measurements to plug the gaps, and all of you dropped everything =
you were doing to work on this, and told all your friends and colleagues =
to do the same, and they did so, we would still not have a very =
convincing answer to the bias question.

(2) Build path impairment measurement into the specifications and =
implementations of the extensions.

Or "fallback on steroids". This gives a tautological answer to the bias =
question, and if we're proposing vastly overengineering the extension =
mechanisms anyway we might as well *explicitly* have them measure =
in-line as part of the protocol interaction and/or simultaneously with =
protocol interactions in independent flows.=20

Assuming one can devise a representation for path impairment that is =
useful and does not require the exposure of routable IP addresses or =
derived anonymized data from which routable IP addresses can be =
inferred, (2) can even be used as a source of data for (1) given the =
growing number of operating environments that already have a concept for =
"user consent to passive monitoring for debugging and development =
purposes" subject to the same magnitude of bias that (1) is already =
subject to.

Cheers,

Brian

>> That's
>> definitely a good experiment to get widespread data on.  I also don't
>> really understand specifically how one decides how long to wait for
>> the OOB segment before assuming it isn't coming,
>=20
> You treat the OOB and SYN as a pair, and treat it with the same =
time-out
> on both ends as far as giving up or retransmitting. How long you keep
> queued copies to deal with reordering or other arrival-timing issues =
is
> up to the receiver.
>=20
>> but that could be
>> tweakable or algorithmically determined, so I don't think it's a =
major
>> issue.
>=20
> Yes, esp. for the caching part.
>=20
> Joe
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri Nov  7 11:26:15 2014
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Joe,

At 00:55 06/11/2014, Joe Touch wrote:
>Catching up...
>
>On 10/27/2014 9:42 PM, Wesley Eddy wrote:
>...
> > For what it's worth, I also think the OOB approach is more desirable
> > than the "dual-stack" type, however, am more uncertain about firewall
> > behavior towards the segments with both SYN & ACK bits clear.
>
>Let's be clear - we're ALL in the dark about any of these, both as to
>whether they will end up working or to what extent they would be
>over-engineered to get around phantoms that don't exist.
>
> > That's
> > definitely a good experiment to get widespread data on.  I also don't
> > really understand specifically how one decides how long to wait for
> > the OOB segment before assuming it isn't coming,
>
>You treat the OOB and SYN as a pair, and treat it with the same time-out
>on both ends as far as giving up or retransmitting. How long you keep
>queued copies to deal with reordering or other arrival-timing issues is
>up to the receiver.

Not so IMO. The OOB has to be sent a 'guard time' after the SYN, to 
minimise the chance of it getting reordered and then getting tangled 
up in a firewall before the SYN arrives; the firewall will chuck any 
segment for which it has not seen a prior SYN.

So altho the timeouts might be the same it doesn't seem sensible to 
start the timers at the same time. Two possibilities:
a) the guard time is standardised.
b) the guard time is up to the client

If a), we (the IETF) don't know the operating conditions, so it's 
difficult to agree on a standard value. We can't calibrate it in 
units of RTT, because neither end knows the RTT at the stage when 
they are sending the SYN.

If b), the server end doesn't know what the guard time is, so its 
choice of when to start the timer for the OOB isn't straightforward.

I guess the data that is being collected by ALex Z & co /might/ help 
choose a guard time that avoids the worst re-ordering effects, 
whether it's chosen at standardisation time (by the IETF) or at 
run-time (by the client).



Bob


> > but that could be
> > tweakable or algorithmically determined, so I don't think it's a major
> > issue.
>
>Yes, esp. for the caching part.
>
>Joe

________________________________________________________________
Bob Briscoe,                                                  BT 


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Hi, Bob,

On 11/7/2014 11:26 AM, Bob Briscoe wrote:
> Joe,
> 
> At 00:55 06/11/2014, Joe Touch wrote:
...
>> You treat the OOB and SYN as a pair, and treat it with the same time-out
>> on both ends as far as giving up or retransmitting. How long you keep
>> queued copies to deal with reordering or other arrival-timing issues is
>> up to the receiver.
> 
> Not so IMO. The OOB has to be sent a 'guard time' after the SYN, to
> minimise the chance of it getting reordered and then getting tangled up
> in a firewall before the SYN arrives; the firewall will chuck any
> segment for which it has not seen a prior SYN.

Firewalls aren't the issue. NATs are, and unless reordering is prevalent
that won't be an issue.

Reordering would mostly happen on multiple paths, but you'd need
multiple paths between the client and the NAT - i.e., in the private
side. That's rare if it happens at all.

I thus don't agree that guard times are needed, which means we don't
need to worry about determining them.

> So although the timeouts might be the same it doesn't seem sensible to
> start the timers at the same time. 

I currently believe that there should be one timer. If you need to
resend, you resend BOTH the SYN and OOB. The server side can cache to
deal with reordering that happens there, but that doesn't require
guard-time at client transmission.

Joe


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David,

At 21:01 04/11/2014, David Borman wrote:

> > On Nov 3, 2014, at 5:21 PM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> >
> > David,
> >
> > Thx for updating the draft & filename, and particularly for the 
> adding the roster of existing TCP options. However, to be honest, I 
> think tcp4way has got caught up in following a false trail. Let me 
> explain, and I'll add other critique afterwards too.
> >
> > 1) Following a false trail?
> >
> > Tcp4way aims to give the client a second chance to negotiate some 
> more TCP options on a subsequent SYN/ACK. The original reason you 
> wanted this (from emails back in the summer) was because it would 
> be really complex to allow a second segment from the client to 
> negotiate more options if the hosts had already made an interim 
> decision after the first round. They might need to revoke some of 
> their earlier decisions, which would be extremely complex and often impossible.
>
>No.  My original point is that negotiating TCP options in packets 
>without the SYN bit requires additional complexity because ACK-only 
>packets are not reliably transmitted.  The 4WHS adds a client 
>SYN/ACK packet that is reliably transmitted, and is distinct from 
>the original client SYN packet.

OK, understood (altho RFC793 does not require the ACK of the 3rd leg 
to be pure. It can ACK the SYN/ACK and send data - I can dig out some 
recent literature searches I did on why there is usually a 3rd ACK, 
if anyone's interested). But, granted, in the general case your point 
is valid that there might not be any data at this point.


>For a non-directional option, it doesn't matter which side sends it 
>first, hence the server SYN/ACK can propose using additional 
>non-directional options, which can then be confirmed in the client 
>SYN/ACK of the 4WHS.
>
>Because both sides know that the other side supports extended option 
>space, both of the SYN/ACK packets can take advantage of it.
>
> > However, all that tcp4way has achieved is to send more segments 
> before making a decision. That means more segments on which TCP 
> options cannot be /used/ yet.
>
>Again, no.  Options in the initial SYN and SYN/ACK are not 
>delayed.  Nothing from the client side is delayed, because it 
>decides the full set of what is being used when it gets the servers 
>SYN/ACK, because it knows which options from the SYN were accepted 
>by the server, and which options proposed by the server it is 
>willing to acknowledge when it sends the SYN/ACK.  It's the same 
>point as with the 3WHS.
>
>The server won't know if the client supports the additional options 
>that the server proposed in the SYN/ACK until it gets the client's 
>SYN/ACK, so from the server standpoint, there's one more RTT for 
>those additional options.

Only those options that fit on the first SYN can start being used 
early. The rest still have to wait. Given the point of this exercise 
is to make more space for options, you are saying that you can only 
provide more space for those options that are prepared to wait.


> >
> > I've calculated how many segments in both directions before a 
> 4WHS server enters the established state:
> > * Without TFO: 3 segments, including 2 that could carry data
> > * A resumed connection with TFO & IW=3: 7 segments, all 
> potentially with data.
> > * A resumed connection with TFO & IW=10: 21 segments, all 
> potentially with data.
> >
> > So all tcp4way achieves is to prevent TCP options being included 
> on more segments.
>
>I don't follow this logic at all.

That's possibly my point. I don't think your logic is relevant to the 
problem at hand.

>With the 4WHS the server enters established state at the exact same 
>time as it does with the 3WHS, when its SYN is 
>acknowledged.  Whether that is an ACK-only packet with a 3WHS, or a 
>SYN/ACK with a 4WHS, from the server side it still happens at the 
>same point in the exchange.  From the servers perspective, the only 
>difference is that for the 4WHS it sends a final ACK and does final 
>option enablement, and for the 3WHS it doesn't.

My point is that most TCP options are useful on /segments/. The 
problem at hand has nothing to do with the state that the TCP 
state-machine has reached. It has to do with:
a) minimising the round trip times before all TCP options can start to be used;
b) minimising how many segments might be exchanged before all TCP 
options can start to be used.


> >
> > I believe it was wrong to conclude (back in the summer) that the 
> problem is all about holding back a move to the established state. 
> Most existing TCP options apply to segments of any kind, 
> irrespective of whether the host is in the established state when 
> it sends them, and irrespective of whether they even carry data.
>
>I never said the problem was about holding back established state.
>
>I said that if the servers SYN/ACK contained additional options, 
>with the 4WHS the client can respond with a SYN/ACK to complete 
>negotiation of those additional options, whereas the 3WHS doesn't 
>have that possibility.  And the only reason those options wouldn't 
>have been in the SYN is because there wasn't enough space, but with 
>EDO there is now space in the SYN/ACK.

OK I accept that your goal was not about holding back established state.

But I'm not impressed. You just allowed up to 21 segments to go past 
without being able to use some TCP options that you are still in the 
process of negotiating.


> >
> > In all the following examples, if the client wants to support 
> option X and the server can support X, the server would ideally 
> like to start using X on the first SYN/ACK, whether or not the 
> client is offering to delay moving into the established state for a round:
>
>If X was in the SYN, it can.  If X wasn't in the SYN, it can't, 
>because it doesn't know that the client supports X, it'll have to 
>wait for the SYN/ACK to know if the client supports X.

Yes. And that's why I'm not particularly impressed.


> > * TCP-AO
>TCP-AO is not negotiated.
>
> > * The following sub-options of the tcpcrypt CRYPT option: 
> DECLINE, PKCONF*, NEXTK1*, NEXTK2*, INIT1, INIT2, UNKNOWN, 
> SYNCOOOKIE, ACKCOOKIE, REKEY, REKEYSTREAM.
> > * the tcpcrypt MAC option
> > * MPTCP_JOIN, ADD_ADDR(2), DSS, MP_PRIO, MP_FAIL
> > * SACK permitted
> > * MSS
> > * TFO
> > * TS (if used for RTT calculation)
> > * UTO
> >
> > The only ones that are meaningless on the first SYN/ACK are:
> > * SACK (because the SACK block for zero previous ACKs would be 
> pretty uninteresting. However SACK would be useful on ACKs from the 
> client while still in SYN-RCVD)
> > * TS (if used for PAWS, 'cos there's no risk of wrap on the first 
> few segments)
> > * WS (The client can't use a large RWND until it has opened up 
> cwnd thru a few rounds of slow-start. However, this is arguable, 
> because there might be occasions where a connection can safely 
> inherit a large cwnd).
> >
> > Some option are only meaningful once data has started (e.g. UTO, 
> MPTCP-DSS, WS), but even those can be used prior to connection establishment.
>
> >
> > We can't know which of these two lists yet-to-be-defined new 
> options will be in. If the past is a good predictor of the future, 
> the first list is more likely (ie. the ones tcp4way doesn't help much).
>
>I'm not quite sure what your point is with just an enumerated list.

(Other than mistakenly including TCP-AO) I've divided up the options 
into those that make sense as early as the first SYN/ACK and those 
that could probably wait. So,...

* A 4WHS client has to consider all the options it thinks might be 
needed and decide which can wait. I'm not convinced that it makes 
sense to wait for any of the options in the first list. For some of 
them, it certainly gets messy to start without them, then to add them.
* All the particularly large options are in the first list.
* The very large majority of connections only consists of a few 
segments. 4WHS provides extra space after most connections will have 
finished. So, in the large majority of connections, 4WHS doesn't 
solve any problem at all.

>For each of them, the question is, if the option is not included in 
>the initial client SYN, can it make sense in the servers 
>SYN/ACK?  For example, if a server doesn't receive TFO in the SYN, 
>it could still send TFO in the SYN/ACK, which would just be ignored 
>by non-TFO implementations.  Directional options, where the servers 
>option is dependent on first knowing the contents of the clients 
>option, are not candidates.

The example you have chosen illustrates my concern in a number of ways.

TFO isn't a problem on the first connection in which it is used. It 
consumes 2B on the SYN, and 6-18B on the SYN/ACK.

TFO hits the space problem when it is used on a subsequent 
connection. The client sends the cookie it got from the previous 
connection /in the SYN/. There is no TFO option in the SYN/ACK on the 
resumed connection. The whole point of TFO is for the client to prove 
to the server that it had a previous valid connection.

Secondly, there is no point deferring TFO to the second SYN/ACK, 
because the 'F' in TFO means 'Fast'.


Similarly for tcpcrypt. There's no point having sent up to 21 
segments of data in the clear before any keys have been agreed. OK, 
you might argue that if you're using tcpcrypt you send no data until 
the keys have been agreed. However, the point of tcpcrypt is meant to 
be opportunistic encryption. I.e. it can be turned on for /all/ 
connections with zero config and minimal performance hit. If turning 
on tcpcrypt causes every connection to have to wait an extra round 
before it can send data, it will not be ubiquitous. It will be nobiquitous.


>The 4WHS does not give the client a second chance to initiate option 
>negotiation, that would require a 5WHS.
>
> >
> > 2) Latency
> >
> > Tcp4way always introduces latency. How much depends on which scenario:
> > * TCP client sends request first (by far the majority of 
> connections): 1 RTT
> > * TCP server sends data first (e.g. resuming a push connection): 0.5 RTT
> >
> > As JohnL said, "any added latency is nearly fatal". I strongly 
> agree with every word, except "nearly".
>
>If you are talking about latency of delivering user data, there 
>isn't any.  See my message entitled "We already support multiple 
>versions of the TCP header", and look for the section entitled 
>"Delivering User Data".  Those changes will be in the next revision 
>of the 4WHS draft.

There is latency added to any user-data (or empty segments) that want 
the benefit of the TCP options we're trying to make space for.

In 4WHS, it's the extra space that suffers the latency. It doesn't 
appear soon enough to be useful.


> > 3) Firewalls & Intrusion Detection Systems
>...
>
>I don't subscribe to the notion that today's deployed middle boxes 
>get to dictate all future development of the TCP protocol.  Yes, 
>ignoring them would be silly, but kowtowing to them is equally silly.

I'll substitute your corrected text for your next para...

>We need to first decide what we want TCP to look like, and then find 
>a way to transition from current implementations to the new design, 
>which might include middle boxes having to change, rather than 
>letting current middle box implementations dictate how we can or 
>can't change TCP.

With inner space, I also started from where we should want TCP to end 
up, and found a way to transition to it. Here's my starting principles:
1) Options should be in the TCP header. Extensions should be within 
the TCP Data.
    [I just submitted a paper to the IAB SEMI workshop that gives the 
rationale for that:
    <http://bobbriscoe.net/projects/2020comms/tcp/briscoe-inspace_semi.pdf>.
    ]
2) Minimal delay before options can be used
3) Choice between fire-and-forget options and reliable-in-order options
4) Host control over which options and base header fields are 
available for interaction with middleboxes, and which are strictly e2e.
5) Enforcement of the above choice through authentication and/or encryption.
6) Incremental deployment (against legacy servers and legacy middleboxes)

You say "transition ... might include middleboxes having to change". 
That earns you a patronising smile. Nonetheless, starting from 
principles is a sentiment I can wholeheartedly agree with.


> > __________________
> > They are my 3 main criticisms. The rest are nits...
> >
> > 4) Nit: Reflection of Reserved Flags
>...
>
>Did you miss sections 4.3.2 and 4.3.3 where I discussed why this is 
>not an issue?

I read these sections and they have misunderstood the problem.

There is an easy way to solve the problem tho. (which I suggested in 
the text that you've removed from the thread).


> > 5) Nit: Remembering EDO support across connections
> >
> > S.7.3.2.1 says that server EDO support can be saved alongside a 
> TFO cookie. I don't believe this is safe. During an upgrade, there 
> might be some server replicas that support EDO and others that 
> don't. The legacy servers would then ignore the EDO option, but 
> still pass TCP options as a corrupt payload to the app.
> >
> > [BTW, it is safe for a client to hold solely the TFO capability 
> of a server in a cookie, because (unlike EDO) TFO doesn't put data 
> that is not intended for the app in the payload. So, if a replica 
> server that picks up the resumed connection does not support TFO, 
> it will simply ignore the cookie and just hold the data back until 
> the 3WHS completes.]
>
>Good points, I'll take that out.
>
> >
> > 6) Nit: Unknown Option
> >
> > Altho it would be nice to think that this might help, there 
> doesn't seem to be a case where it will. Can you give an example 
> where the Unknown Options is useful?
>
>I did, UTO.  It would also be useful for late enablement of 
>TS.  Basically, it is useful for using any option which does not 
>require that use of that option be negotiated in the initial SYN exchange.
>
>The main purpose of the Unknown Option option is to provide a 
>generic way to inform the other side that you don't understand a 
>particular option, when that option first appears mid-stream, rather 
>than support for that option being negotiated in the initial SYN 
>exchange.  Otherwise, if you send an option for the first time 
>mid-stream, the only way to know that the other side doesn't support 
>it is by lack of any response to that option.  That requires 
>tracking what sequence number you first sent it with, and waiting 
>for the ACK for that sequence number to decide on a 
>non-response.  But sequence number tracking doesn't work if you 
>don't have any new data to send, so you need another mechanism.  If 
>you know the other side support the Unknown Option option, then you 
>know it'll tell you if you send any other options that it doesn't understand.

You might want to say all that in the draft. It's impossible to infer 
all this without you having said it.

BTW, this is why one of my starting principles was:
   "Choice between fire-and-forget options and reliable-in-order options"

Inner-space-01 gives you the latter. Inner-space-02 (sitting on my 
machine waiting to send) gives you both.

The Unknown Option option only solves the negative part of this 
problem, i.e. inferring /lack/ of support. Inner space solves the 
positive side as well, i.e. inferring both hosts support option X, 
and co-ordinating the point in each half-connection when X starts to 
become enabled.

This makes it much easier to design and implement options that 
require co-ordination between both ends for when a change occurs 
(e.g. a re-key, or switching something on or off), precisely in the 
case you point out, when there is no new data.

This relates to a point I was making to Yoshifumi: simplicity isn't 
just in the scheme itself, it also concerns improving how simple it 
is to design and implement new TCP options.


> > 7) Nit: T/TCP is obsolete.
> >
> > T/TCP had vulnerabilities and was obsoleted by RFC6247.
>
>Yes, that's why the document says:
>
>       Though now obsolete [RFC6247], this is a good
>       example of a directional TCP option.

OK. My mistake.

Cheers



Bob


> > Bob
>
>                 -David

________________________________________________________________
Bob Briscoe,                                                  BT 


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Joe,

At 17:46 04/11/2014, Joe Touch wrote:


>On 11/4/2014 8:52 AM, Bob Briscoe wrote:
>...
> > EDO, tcp4way, EOS, Inner Space /all/ use payload as option space,
> > because if an option like EDO that extends the header is stripped, the
> > data receiver thinks the data offset has reverted to DO, so the place
> > where the options had been placed becomes payload.
> >
> > I explained to Joe in the posting pasted below, why there is a safety
> > issue when header and payload do not necessarily share the same fate.
>
>That's also true for DO, FWIW.
>
>However, the issue of how the option is encoded is orthogonal to these
>solutions. Bob's in-band encoding would work in any of them, but also
>ends up "encoding TCP inside TCP", which can easily lead to escalation
>(as I've pointed out on the list already).

I am encoding TCP /options/ inside TCP Data, which is significantly 
different from encoding TCP inside TCP (which I agree would not be 
appropriate).


>For that reason, I don't support the idea of in-band encoding, but
>that's independent of the way in which the space itself is indicated and
>coordinated.

I believe TCP options should have always been within the TCP Data (as 
a design principle, not just as a hack for middlebox traversal). 
That's where I started from.

See the couple of pages position paper I submitted to the IAB SEMI 
workshop: 
<http://bobbriscoe.net/projects/2020comms/tcp/briscoe-inspace_semi.pdf>



>...
> > This is why Joe has updated EDO to now require an EDO option on every
> > segment, even if no extra option space is needed. Then if the receiver
> > sees the EDO option disappear at a certain point in the connection, it
> > knows that the payload could be corrupt.
> >
> > This is still not as inherently safe as Inner Space.
>
>And neither would be as safe as running encrypted TCP inside TCP.
>
>The question is whether we need to deal with this issue, i.e., whether
>such middlebox asymmetries can occur in practice.

I definitely want to be able to use encryption of TCP data to also 
encrypt TCP options. Again, this is not TCP in TCP. It's TCP options in TCP.


> > ...We need to agree
> > whether latency is part of the design brief.
>
>+1

+2

Vote early, vote often.


Bob


>Joe

________________________________________________________________
Bob Briscoe,                                                  BT 


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On 11/7/2014 1:50 PM, Bob Briscoe wrote:
> Joe,
> 
> At 17:46 04/11/2014, Joe Touch wrote:
> 
> 
>> On 11/4/2014 8:52 AM, Bob Briscoe wrote:
>> ...
>> > EDO, tcp4way, EOS, Inner Space /all/ use payload as option space,
>> > because if an option like EDO that extends the header is stripped, the
>> > data receiver thinks the data offset has reverted to DO, so the place
>> > where the options had been placed becomes payload.
>> >
>> > I explained to Joe in the posting pasted below, why there is a safety
>> > issue when header and payload do not necessarily share the same fate.
>>
>> That's also true for DO, FWIW.
>>
>> However, the issue of how the option is encoded is orthogonal to these
>> solutions. Bob's in-band encoding would work in any of them, but also
>> ends up "encoding TCP inside TCP", which can easily lead to escalation
>> (as I've pointed out on the list already).
> 
> I am encoding TCP /options/ inside TCP Data, which is significantly
> different from encoding TCP inside TCP (which I agree would not be
> appropriate).

There's little difference to me.

>> For that reason, I don't support the idea of in-band encoding, but
>> that's independent of the way in which the space itself is indicated and
>> coordinated.
> 
> I believe TCP options should have always been within the TCP Data (as a
> design principle, not just as a hack for middlebox traversal). That's
> where I started from.

Just as I start from the converse position.

IMO, TCP has a header. It's either extensible or not. If it is, then
anyone who interprets the header *and* knows it can't understand it
KNOWS it's doing something wrong if it acts on it.

Joe


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On 7 November 2014 09:38, Joe Touch <touch@isi.edu> wrote:

> Hi, Bob,
>
> On 11/7/2014 11:26 AM, Bob Briscoe wrote:
> > Joe,
> >
> > At 00:55 06/11/2014, Joe Touch wrote:
> ...
> >> You treat the OOB and SYN as a pair, and treat it with the same time-out
> >> on both ends as far as giving up or retransmitting. How long you keep
> >> queued copies to deal with reordering or other arrival-timing issues is
> >> up to the receiver.
> >
> > Not so IMO. The OOB has to be sent a 'guard time' after the SYN, to
> > minimise the chance of it getting reordered and then getting tangled up
> > in a firewall before the SYN arrives; the firewall will chuck any
> > segment for which it has not seen a prior SYN.
>
> Firewalls aren't the issue. NATs are, and unless reordering is prevalent
> that won't be an issue.
>
> Reordering would mostly happen on multiple paths, but you'd need
> multiple paths between the client and the NAT - i.e., in the private
> side. That's rare if it happens at all.
>

It's rare right now, but there is lots of pressure to make reordering
protection in 802.11 aggregation optional, which will make it quite common
since I would expect it to be off by default on a lot of equipment with
firmware shipping late 2015 and thereafter.

>
> I thus don't agree that guard times are needed, which means we don't
> need to worry about determining them.
>

50ms would certainly do it, although that's more than many RTTs.  5ms would
be adequate in many cases.


>
> > So although the timeouts might be the same it doesn't seem sensible to
> > start the timers at the same time.
>
> I currently believe that there should be one timer. If you need to
> resend, you resend BOTH the SYN and OOB. The server side can cache to
> deal with reordering that happens there, but that doesn't require
> guard-time at client transmission.
>
> Joe
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



-- 
Andrew McGregor | SRE | andrewmcgr@google.com | +61 4 1071 2221

--001a113e44b4f2012905075158a5
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On 7 November 2014 09:38, Joe Touch <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:touch@isi.edu" target=3D"_blank">touch@isi.edu</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">Hi, Bob,<br>
<span class=3D""><br>
On 11/7/2014 11:26 AM, Bob Briscoe wrote:<br>
&gt; Joe,<br>
&gt;<br>
&gt; At 00:55 06/11/2014, Joe Touch wrote:<br>
</span>...<br>
<span class=3D"">&gt;&gt; You treat the OOB and SYN as a pair, and treat it=
 with the same time-out<br>
&gt;&gt; on both ends as far as giving up or retransmitting. How long you k=
eep<br>
&gt;&gt; queued copies to deal with reordering or other arrival-timing issu=
es is<br>
&gt;&gt; up to the receiver.<br>
&gt;<br>
&gt; Not so IMO. The OOB has to be sent a &#39;guard time&#39; after the SY=
N, to<br>
&gt; minimise the chance of it getting reordered and then getting tangled u=
p<br>
&gt; in a firewall before the SYN arrives; the firewall will chuck any<br>
&gt; segment for which it has not seen a prior SYN.<br>
<br>
</span>Firewalls aren&#39;t the issue. NATs are, and unless reordering is p=
revalent<br>
that won&#39;t be an issue.<br>
<br>
Reordering would mostly happen on multiple paths, but you&#39;d need<br>
multiple paths between the client and the NAT - i.e., in the private<br>
side. That&#39;s rare if it happens at all.<br></blockquote><div><br></div>=
<div>It&#39;s rare right now, but there is lots of pressure to make reorder=
ing protection in 802.11 aggregation optional, which will make it quite com=
mon since I would expect it to be off by default on a lot of equipment with=
 firmware shipping late 2015 and thereafter.</div><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex">
<br>
I thus don&#39;t agree that guard times are needed, which means we don&#39;=
t<br>
need to worry about determining them.<br></blockquote><div><br></div><div>5=
0ms would certainly do it, although that&#39;s more than many RTTs. =C2=A05=
ms would be adequate in many cases.</div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
<br>
&gt; So although the timeouts might be the same it doesn&#39;t seem sensibl=
e to<br>
<span class=3D"">&gt; start the timers at the same time.<br>
<br>
</span>I currently believe that there should be one timer. If you need to<b=
r>
resend, you resend BOTH the SYN and OOB. The server side can cache to<br>
deal with reordering that happens there, but that doesn&#39;t require<br>
guard-time at client transmission.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Joe<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div class=3D"gmail_signature"><div dir=3D"ltr"><span style=3D"color:rgb(85=
,85,85);font-family:sans-serif;font-size:small;line-height:1.5em;border-wid=
th:2px 0px 0px;border-style:solid;border-color:rgb(213,15,37);padding-top:2=
px;margin-top:2px">Andrew McGregor=C2=A0|</span><span style=3D"color:rgb(85=
,85,85);font-family:sans-serif;font-size:small;line-height:1.5em;border-wid=
th:2px 0px 0px;border-style:solid;border-color:rgb(51,105,232);padding-top:=
2px;margin-top:2px">=C2=A0SRE=C2=A0|</span><span style=3D"color:rgb(85,85,8=
5);font-family:sans-serif;font-size:small;line-height:1.5em;border-width:2p=
x 0px 0px;border-style:solid;border-color:rgb(0,153,57);padding-top:2px;mar=
gin-top:2px">=C2=A0<a href=3D"mailto:andrewmcgr@google.com" target=3D"_blan=
k">andrewmcgr@google.com</a>=C2=A0|</span><span style=3D"color:rgb(85,85,85=
);font-family:sans-serif;font-size:small;line-height:1.5em;border-width:2px=
 0px 0px;border-style:solid;border-color:rgb(238,178,17);padding-top:2px;ma=
rgin-top:2px">=C2=A0+61 4 1071 2221</span><br></div></div>
</div></div>

--001a113e44b4f2012905075158a5--


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Date: Fri, 07 Nov 2014 20:32:26 -0800
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Subject: Re: [tcpm] Inner Option Space revised: inner-space-01
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On 11/7/2014 8:27 PM, Andrew Mcgregor wrote:
> 
> 
> On 7 November 2014 09:38, Joe Touch <touch@isi.edu
> <mailto:touch@isi.edu>> wrote:
> 
>     Hi, Bob,
> 
>     On 11/7/2014 11:26 AM, Bob Briscoe wrote:
>     > Joe,
>     >
>     > At 00:55 06/11/2014, Joe Touch wrote:
>     ...
>     >> You treat the OOB and SYN as a pair, and treat it with the same time-out
>     >> on both ends as far as giving up or retransmitting. How long you keep
>     >> queued copies to deal with reordering or other arrival-timing issues is
>     >> up to the receiver.
>     >
>     > Not so IMO. The OOB has to be sent a 'guard time' after the SYN, to
>     > minimise the chance of it getting reordered and then getting tangled up
>     > in a firewall before the SYN arrives; the firewall will chuck any
>     > segment for which it has not seen a prior SYN.
> 
>     Firewalls aren't the issue. NATs are, and unless reordering is prevalent
>     that won't be an issue.
> 
>     Reordering would mostly happen on multiple paths, but you'd need
>     multiple paths between the client and the NAT - i.e., in the private
>     side. That's rare if it happens at all.
> 
> It's rare right now, but there is lots of pressure to make reordering
> protection in 802.11 aggregation optional, which will make it quite
> common since I would expect it to be off by default on a lot of
> equipment with firmware shipping late 2015 and thereafter.

Reordering protection isn't the same as whether reordering will
typically happen.

However, we're talking about the packets with the same src/dst
addresses. Why would 802.11 aggregation affect their order? (I.e., I can
see it affecting the relative order of packets to/from different
destinations, but not the same ones)...

Joe


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--047d7bd6ba9a769651050751ab6d
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On 7 November 2014 18:32, Joe Touch <touch@isi.edu> wrote:

>
>
> On 11/7/2014 8:27 PM, Andrew Mcgregor wrote:
> >
>
> > It's rare right now, but there is lots of pressure to make reordering
> > protection in 802.11 aggregation optional, which will make it quite
> > common since I would expect it to be off by default on a lot of
> > equipment with firmware shipping late 2015 and thereafter.
>
> Reordering protection isn't the same as whether reordering will
> typically happen.
>
> However, we're talking about the packets with the same src/dst
> addresses. Why would 802.11 aggregation affect their order? (I.e., I can
> see it affecting the relative order of packets to/from different
> destinations, but not the same ones)...
>
> Joe
>

The 802.11 MAC can aggregate multiple IP packets into what's called an
AMPDU (and has to, in order to achieve decent performance).  Individual
packets within an AMPDU are checksummed separately, and can be
retransmitted individually or as part of the next AMPDU.  Right now,
reordering protection requires that IP packets (strictly, ethernet frames)
that have been correctly received but are behind (in link-layer sequence
number space) one that has not been received or explicitly NAKed have to be
buffered until receipt of the delaying frame or a NAK or a rather long
timeout.  Since the raw single-shot failure rate can be up to around 45%,
this happens a lot, and experiment indicates that if the traffic is
tolerant to a reasonable reordering window (as most current TCPs are, for
example) end-to-end performance is significantly better with reordering
protection disabled.

So yes, if reordering protection is off, packets within a single flow will
get reordered fairly frequently.

-- 
Andrew McGregor | SRE | andrewmcgr@google.com | +61 4 1071 2221

--047d7bd6ba9a769651050751ab6d
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On 7 November 2014 18:32, Joe Touch <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:touch@isi.edu" target=3D"_blank">touch@isi.edu</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid=
;padding-left:1ex"><span class=3D""><br>
<br>
On 11/7/2014 8:27 PM, Andrew Mcgregor wrote:<br>
&gt;<br></span><span class=3D""><br>
&gt; It&#39;s rare right now, but there is lots of pressure to make reorder=
ing<br>
&gt; protection in 802.11 aggregation optional, which will make it quite<br=
>
&gt; common since I would expect it to be off by default on a lot of<br>
&gt; equipment with firmware shipping late 2015 and thereafter.<br>
<br>
</span>Reordering protection isn&#39;t the same as whether reordering will<=
br>
typically happen.<br>
<br>
However, we&#39;re talking about the packets with the same src/dst<br>
addresses. Why would 802.11 aggregation affect their order? (I.e., I can<br=
>
see it affecting the relative order of packets to/from different<br>
destinations, but not the same ones)...<br>
<span class=3D""><font color=3D"#888888"><br>
Joe<br>
</font></span></blockquote></div><br>The 802.11 MAC can aggregate multiple =
IP packets into what&#39;s called an AMPDU (and has to, in order to achieve=
 decent performance).=C2=A0 Individual packets within an AMPDU are checksum=
med separately, and can be retransmitted individually or as part of the nex=
t AMPDU.=C2=A0 Right now, reordering protection requires that IP packets (s=
trictly, ethernet frames) that have been correctly received but are behind =
(in link-layer sequence number space) one that has not been received or exp=
licitly NAKed have to be buffered until receipt of the delaying frame or a =
NAK or a rather long timeout.=C2=A0 Since the raw single-shot failure rate =
can be up to around 45%, this happens a lot, and experiment indicates that =
if the traffic is tolerant to a reasonable reordering window (as most curre=
nt TCPs are, for example) end-to-end performance is significantly better wi=
th reordering protection disabled.</div><div class=3D"gmail_extra"><br></di=
v><div class=3D"gmail_extra">So yes, if reordering protection is off, packe=
ts within a single flow will get reordered fairly frequently.</div><div cla=
ss=3D"gmail_extra"><div><br></div>-- <br><div class=3D"gmail_signature"><di=
v dir=3D"ltr"><span style=3D"color:rgb(85,85,85);font-family:sans-serif;fon=
t-size:small;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;=
border-color:rgb(213,15,37);padding-top:2px;margin-top:2px">Andrew McGregor=
=C2=A0|</span><span style=3D"color:rgb(85,85,85);font-family:sans-serif;fon=
t-size:small;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;=
border-color:rgb(51,105,232);padding-top:2px;margin-top:2px">=C2=A0SRE=C2=
=A0|</span><span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-s=
ize:small;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;bor=
der-color:rgb(0,153,57);padding-top:2px;margin-top:2px">=C2=A0<a href=3D"ma=
ilto:andrewmcgr@google.com" target=3D"_blank">andrewmcgr@google.com</a>=C2=
=A0|</span><span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-s=
ize:small;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;bor=
der-color:rgb(238,178,17);padding-top:2px;margin-top:2px">=C2=A0+61 4 1071 =
2221</span><br></div></div>
</div></div>

--047d7bd6ba9a769651050751ab6d--


From nobody Fri Nov  7 21:12:37 2014
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Subject: Re: [tcpm] Inner Option Space revised: inner-space-01
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On 11/7/2014 8:50 PM, Andrew Mcgregor wrote:
...
>     Reordering protection isn't the same as whether reordering will
>     typically happen.
> 
>     However, we're talking about the packets with the same src/dst
>     addresses. Why would 802.11 aggregation affect their order? (I.e., I can
>     see it affecting the relative order of packets to/from different
>     destinations, but not the same ones)...
> 
>     Joe
> 
> The 802.11 MAC can aggregate multiple IP packets into what's called an
> AMPDU (and has to, in order to achieve decent performance).  Individual
> packets within an AMPDU are checksummed separately, and can be
> retransmitted individually or as part of the next AMPDU.  Right now,
> reordering protection requires that IP packets (strictly, ethernet
> frames) that have been correctly received but are behind (in link-layer
> sequence number space) one that has not been received or explicitly
> NAKed have to be buffered until receipt of the delaying frame or a NAK
> or a rather long timeout.  Since the raw single-shot failure rate can be
> up to around 45%, this happens a lot, and experiment indicates that if
> the traffic is tolerant to a reasonable reordering window (as most
> current TCPs are, for example) end-to-end performance is significantly
> better with reordering protection disabled.
>
> So yes, if reordering protection is off, packets within a single flow
> will get reordered fairly frequently.

The client can still get through a NAT a few different ways:

	- a gap (as suggested before)

	- bursting a small number of copies of SYN/OOB pairs
		the first SYN through a NAT would make way for
		a following OOB

First, though, tt'd be useful to understand what the error properties
are, esp. whether they're burst or uncorrellated.

And - with a single-shot failure rate that high, why FEC isn't being
considered.

Joe


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From: Andrew Mcgregor <andrewmcgr@google.com>
To: Joe Touch <touch@isi.edu>
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--001a113b0166c5e9580507521bbe
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On 7 November 2014 19:11, Joe Touch <touch@isi.edu> wrote:

>
>
> On 11/7/2014 8:50 PM, Andrew Mcgregor wrote:
> ...
> >     Reordering protection isn't the same as whether reordering will
> >     typically happen.
> >
> >     However, we're talking about the packets with the same src/dst
> >     addresses. Why would 802.11 aggregation affect their order? (I.e., I
> can
> >     see it affecting the relative order of packets to/from different
> >     destinations, but not the same ones)...
> >
> >     Joe
> >
> > The 802.11 MAC can aggregate multiple IP packets into what's called an
> > AMPDU (and has to, in order to achieve decent performance).  Individual
> > packets within an AMPDU are checksummed separately, and can be
> > retransmitted individually or as part of the next AMPDU.  Right now,
> > reordering protection requires that IP packets (strictly, ethernet
> > frames) that have been correctly received but are behind (in link-layer
> > sequence number space) one that has not been received or explicitly
> > NAKed have to be buffered until receipt of the delaying frame or a NAK
> > or a rather long timeout.  Since the raw single-shot failure rate can be
> > up to around 45%, this happens a lot, and experiment indicates that if
> > the traffic is tolerant to a reasonable reordering window (as most
> > current TCPs are, for example) end-to-end performance is significantly
> > better with reordering protection disabled.
> >
> > So yes, if reordering protection is off, packets within a single flow
> > will get reordered fairly frequently.
>
> The client can still get through a NAT a few different ways:
>
>         - a gap (as suggested before)
>
>         - bursting a small number of copies of SYN/OOB pairs
>                 the first SYN through a NAT would make way for
>                 a following OOB
>
> Indeed, those can work.


> First, though, tt'd be useful to understand what the error properties
> are, esp. whether they're burst or uncorrellated.
>

Tend to be bursty, there are some studies on this so data is available.

>
> And - with a single-shot failure rate that high, why FEC isn't being
> considered.
>
> Joe
>

I don't know for sure, but if I had to speculate I'd say the answer would
be 'patents'.

-- 
Andrew McGregor | SRE | andrewmcgr@google.com | +61 4 1071 2221

--001a113b0166c5e9580507521bbe
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On 7 November 2014 19:11, Joe Touch <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:touch@isi.edu" target=3D"_blank">touch@isi.edu</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex"><br>
<br>
On 11/7/2014 8:50 PM, Andrew Mcgregor wrote:<br>
...<br>
<span class=3D"">&gt;=C2=A0 =C2=A0 =C2=A0Reordering protection isn&#39;t th=
e same as whether reordering will<br>
&gt;=C2=A0 =C2=A0 =C2=A0typically happen.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0However, we&#39;re talking about the packets with t=
he same src/dst<br>
&gt;=C2=A0 =C2=A0 =C2=A0addresses. Why would 802.11 aggregation affect thei=
r order? (I.e., I can<br>
&gt;=C2=A0 =C2=A0 =C2=A0see it affecting the relative order of packets to/f=
rom different<br>
&gt;=C2=A0 =C2=A0 =C2=A0destinations, but not the same ones)...<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0Joe<br>
&gt;<br>
&gt; The 802.11 MAC can aggregate multiple IP packets into what&#39;s calle=
d an<br>
&gt; AMPDU (and has to, in order to achieve decent performance).=C2=A0 Indi=
vidual<br>
&gt; packets within an AMPDU are checksummed separately, and can be<br>
&gt; retransmitted individually or as part of the next AMPDU.=C2=A0 Right n=
ow,<br>
&gt; reordering protection requires that IP packets (strictly, ethernet<br>
&gt; frames) that have been correctly received but are behind (in link-laye=
r<br>
&gt; sequence number space) one that has not been received or explicitly<br=
>
&gt; NAKed have to be buffered until receipt of the delaying frame or a NAK=
<br>
&gt; or a rather long timeout.=C2=A0 Since the raw single-shot failure rate=
 can be<br>
&gt; up to around 45%, this happens a lot, and experiment indicates that if=
<br>
&gt; the traffic is tolerant to a reasonable reordering window (as most<br>
&gt; current TCPs are, for example) end-to-end performance is significantly=
<br>
&gt; better with reordering protection disabled.<br>
&gt;<br>
&gt; So yes, if reordering protection is off, packets within a single flow<=
br>
&gt; will get reordered fairly frequently.<br>
<br>
</span>The client can still get through a NAT a few different ways:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 - a gap (as suggested before)<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 - bursting a small number of copies of SYN/OOB =
pairs<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 the first SYN throu=
gh a NAT would make way for<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 a following OOB<br>
<br></blockquote><div>Indeed, those can work.</div><div>=C2=A0</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">
First, though, tt&#39;d be useful to understand what the error properties<b=
r>
are, esp. whether they&#39;re burst or uncorrellated.<br></blockquote><div>=
<br></div><div>Tend to be bursty, there are some studies on this so data is=
 available.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
And - with a single-shot failure rate that high, why FEC isn&#39;t being<br=
>
considered.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Joe<br>
</font></span></blockquote></div><br>I don&#39;t know for sure, but if I ha=
d to speculate I&#39;d say the answer would be &#39;patents&#39;.<br clear=
=3D"all"><div><br></div>-- <br><div class=3D"gmail_signature"><div dir=3D"l=
tr"><span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:sma=
ll;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-col=
or:rgb(213,15,37);padding-top:2px;margin-top:2px">Andrew McGregor=C2=A0|</s=
pan><span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:sma=
ll;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-col=
or:rgb(51,105,232);padding-top:2px;margin-top:2px">=C2=A0SRE=C2=A0|</span><=
span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:small;li=
ne-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-color:rg=
b(0,153,57);padding-top:2px;margin-top:2px">=C2=A0<a href=3D"mailto:andrewm=
cgr@google.com" target=3D"_blank">andrewmcgr@google.com</a>=C2=A0|</span><s=
pan style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:small;lin=
e-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-color:rgb=
(238,178,17);padding-top:2px;margin-top:2px">=C2=A0+61 4 1071 2221</span><b=
r></div></div>
</div></div>

--001a113b0166c5e9580507521bbe--


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From nobody Sat Nov  8 19:49:50 2014
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Subject: Re: [tcpm] Comments to draft-ietf-tcpm-tcp-edo-01
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Hi, Alex,

Thanks for the feedback.

Some comments below.

Joe

On 11/8/2014 12:58 PM, Zimmermann, Alexander wrote:
> Hi Joe,
> 
> yesterday I found some time to re-read the draft. A couple of points
> 
> *** Technical ***
> 
> * The whole drafts reads like we have *two* EDO options, aka the „TCP EDO
> request option“ and the „TCP EDO length option“, but actually we only have
> one option, the EDO option. I found this a bit confusing. Saying this, why
> do we not include the Header_length field in any EDO frame, i.e. also in
> the SYN. For signaling (EDO negotiation) we can use values for the
> Header_length that are less than the DO value.

The SYN is the last place we want to burn space that has no purpose.
That's why the option has two variants - the signal and the actual use.
I'll see what I can do to clarify that.

> * Page 4: „A null EDO length option contains the same value as the DO field,
> i.e., it does not extend the TCP option space“ Personally, I don’t like the
> term „null“ because it gives (at least me) the wrong impression that the
> Header_length contains 0. Could we say „neutral“ instead of „null“?

"redundant" might be more accurate; point taken.

> * Page 5: „When processing a segment, EDO needs to be visible within the
> area indicated by the Data Offset field“. Be more expect here, by saying
> that the EDO option must be in the first 40 Bytes after the fixed TCP
> header. IMO it makes it a bit more clearer.

I will clarify, but it's not that it's just in the first 40 bytes. It's
that it's within the space indicated by DO as well (DO can be less than
15 - and EDO needs to be visible within that area).

> *** Editorial ***
> 
> * Page 6: Personally, I would swap the text and speak first about the
> placement of the EDO option and then about the alignment.
> 
> * Section 7 Related Work - into an appendix?

I don't have the draft in front of me; I'll take a look at these as
well, but I'd like to hear from others before making such changes.

> *** Nits ***
> 
> * page 13: „The following discusses three of the most notable past …“
> Actually, you discuss 4 approaches.

Agreed; this isn't a Monty Python skit ;-)

> * Sec. 12: Acknowledgments. Last word. There is a missing „n“ at the
> end of the name :-)

Will fix, of course.

---


From nobody Sun Nov  9 04:18:13 2014
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Hi Michael,

Thanks for your comments!  Replies inline below.

On 2014-11-03 00:33, Scharf, Michael (Michael) wrote:
> Hi,
>
> I've read -04 and I have a couple of questions and suggestions:
>
> * Section 4: The "total number of outstanding and previously unsent seg=
ments" in sum of two variables, right? If so, why not use the term "sum"?=


Correct, so "sum" is better. We will update this.

>
> * Section 4: I also wonder if the text should define "previously unsent=
 segments". The term "previously unsent data" is very well-known in the R=
FC series. A quick search revealed that "previously unsent segments" is u=
sually only used to refer to situations when segments with this property =
have indeed been transmitted. Is my understanding correct that draft-ietf=
-tcpm-rtorestart-04 uses this term differently? I think it here refers to=
 data that will be sent *in future*. In that case, I wonder if it is real=
ly clear how the sender can determine the number of "previously unsent se=
gments". For instance, that number may depend on the segmentation strateg=
y of the sender, in particular for applications that use many small write=
() calls. In this case a TCP stack may e.g. decide to send partial segmen=
ts. I am not sure if the actual number of segments created from the "unse=
nt data" is really known in advance always. Or do I miss something?

You are correct that "previously unsent segments" refers to data that=20
will be sent in the future. This is the same meaning as in RFC 3517 and=20
RFC 4653. A definition of this term could be added to Section 2, but I=20
am not sure if it is needed? To better understand the possible conflict=20
in terminology, could you point to some of RFCs that use the term to=20
refer to already transmitted segments? Intuitively that seems a bit=20
strange to me.

As discussed in section 5.3, the number of unsent segments can be=20
estimated from the amount of unsent data and the SMSS. Note that we are=20
only talking about data that is already available in the send queue, and =

later write calls can of course create additional segments. I guess a=20
TCP stack that has one SMSS of data in its send queue is free to send=20
this as multiple segments if it wants to, but I assume that would not be =

the normal behavior. As the estimate is not very critical it is not=20
really a problem if this happens occasionally.

The addition of "previously unsent segments" is there to capture a=20
corner case (see issue raised by Alex in=20
http://www.ietf.org/mail-archive/web/tcpm/current/msg08780.html), so it=20
will only be occasionally used. If we then get an incorrect estimate,=20
say a sender with one SMSS of data would instead send this as two SMSS,=20
and the algorithm triggers "incorrectly", this would be equivalent to=20
using a RTOR threshold of rrthresh + 1 so nothing critical happens. (In=20
some cases rrthresh + 1 may even be better.)

>
> * Section 5.1: "Given RTOR's ability to only work when it is beneficial=
 for the loss recovery process, it is suitable as a system-wide default m=
echanism for TCP traffic." I think other text in the draft asks for furth=
er experimentation regarding the actual trade-off between benefit and ris=
k. Given that, I think another wording should be used (e.g., "Given that =
RTOR is a mostly conservative algorithm", ...). Given that this is not a =
PS document, I'd also prefer a more careful phrasing regarding the recomm=
endation as system-wide default (e.g., "it is suitable for experimentatio=
n as system-wide default").

Agreed, we will update the text.

> * Section 5.2: As mentioned in the last face-to-face meeting, spurious =
timeout negatively affect the network by needlessly sending data. This ne=
gative effect is not mentioned in this section. Why?

The initial intention was to focus on aspects of spurious timeouts that=20
were specific to RTOR. But looking at the text I see that this is not at =

all clear in the current version. The best resolution is probably to not =

try to scope it that way. We can add some text also about the increased=20
network load in the next version.

>
> * Section 5.2: Mobile networks have a variable RTT, including e.g. long=
er delays during handovers (either without or with transient loss during =
the handover). A lot of past work on spurious timeout recovery has been m=
otivated by these delay spikes. Reducing the RTO almost certainly has an =
impact in such environments; I think performance can both increase or dec=
rease. If no data on that is available, mobile networks seem to me an are=
a where more experimentation would be particularly useful, right?

Yes, as the RTTs in mobile networks can be higher and are often variable =

it is an interesting area. It could be mentioned as an example in 5.2.

>
> * Section 5.2: Is there any data on the risk of spurious timeouts in ne=
tworks where RTT and the minimum RTO are of the same order of magnitude? =
For instance, in satellite networks the RTT can be huge.

Our Linux experiments over emulated networks cover RTTs that go beyond=20
the value of the minimum RTO, i.e. RTTs larger than 200 ms, see=20
http://riteproject.eu/resources/rto-restart/ for some sample results. I=20
am not aware of any real experiments in satellite networks or similar. A =

general problem when experimenting with this is that spurious timeouts=20
happen only occasionally so a lot of data is needed before you can=20
determine any trends.

>
> * Section 5.2: "However, with respect to RTOR spurious timeouts are onl=
y a problem for applications transmitting multiple bursts of data within =
a single flow." I think this section should be reworded and it should mor=
e explicitly discuss the impact of RTOR on HTTP/1.1 and HTTP/2.0 traffic,=
 including e.g. adaptive video streaming. They all transmit "multiple bur=
sts of data within a single flow". To me, the current sentence could impl=
y that RTOR would be a "problem" for a vast majority of Internet traffic.=
 A more explicit discussion on how RTOR affects HTTP/1.1 and HTTP/2.0 wou=
ld IMHO make a lot of sense.

Ok, we will work on the wording and use HTTP as an example. And also try =

to better highlight the tradeoffs involved. In our experiments with=20
HTTP/1.1 traffic RTOR performs better than the baseline.

>
> * Section 5.3: Given that not all stacks are segment-based, this sectio=
n seems really relevant. However, the description how to determine outsta=
nding segments is vague ("it is possible to exactly determine..."). I thi=
nk this could be reworded to provide more guidance to implementers. The s=
ame applies to the calculation of "previously unsent segments"; for insta=
nce, the text does not explain what happens if the "number of bytes in th=
e send queue" is not a multiple of the SMSS (see also above regarding the=
 definition of this variable).

OK, we can reuse the full discussion on how to determine outstanding=20
segments from RFC 5827 rather than summarizing and pointing to it. We=20
can of course also clarify the number of segments to be estimated when=20
the "number of bytes in the send queue" is not a multiple of the SMSS. I =

somehow thought this was obvious.

>
> * Section 6: "In contrast, RTOR is trying to make the RTO more appropri=
ate in cases where there is no need to be overly cautious." I think this =
sentence should use a more neutral language. I think whether an RTO value=
 is "appropriate" is impossible to know (in advance), and whether a more =
aggressive timeout is "cautious" or not may depend on the network under c=
onsideration. For instance, an alternative would be to compare the comple=
xity of RTOR and TLP.

Ok. We will look over the wording in this part.

>
> * Section 9: I wonder if an attacker could try to send certain ACK patt=
erns to increase the risk of timeouts e.g. at a Web Server, leveraging th=
e smaller RTO value. If successful, the RTOs could significantly reduce a=
pplication performance. Probably, such an attacker could cause much more =
harm by other means, i.e., this may not be a significant security risk. B=
ut the current security analysis is very short.

The section is very short, but this is because we do not really see any=20
new security issues.

  If I understand your example above correctly, the receiver is trying=20
to somehow reduce performance for its own application by manipulating=20
the ACK pattern. But if you are trying to mess up things for your own=20
application you can deliver incorrect data, throw away any data received =

or do whatever you want so I do not think trying to possibly slow down=20
the sender is the thing to worry about in this attack scenario. If=20
someone finds a relevant security issue it should of course be=20
documented, but I do not think it makes sense to put in something like=20
the above just to make the section longer.

Thanks again for your comments!

Cheers,
Anna

>
> Thanks
>
> Michael
>
>
>> -----Original Message-----
>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Per Hurtig
>> Sent: Tuesday, October 28, 2014 12:18 AM
>> To: tcpm@ietf.org
>> Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rtorestart-04.txt
>>
>> Hi all,
>>
>> we have now updated the draft to address the outstanding comments. The=

>> main changes from last version are:
>>
>>     o  Changed the algorithm to allow RTOR when there is unsent data
>> available, but the cwnd does not allow transmission.
>>     o  Changed the algorithm to not trigger if "RTO - T_earliest" <=3D=
 0,
>> to avoid that ACKs to previous retransmissions trigger premature
>> timeouts.
>>     o  Made minor adjustments throughout the document to adjust for th=
e
>> algorithmic changes.
>>     o  Improved the wording throughout the document.
>>
>>
>> for experimental results and a Linux implementation, please visit:
>> http://riteproject.eu/resources/rto-restart/
>>
>>
>>
>> Cheers,
>> Per
>>
>> On 2014-10-28 00:11, internet-drafts@ietf.org wrote:
>>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>>    This draft is a work item of the TCP Maintenance and Minor
>> Extensions Working Group of the IETF.
>>>           Title           : TCP and SCTP RTO Restart
>>>           Authors         : Per Hurtig
>>>                             Anna Brunstrom
>>>                             Andreas Petlund
>>>                             Michael Welzl
>>> 	Filename        : draft-ietf-tcpm-rtorestart-04.txt
>>> 	Pages           : 13
>>> 	Date            : 2014-10-27
>>>
>>> Abstract:
>>>      This document describes a modified algorithm for managing the TC=
P
>> and
>>>      SCTP retransmission timers that provides faster loss recovery
>> when
>>>      there is a small amount of outstanding data for a connection.
>> The
>>>      modification, RTO Restart (RTOR), allows the transport to restar=
t
>> its
>>>      retransmission timer more aggressively in situations where fast
>>>      retransmit cannot be used.  This enables faster loss detection
>> and
>>>      recovery for connections that are short-lived or application-
>> limited.
>>>
>>> The IETF datatracker status page for this draft is:
>>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-rtorestart/
>>>
>>> There's also a htmlized version available at:
>>> http://tools.ietf.org/html/draft-ietf-tcpm-rtorestart-04
>>>
>>> A diff from the previous version is available at:
>>> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rtorestart-04
>>>
>>>
>>> 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/
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>
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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
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Thread-Topic: Comments to draft-ietf-tcpm-tcp-edo-01
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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: Joe Touch <touch@isi.edu>
Thread-Topic: Comments to draft-ietf-tcpm-tcp-edo-01
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Date: Sun, 09 Nov 2014 12:03:41 -0800
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Comments below...

On 11/9/2014 11:54 AM, Zimmermann, Alexander wrote:
> Hi Joe,
> 
>> Am 08.11.2014 um 17:49 schrieb Joe Touch <touch@isi.edu>:
>>
>> Hi, Alex,
>>
>> Thanks for the feedback.
>>
>> Some comments below.
>>
>> Joe
>>
>> On 11/8/2014 12:58 PM, Zimmermann, Alexander wrote:
>>> Hi Joe,
>>>
>>> yesterday I found some time to re-read the draft. A couple of points
>>>
>>> *** Technical ***
>>>
>>> * The whole drafts reads like we have *two* EDO options, aka the „TCP EDO
>>> request option“ and the „TCP EDO length option“, but actually we only have
>>> one option, the EDO option. I found this a bit confusing. Saying this, why
>>> do we not include the Header_length field in any EDO frame, i.e. also in
>>> the SYN. For signaling (EDO negotiation) we can use values for the
>>> Header_length that are less than the DO value.
>>
>> The SYN is the last place we want to burn space that has no purpose.
> 
> OK, I see. Point taken. But the same applies to the SYN/ACK too. 

Yes. The SYN/ACK either needs to similarly conserve space and use just
the flag variant *if* EDO extension isn't allowed (that's how the
current spec is written).

If we allow use of the EDO extension space in the SYN/ACK (e.g., if
we're not as worried about asymmetric middlebox corruption), then we'd
use the regular EDO variant which would extend the space --- in that
case, the space the option consumes isn't an issue.

> So,
> additional question: why do we not send „TCP EDO request option“ back
> to the client?

That's what the current draft says to do for SYN/ACK, for the reason above.


> Moreover, w/ RFC2018 we have more or less the same. Some negotiation and
> the actual usage of the option. Any harm to burn two „kinds“ like RFC2018?

We've since learned better ;-)

In specific, we want fate-sharing of the option kind. We don't want the
impact of a bug that works on one kind but doesn't handle the other.

There's also no benefit. The difference between the option types can be
determined by the length of a single type.

Joe


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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: Joe Touch <touch@isi.edu>
Thread-Topic: Comments to draft-ietf-tcpm-tcp-edo-01
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Subject: Re: [tcpm] Comments to draft-ietf-tcpm-tcp-edo-01
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From nobody Sun Nov  9 16:36:35 2014
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Date: Sun, 09 Nov 2014 19:36:31 -0500
From: Brandon Williams <brandon.williams@akamai.com>
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To: "Sujeet Nayak A (sua)" <sua@cisco.com>,  "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: Re: [tcpm] Updated SHA-2 AO draft
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Hi Sujeet,

I'm still concerned about whether there's enough space in the SYN header 
for this option. My concern is based mostly on a what I think is a flaw 
in the analysis from RFC5925 section 7.2, which leaves out the 4-byte 
MSS option. As the analysis from RFC6824 indicates, SYN packets 
typically include the MSS option, leaving only 21 bytes available in an 
option packed SYN and 16 in a worst-case unpacked SYN. TCP-AO with SHA1 
works in either case, but SHA2 only works (just barely) in an option 
packed stack.

Section 4 of your draft should probably be updated to account for the 
commonality of the MSS option and the related requirement either to drop 
one of the common SYN options or pack the option space.

--Brandon

On 10/28/2014 02:12 AM, Sujeet Nayak A (sua) wrote:
> Hi,
> Thanks everyone for your valuable review comments so far. Brian and
> myself have updated the draft to produce the next version.
> https://tools.ietf.org/html/draft-nayak-tcp-sha2-01
>
> Some of the high level changes made are:
>
>   * Because of TCP option space issue, SHA512 has been moved out of the
>     draft (a note added in the "Security Consideration" for future
>     support, when needed).
>   * Moved the motivation contents into the introduction section.
>   * Taken care of some of the RFC language related comments.
>
> Pl. review and let me know your feedback. On the other hand, if there is
> a consensus that, the contents need to update RFC5926, and if that RFC
> allows such an update, then we are happy to work with Greg on it.
>
> Regards,
>
> Sujeet

-- 
Brandon Williams; Senior Principal Software Engineer
Emerging Products Engineering; Akamai Technologies Inc.


From nobody Sun Nov  9 16:39:51 2014
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Subject: Re: [tcpm] Comments to draft-ietf-tcpm-tcp-edo-01
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> On Nov 9, 2014, at 1:28 PM, Zimmermann, Alexander <Alexander.Zimmermann@ne=
tapp.com> wrote:
>=20
>=20
>> Am 09.11.2014 um 10:03 schrieb Joe Touch <touch@isi.edu>:
>>=20
>> Comments below=E2=80=A6
>=20
> Dito
>=20
>>=20
>>> On 11/9/2014 11:54 AM, Zimmermann, Alexander wrote:
>>> Hi Joe,
>>>=20
>>>> Am 08.11.2014 um 17:49 schrieb Joe Touch <touch@isi.edu>:
>>>>=20
>>>> Hi, Alex,
>>>>=20
>>>> Thanks for the feedback.
>>>>=20
>>>> Some comments below.
>>>>=20
>>>> Joe
>>>>=20
>>>>> On 11/8/2014 12:58 PM, Zimmermann, Alexander wrote:
>>>>> Hi Joe,
>>>>>=20
>>>>> yesterday I found some time to re-read the draft. A couple of points
>>>>>=20
>>>>> *** Technical ***
>>>>>=20
>>>>> * The whole drafts reads like we have *two* EDO options, aka the =E2=80=
=9ETCP EDO
>>>>> request option=E2=80=9C and the =E2=80=9ETCP EDO length option=E2=80=9C=
, but actually we only have
>>>>> one option, the EDO option. I found this a bit confusing. Saying this,=
 why
>>>>> do we not include the Header_length field in any EDO frame, i.e. also i=
n
>>>>> the SYN. For signaling (EDO negotiation) we can use values for the
>>>>> Header_length that are less than the DO value.
>>>>=20
>>>> The SYN is the last place we want to burn space that has no purpose.
>>>=20
>>> OK, I see. Point taken. But the same applies to the SYN/ACK too.
>>=20
>> Yes. The SYN/ACK either needs to similarly conserve space and use just
>> the flag variant *if* EDO extension isn't allowed (that's how the
>> current spec is written).
>>=20
>> If we allow use of the EDO extension space in the SYN/ACK (e.g., if
>> we're not as worried about asymmetric middlebox corruption), then we'd
>> use the regular EDO variant which would extend the space --- in that
>> case, the space the option consumes isn't an issue.
>>=20
>>> So,
>>> additional question: why do we not send =E2=80=9ETCP EDO request option=E2=
=80=9C back
>>> to the client?
>>=20
>> That=E2=80=99s what the current draft says to do for SYN/ACK, for the rea=
son above.
>=20
> IMO, that not the case. Page 4, sec 4, second para:

Yes.  I had thought about that after the revision and thought I changed it.=20=


>=20
> =E2=80=9E If receiver of that SYN agrees to support EDO, it responds with a=
 null EDO
> length option in the SYN/ACK. A null EDO length option contains the same v=
alue
> as the DO field, i.e., it does not extend the TCP option space.=E2=80=9C
>=20
> If I understand this correctly the server sends back the =E2=80=9ETCP EDO l=
ength option=E2=80=9C
> meaning that the =E2=80=9EHeader_length=E2=80=9C is non empty. If we all a=
gree (and I=E2=80=99m not sure
> about this) that asymmetric middle box corruption is common and we cannot
> extent the option space in a SYN/ACK via EDO, why we could not echo back:
>=20
> +--------+--------+
> |  Kind  | Length |
> +--------+--------+
>=20
> from the server side?

Yes. That's what I had intended but missed in the revision. Sorry for the co=
nfusion. I'll fix in the next rev.=20

Joe

>=20
> Alex
>=20
>>=20
>>=20
>>> Moreover, w/ RFC2018 we have more or less the same. Some negotiation and=

>>> the actual usage of the option. Any harm to burn two =E2=80=9Ekinds=E2=80=
=9C like RFC2018?
>>=20
>> We've since learned better ;-)
>>=20
>> In specific, we want fate-sharing of the option kind. We don't want the
>> impact of a bug that works on one kind but doesn't handle the other.
>>=20
>> There's also no benefit. The difference between the option types can be
>> determined by the length of a single type.
>>=20
>> Joe
>=20


From nobody Mon Nov 10 08:00:11 2014
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Anna Brunstrom <anna.brunstrom@kau.se>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] draft-ietf-tcpm-rtorestart-04
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Subject: Re: [tcpm] draft-ietf-tcpm-rtorestart-04
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Hi Anna,

Some follow-up comments. I omit the parts for which there seems to be agree=
ment on the need for a document update.

> > * Section 4: I also wonder if the text should define "previously
> unsent segments". The term "previously unsent data" is very well-known
> in the RFC series. A quick search revealed that "previously unsent
> segments" is usually only used to refer to situations when segments
> with this property have indeed been transmitted. Is my understanding
> correct that draft-ietf-tcpm-rtorestart-04 uses this term differently?
> I think it here refers to data that will be sent *in future*. In that
> case, I wonder if it is really clear how the sender can determine the
> number of "previously unsent segments". For instance, that number may
> depend on the segmentation strategy of the sender, in particular for
> applications that use many small write() calls. In this case a TCP
> stack may e.g. decide to send partial segments. I am not sure if the
> actual number of segments created from the "unsent data" is really
> known in advance always. Or do I miss something?
>=20
> You are correct that "previously unsent segments" refers to data that
> will be sent in the future. This is the same meaning as in RFC 3517 and
> RFC 4653. A definition of this term could be added to Section 2, but I
> am not sure if it is needed? To better understand the possible conflict
> in terminology, could you point to some of RFCs that use the term to
> refer to already transmitted segments? Intuitively that seems a bit
> strange to me.

I didn't say that there is a conflict in terminology. I might be wrong, but=
 to me draft-ietf-tcpm-rtorestart-04 defines the term "previously unsent se=
gments" as a *new* TCP state variable.

The value of that variable may be obvious for certain architectures of a TC=
P stack, in particular if segmentation into packets occurs on socket write(=
) calls, i.e., before data is actually sent to network. As far as I recall,=
 this is how the Linux stack works internally. But I'd like to avoid that a=
n RFC describes an algorithm that can only be implemented for certain archi=
tecture of a TCP stack.

RFC 3517 and RFC 4653 either refer to the "amount of previously unsent data=
" (which doesn't need to be further defined IMHO), or the "transmission of =
previously unsent segments" (which is not what draft-ietf-tcpm-rtorestart-0=
4 refers to). For instance, here is the exact wording of the closest matche=
s I could find in RFC 3517 and RFC 4563:

RFC 3517 Section 5 Processing and Acting Upon SACK Information

     "(2) If no sequence number 'S2' per rule (1) exists but there
          exists available unsent data and the receiver's advertised
          window allows, the sequence range of one segment of up to SMSS
          octets of previously unsent data starting with sequence number
          HighData+1 MUST be returned."

   =3D> "previously unsent data", NOT segments

RFC 3517 Section 5 Algorithm Details

  "Note: The first and second
   duplicate ACKs can also be used to trigger the transmission of
   previously unsent segments using the Limited Transmit algorithm
   [RFC3042]."

   =3D> "transmission of previously unsent segments", NOT the number of seg=
ments queued

RFC 4653 Section 2 NCR Description

  "The first Extended Limited Transmit variant, Careful Limited
   Transmit, calls for the transmission of one previously unsent
   segment, in response to duplicate acknowledgments, for every two
   segments that are known to have left the network."

   =3D> "transmission of previously unsent segments", NOT the number of seg=
ments queued

RFC 4653 Section 3.2 Terminating Extended Limited Transmit and Preventing B=
ursts

  "(T.3) A TCP is now permitted to transmit previously unsent data as
         allowed by cwnd, FlightSize, application data availability, and
         the receiver's advertised window."

   =3D> "previously unsent data", NOT segments

RFC 4653 Section 3.3. Extended Limited Transmit

  "(E.2) If the comparison in equation (1), below, holds and there are
         SMSS bytes of previously unsent data available for
         transmission, then the sender MUST transmit one segment of SMSS
         bytes."

   =3D> "previously unsent data", NOT segments

Unless there is a definition of "number of previously unsent segments", I t=
hink this variable has to be defined and it has to be explained how it is c=
alculated.
=20
> As discussed in section 5.3, the number of unsent segments can be
> estimated from the amount of unsent data and the SMSS. Note that we are
> only talking about data that is already available in the send queue,
> and
> later write calls can of course create additional segments. I guess a
> TCP stack that has one SMSS of data in its send queue is free to send
> this as multiple segments if it wants to, but I assume that would not
> be
> the normal behavior. As the estimate is not very critical it is not
> really a problem if this happens occasionally.

If Nagle's algorithm is disabled, sending out multiple segments would be th=
e normal behavior in this case.

I could imagine that disabling Nagle's algorithm could be pretty common amo=
ng thin stream applications. This is why the draft should IMHO discuss if d=
isabling Nagle's algorithm affects the method.

> The addition of "previously unsent segments" is there to capture a
> corner case (see issue raised by Alex in
> http://www.ietf.org/mail-archive/web/tcpm/current/msg08780.html), so it
> will only be occasionally used. If we then get an incorrect estimate,
> say a sender with one SMSS of data would instead send this as two SMSS,
> and the algorithm triggers "incorrectly", this would be equivalent to
> using a RTOR threshold of rrthresh + 1 so nothing critical happens. (In
> some cases rrthresh + 1 may even be better.)

If a wrong parameter for the estimate is possible but not critical, this sh=
ould be explained in the document.

In general, the cited email from Alex stated:

"* Sec 3: IMO the algo/doc is too much Linux driven. I would like to see a
  segment-based *and* byte-based version of the algo, like RFC 5827."

I fully agree to this statement and I think it has not been completely addr=
essed in -04 so far.
=20
> > * Section 9: I wonder if an attacker could try to send certain ACK
> patterns to increase the risk of timeouts e.g. at a Web Server,
> leveraging the smaller RTO value. If successful, the RTOs could
> significantly reduce application performance. Probably, such an
> attacker could cause much more harm by other means, i.e., this may not
> be a significant security risk. But the current security analysis is
> very short.
>=20
> The section is very short, but this is because we do not really see any
> new security issues.
>=20
>   If I understand your example above correctly, the receiver is trying
> to somehow reduce performance for its own application by manipulating
> the ACK pattern. But if you are trying to mess up things for your own
> application you can deliver incorrect data, throw away any data
> received
> or do whatever you want so I do not think trying to possibly slow down
> the sender is the thing to worry about in this attack scenario. If
> someone finds a relevant security issue it should of course be
> documented, but I do not think it makes sense to put in something like
> the above just to make the section longer.

I was thinking e.g. about an off-path attacker that tries to inject ACKs in=
to the TCP connection to trigger RTOs to harm application performance. I am=
 not arguing that this is really a relevant attack scenario, but it seems n=
ot impossible to try that.

Michael





From nobody Mon Nov 10 08:38:45 2014
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Bob Briscoe <bob.briscoe@bt.com>, David Borman <dab@weston.borman.com>
Thread-Topic: [tcpm] I-D Action: draft-borman-tcpm-tcp4way-00.txt
Thread-Index: AQHP76aOj3Voxfk6ZkOwSbwZsagYNpxQILiAgAoEuEA=
Date: Mon, 10 Nov 2014 16:38:30 +0000
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References: <20141024160137.27218.68002.idtracker@ietfa.amsl.com> <3885D312-2D24-4E5D-BE87-A4F877CB64D2@weston.borman.com> <201411032321.sA3NLu9p018412@bagheera.jungle.bt.co.uk>
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Subject: Re: [tcpm] I-D Action: draft-borman-tcpm-tcp4way-00.txt
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Hi,

Just to share some (one of the many test run results) of the actual data co=
llected for port 80:

I tested by sending a <SYN,ECE,CWR,NS> packet; the expected response for no=
n-ECN would be a simple <SYN,ACK>, and for ECN-enabled hosts, a <SYN,ACK,EC=
E>.





Live:  15210 (89.0255)      [targets which actually established a connectio=
n, regular + ECN]

not-ECN: 8465 {55.6542}

SA:        8214 {54.0039} [a regular SYN,ACK, RFC793]

Bad_SAEC:  21 {0.138067}  [the Nonce bit was not reflected, the other ECN b=
its were]

Bad_SAECN: 210 {1.38067}  [all unknown flags are reflected ]

Bad_SAC:   0 {0}          [never observed a host with this combination ]

Bad_SACN:  0 {0}          [never observed a host with this combination ]

Bad_SAEN:  0 {0}          [never observed a host with this combination (RFC=
 3540 Nonce) ]

Bad_SAN:   20 {0.131492}  [ECN flags cleared, but NS bit reflected]



So, there are middleboxes that "understand" the ECN flags ECE and CWR (and =
properly clear them), but still simply reflect the unknown NS flag (SAN).
Other middleboxes (or paths) clear the "unknown" NS flag, but reflect both =
ECE and CWR (SAEC); the majority of broken implementations simply reflect a=
ll unknown flags (SAECN); Finally, SAEN is not strictly a forbidden combina=
tion under ECN Nonce, but we have never seen ECN Nonce support in the wild =
anyway.

Another note to make: On port 80, more broken behavior was seen than on por=
t 21, probably due to a the higher number of paths featuring some kind of w=
eb-enhancing device (load balancer, proxy, ...);

Overall, the use of new flags will have to deal with a population of up to =
2.0% of paths/hosts which won't deal with unknown flags properly.


Richard Scheffenegger


4) Nit: Reflection of Reserved Flags

Simple reflection of Reserved flags cannot be used to confirm that the serv=
er agrees with the client. There are TCP implementations that just reflect =
whatever Reserved flags the client sends, whether or not they support them.

This is fairly easy to work-round. Similar to ECN negotiation [RFC3168, Sec=
 6.1.1.2] you just define the confirmation as a different pattern of flags =
(but this burns more flags).

Richard Scheffenegger was testing the Alexa top 1M Web servers in Apr with =
flags on SYN as follows:
        > CWR ECE SYN
The correct response of ECN-capable servers was:
         <     ECE SYN ACK
I know he found a fair number (sorry I don't have the %) that just dumbly r=
eflected:
        < CWR ECE SYN ACK
And when he set the nonce flag on a SYN sent to these same broken Web serve=
rs, all of them just dumbly reflected back the new pattern:
        > NS CWR ECE SYN
        < NS CWR ECE SYN ACK





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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Just to sh=
are some (one of the many test run results) of the actual data collected fo=
r port 80:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I tested b=
y sending a &lt;SYN,ECE,CWR,NS&gt; packet; the expected response for non-EC=
N would be a simple &lt;SYN,ACK&gt;, and for ECN-enabled hosts, a &lt;SYN,A=
CK,ECE&gt;.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Live:&nbsp; 15210 (89.0255)&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [targets which actually established a connect=
ion, regular &#43; ECN]<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">not-ECN: 8465 {55.6542}<o:p>=
</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">SA:&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; 8214 {54.0039} [a regular SYN,ACK, RFC793]<o:p></o:p></sp=
an></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Bad_SAEC:&nbsp; 21 {0.138067=
}&nbsp; [the Nonce bit was not reflected, the other ECN bits were]<o:p></o:=
p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Bad_SAECN: 210 {1.38067}&nbs=
p; [all unknown flags are reflected ]<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Bad_SAC:&nbsp;&nbsp; 0 {0}&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [never observed a host=
 with this combination ]<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Bad_SACN:&nbsp; 0 {0}&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [never observed a host with=
 this combination ]<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Bad_SAEN:&nbsp; 0 {0}&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [never observed a host with=
 this combination (RFC 3540 Nonce) ]<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Bad_SAN:&nbsp;&nbsp; 20 {0.1=
31492}&nbsp; [ECN flags cleared, but NS bit reflected]<o:p></o:p></span></p=
>
<p class=3D"MsoPlainText"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">So, there =
are middleboxes that &#8220;understand&#8221; the ECN flags ECE and CWR (an=
d properly clear them), but still simply reflect the unknown NS flag (SAN).=
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Other midd=
leboxes (or paths) clear the &#8220;unknown&#8221; NS flag, but reflect bot=
h ECE and CWR (SAEC); the majority of broken implementations simply reflect
 all unknown flags (SAECN); Finally, SAEN is not strictly a forbidden combi=
nation under ECN Nonce, but we have never seen ECN Nonce support in the wil=
d anyway.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Another no=
te to make: On port 80, more broken behavior was seen than on port 21, prob=
ably due to a the higher number of paths featuring some kind
 of web-enhancing device (load balancer, proxy, &#8230;); <o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Overall, t=
he use of new flags will have to deal with a population of up to 2.0% of pa=
ths/hosts which won&#8217;t deal with unknown flags properly.<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Richard Scheffenegger</sp=
an><span style=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;;color:#1F497D"><br>
<br>
</span><o:p></o:p></p>
</div>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<p class=3D"MsoNormal" style=3D"margin-left:5.25pt"><span lang=3D"EN-US"><b=
r>
<b><u>4) Nit: Reflection of Reserved Flags<br>
<br>
</u></b>Simple reflection of Reserved flags cannot be used to confirm that =
the server agrees with the client.
</span>There are TCP implementations that just reflect whatever Reserved fl=
ags the client sends, whether or not they support them.<br>
<br>
This is fairly easy to work-round. Similar to ECN negotiation [RFC3168, Sec=
 6.1.1.2] you just define the confirmation as a different pattern of flags =
(but this burns more flags).<br>
<br>
Richard Scheffenegger was testing the Alexa top 1M Web servers in Apr with =
flags on SYN as follows:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; CWR ECE SYN <br>
The correct response of ECN-capable servers was:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;&nbsp;&nbsp;&nbsp;&nbs=
p; ECE SYN ACK<br>
I know he found a fair number (sorry I don't have the %) that just dumbly r=
eflected:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt; CWR ECE SYN ACK<br>
And when he set the nonce flag on a SYN sent to these same broken Web serve=
rs, all of them just dumbly reflected back the new pattern:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; NS CWR ECE SYN <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt; NS CWR ECE SYN ACK<br>
<br>
<br>
<o:p></o:p></p>
<p><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_356d8ca2226f4567a77e20cab933fbdehioexcmbx05prdhqnetappc_--


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The following errata report has been held for document update 
for RFC5681, "TCP Congestion Control". 

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=5681&eid=4068

--------------------------------------
Status: Held for Document Update
Type: Editorial

Reported by: Michael Taylor <mtaylor@unicoi.com>
Date Reported: 2014-08-04
Held by: Martin Stiemerling (IESG)

Section: GLOBAL

Original Text
-------------


Corrected Text
--------------


Notes
-----
The problem is the use of the phrase "new data" in an imprecise manner to sometimes mean "previously unacknowledged data" and other times mean "never before sent data".

For example, in Section 3.1

   During slow start, a TCP increments cwnd by at most SMSS bytes for
   each ACK received that cumulatively acknowledges new data. 

This should read 

   During slow start, a TCP increments cwnd by at most SMSS bytes for
   each ACK received that cumulatively acknowledges previously 
   unacknowledged data.

I believe that throughout Section 3.1 "new data" refers to "previously unacknowledged data".

However, in Section 3.2 we have

   After the fast retransmit algorithm sends what appears to be the
   missing segment, the "fast recovery" algorithm governs the
   transmission of new data until a non-duplicate ACK arrives.

I believe that here "new data" refers to "previously unsent data".  

This is clearer in the following paragraph from Section 3.2

   1.  On the first and second duplicate ACKs received at a sender, a
       TCP SHOULD send a segment of previously unsent data per [RFC3042]
       provided that the receiver's advertised window allows, the total
       FlightSize would remain less than or equal to cwnd plus 2*SMSS,
       and that new data is available for transmission.

Here we can see the use of "previously unsent data" followed by "new data" 
which refers to the aforementioned "previously unsent data".

While the meaning of "new data" might be clear to those with extensive experience
in TCP it is imprecise and therefore may be quite confusing to those who are learning about the protocol.

--------------------------------------
RFC5681 (draft-ietf-tcpm-rfc2581bis-07)
--------------------------------------
Title               : TCP Congestion Control
Publication Date    : September 2009
Author(s)           : M. Allman, V. Paxson, E. Blanton
Category            : DRAFT STANDARD
Source              : TCP Maintenance and Minor Extensions
Area                : Transport
Stream              : IETF
Verifying Party     : IESG


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--=====================_1463462211==.ALT
Content-Type: text/plain; charset="iso-8859-1"; format=flowed
Content-Transfer-Encoding: quoted-printable

Richard,

Thank you - so useful to have actual data!

However, I'm having trouble understanding a couple of points about the=
 table.

* I assume the first number in each row is the number of responses observed.

* However, I'm not sure what the second numbers=20
are - the ones in () or {}. They seem to add up=20
to a little over 200, so they can't be=20
percentages. The ones in {} are 2.11 times the=20
percentage. The one in () is 1.88 times the percentage}.

* Also, I'm not sure what the difference between=20
the two rows tagged not-ECN and SA is meant to be. Surely SA /is/ not-ECN?



Bob

At 16:38 10/11/2014, Scheffenegger, Richard wrote:
>Hi,
>
>Just to share some (one of the many test run=20
>results) of the actual data collected for port 80:
>
>I tested by sending a <SYN,ECE,CWR,NS> packet;=20
>the expected response for non-ECN would be a=20
>simple <SYN,ACK>, and for ECN-enabled hosts, a <SYN,ACK,ECE>.
>
>
>
>Live:  15210 (89.0255)      [targets which=20
>actually established a connection, regular + ECN]
>not-ECN: 8465 {55.6542}
>SA:        8214 {54.0039} [a regular SYN,ACK, RFC793]
>Bad_SAEC:  21 {0.138067}  [the Nonce bit was not=20
>reflected, the other ECN bits were]
>Bad_SAECN: 210 {1.38067}  [all unknown flags are reflected ]
>Bad_SAC:   0 {0}          [never observed a host with this combination ]
>Bad_SACN:  0 {0}          [never observed a host with this combination ]
>Bad_SAEN:  0 {0}          [never observed a host=20
>with this combination (RFC 3540 Nonce) ]
>Bad_SAN:   20 {0.131492}  [ECN flags cleared, but NS bit reflected]
>
>
>So, there are middleboxes that =93understand=94 the=20
>ECN flags ECE and CWR (and properly clear them),=20
>but still simply reflect the unknown NS flag (SAN).
>Other middleboxes (or paths) clear the =93unknown=94=20
>NS flag, but reflect both ECE and CWR (SAEC);=20
>the majority of broken implementations simply=20
>reflect all unknown flags (SAECN); Finally, SAEN=20
>is not strictly a forbidden combination under=20
>ECN Nonce, but we have never seen ECN Nonce support in the wild anyway.
>
>Another note to make: On port 80, more broken=20
>behavior was seen than on port 21, probably due=20
>to a the higher number of paths featuring some=20
>kind of web-enhancing device (load balancer, proxy, =85);
>
>Overall, the use of new flags will have to deal=20
>with a population of up to 2.0% of paths/hosts=20
>which won=92t deal with unknown flags properly.
>
>
>Richard Scheffenegger
>
>
>4) Nit: Reflection of Reserved Flags
>
>Simple reflection of Reserved flags cannot be=20
>used to confirm that the server agrees with the=20
>client. There are TCP implementations that just=20
>reflect whatever Reserved flags the client=20
>sends, whether or not they support them.
>
>This is fairly easy to work-round. Similar to=20
>ECN negotiation [RFC3168, Sec 6.1.1.2] you just=20
>define the confirmation as a different pattern=20
>of flags (but this burns more flags).
>
>Richard Scheffenegger was testing the Alexa top=20
>1M Web servers in Apr with flags on SYN as follows:
>         > CWR ECE SYN
>The correct response of ECN-capable servers was:
>          <     ECE SYN ACK
>I know he found a fair number (sorry I don't=20
>have the %) that just dumbly reflected:
>         < CWR ECE SYN ACK
>And when he set the nonce flag on a SYN sent to=20
>these same broken Web servers, all of them just=20
>dumbly reflected back the new pattern:
>         > NS CWR ECE SYN
>         < NS CWR ECE SYN ACK
>
>
>

________________________________________________________________
Bob Briscoe,                                                  BT=20
--=====================_1463462211==.ALT
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<body>
Richard,<br><br>
Thank you - so useful to have actual data!<br><br>
However, I'm having trouble understanding a couple of points about the
table.<br><br>
* I assume the first number in each row is the number of responses
observed.<br><br>
* However, I'm not sure what the second numbers are - the ones in () or
{}. They seem to add up to a little over 200, so they can't be
percentages. The ones in {} are 2.11 times the percentage. The one in ()
is 1.88 times the percentage}.<br><br>
* Also, I'm not sure what the difference between the two rows tagged
not-ECN and SA is meant to be. Surely SA /is/ not-ECN?<br><br>
<br><br>
Bob<br><br>
At 16:38 10/11/2014, Scheffenegger, Richard wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">Hi,<br>
&nbsp;<br>
Just to share some (one of the many test run results) of the actual data
collected for port 80:<br>
&nbsp;<br>
I tested by sending a &lt;SYN,ECE,CWR,NS&gt; packet; the expected
response for non-ECN would be a simple &lt;SYN,ACK&gt;, and for
ECN-enabled hosts, a &lt;SYN,ACK,ECE&gt;.<br>
&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
Live:&nbsp; 15210 (89.0255)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [targets which
actually established a connection, regular + ECN]<br>
not-ECN: 8465 {55.6542}<br>
SA:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 8214 {54.0039} [a regular
SYN,ACK, RFC793]<br>
Bad_SAEC:&nbsp; 21 {0.138067}&nbsp; [the Nonce bit was not reflected, the
other ECN bits were]<br>
Bad_SAECN: 210 {1.38067}&nbsp; [all unknown flags are reflected ]<br>
Bad_SAC:&nbsp;&nbsp; 0
{0}&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [never observed
a host with this combination ]<br>
Bad_SACN:&nbsp; 0
{0}&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [never observed
a host with this combination ]<br>
Bad_SAEN:&nbsp; 0
{0}&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [never observed
a host with this combination (RFC 3540 Nonce) ]<br>
Bad_SAN:&nbsp;&nbsp; 20 {0.131492}&nbsp; [ECN flags cleared, but NS bit
reflected]<br>
&nbsp;<br>
&nbsp;<br>
So, there are middleboxes that =93understand=94 the ECN flags ECE and CWR
(and properly clear them), but still simply reflect the unknown NS flag
(SAN).<br>
Other middleboxes (or paths) clear the =93unknown=94 NS flag, but reflect
both ECE and CWR (SAEC); the majority of broken implementations simply
reflect all unknown flags (SAECN); Finally, SAEN is not strictly a
forbidden combination under ECN Nonce, but we have never seen ECN Nonce
support in the wild anyway.<br>
&nbsp;<br>
Another note to make: On port 80, more broken behavior was seen than on
port 21, probably due to a the higher number of paths featuring some kind
of web-enhancing device (load balancer, proxy, =85); <br>
&nbsp;<br>
Overall, the use of new flags will have to deal with a population of up
to 2.0% of paths/hosts which won=92t deal with unknown flags properly.<br>
&nbsp;<br>
&nbsp;<br>
Richard Scheffenegger<br><br>
<br>
<b><u>4) Nit: Reflection of Reserved Flags<br><br>
</u></b>Simple reflection of Reserved flags cannot be used to confirm
that the server agrees with the client. There are TCP implementations
that just reflect whatever Reserved flags the client sends, whether or
not they support them.<br><br>
This is fairly easy to work-round. Similar to ECN negotiation [RFC3168,
Sec 6.1.1.2] you just define the confirmation as a different pattern of
flags (but this burns more flags).<br><br>
Richard Scheffenegger was testing the Alexa top 1M Web servers in Apr
with flags on SYN as follows:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; CWR ECE SYN <br>
The correct response of ECN-capable servers was:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&lt;&nbsp;&nbsp;&nbsp;&nbsp; ECE SYN ACK<br>
I know he found a fair number (sorry I don't have the %) that just dumbly
reflected:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt; CWR ECE SYN ACK<br>
And when he set the nonce flag on a SYN sent to these same broken Web
servers, all of them just dumbly reflected back the new pattern:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; NS CWR ECE SYN <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt; NS CWR ECE SYN
ACK<br><br>
<br>
&nbsp;</blockquote>
<x-sigsep><p></x-sigsep>
________________________________________________________________<br>
Bob
Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
BT</body>
</html>

--=====================_1463462211==.ALT--


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Bob Briscoe <bob.briscoe@bt.com>
Thread-Topic: [tcpm] I-D Action: draft-borman-tcpm-tcp4way-00.txt
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Date: Mon, 10 Nov 2014 18:32:11 +0000
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Hi  Bob,

yes, that list excludes the ECN-enabled sessions.

The ()-bracket behind "Live" is the percentage of responsive IP servers (20=
k IPs were sampled in this instance)
The {}-bracket are the relative percentages of all live hosts, that behaved=
 in a particular way, ie.  54% would be non-ECN, regular SA (not mirroring =
unknown flags),  0.14, 1.38 and 0.13 % would not properly handle unknown TC=
P header flags in some way, and the remainder were ECN-capable TCP stacks. =
Which, by the way is not the same as stating that these sessions (44,35%) w=
ould actually constitute actually working ECN paths - only a small fraction=
 of internet paths which have ECN-capable TCP stacks on both ends deal prop=
erly with the IP-ECN part at this time... I believe something like 2% of pa=
ths (all live hosts) would be supportive of AQM/ECN at an congestion point =
at this time overall - ie. a ECN mark would get through on the IP layer and=
 dealt with properly on the TCP layer.

That was omitted here, as the ECN-sessions were broken down further during =
these tests, ie. if the path would properly handle IP-ECN flags, mirror or =
normalize IP-ECN flags, keep the "TOS" flag of the SYN for the session,  if=
 ECT(1) and CE would elicit proper responses actually etc;

As Brian stated early on during this campaign, there are many more ways of =
brokenness than one would think of...

Richard Scheffenegger




From: Bob Briscoe [mailto:bob.briscoe@bt.com]
Sent: Montag, 10. November 2014 08:13
To: Scheffenegger, Richard
Cc: David Borman; tcpm@ietf.org (tcpm@ietf.org)
Subject: RE: [tcpm] I-D Action: draft-borman-tcpm-tcp4way-00.txt

Richard,

Thank you - so useful to have actual data!

However, I'm having trouble understanding a couple of points about the tabl=
e.

* I assume the first number in each row is the number of responses observed=
.

* However, I'm not sure what the second numbers are - the ones in () or {}.=
 They seem to add up to a little over 200, so they can't be percentages. Th=
e ones in {} are 2.11 times the percentage. The one in () is 1.88 times the=
 percentage}.

* Also, I'm not sure what the difference between the two rows tagged not-EC=
N and SA is meant to be. Surely SA /is/ not-ECN?



Bob

At 16:38 10/11/2014, Scheffenegger, Richard wrote:

Hi,

Just to share some (one of the many test run results) of the actual data co=
llected for port 80:

I tested by sending a <SYN,ECE,CWR,NS> packet; the expected response for no=
n-ECN would be a simple <SYN,ACK>, and for ECN-enabled hosts, a <SYN,ACK,EC=
E>.



Live:  15210 (89.0255)      [targets which actually established a connectio=
n, regular + ECN]
not-ECN: 8465 {55.6542}
SA:        8214 {54.0039} [a regular SYN,ACK, RFC793]
Bad_SAEC:  21 {0.138067}  [the Nonce bit was not reflected, the other ECN b=
its were]
Bad_SAECN: 210 {1.38067}  [all unknown flags are reflected ]
Bad_SAC:   0 {0}          [never observed a host with this combination ]
Bad_SACN:  0 {0}          [never observed a host with this combination ]
Bad_SAEN:  0 {0}          [never observed a host with this combination (RFC=
 3540 Nonce) ]
Bad_SAN:   20 {0.131492}  [ECN flags cleared, but NS bit reflected]


So, there are middleboxes that "understand" the ECN flags ECE and CWR (and =
properly clear them), but still simply reflect the unknown NS flag (SAN).
Other middleboxes (or paths) clear the "unknown" NS flag, but reflect both =
ECE and CWR (SAEC); the majority of broken implementations simply reflect a=
ll unknown flags (SAECN); Finally, SAEN is not strictly a forbidden combina=
tion under ECN Nonce, but we have never seen ECN Nonce support in the wild =
anyway.

Another note to make: On port 80, more broken behavior was seen than on por=
t 21, probably due to a the higher number of paths featuring some kind of w=
eb-enhancing device (load balancer, proxy, ...);

Overall, the use of new flags will have to deal with a population of up to =
2.0% of paths/hosts which won't deal with unknown flags properly.


Richard Scheffenegger


4) Nit: Reflection of Reserved Flags

Simple reflection of Reserved flags cannot be used to confirm that the serv=
er agrees with the client. There are TCP implementations that just reflect =
whatever Reserved flags the client sends, whether or not they support them.

This is fairly easy to work-round. Similar to ECN negotiation [RFC3168, Sec=
 6.1.1.2] you just define the confirmation as a different pattern of flags =
(but this burns more flags).

Richard Scheffenegger was testing the Alexa top 1M Web servers in Apr with =
flags on SYN as follows:
        > CWR ECE SYN
The correct response of ECN-capable servers was:
         <     ECE SYN ACK
I know he found a fair number (sorry I don't have the %) that just dumbly r=
eflected:
        < CWR ECE SYN ACK
And when he set the nonce flag on a SYN sent to these same broken Web serve=
rs, all of them just dumbly reflected back the new pattern:
        > NS CWR ECE SYN
        < NS CWR ECE SYN ACK




________________________________________________________________
Bob Briscoe,                                                  BT

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi&nbsp; B=
ob,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">yes, that =
list excludes the ECN-enabled sessions.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The ()-bra=
cket behind &#8220;Live&#8221; is the percentage of responsive IP servers (=
20k IPs were sampled in this instance)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The {}-bra=
cket are the relative percentages of all live hosts, that behaved in a part=
icular way, ie.&nbsp; 54% would be non-ECN, regular SA (not mirroring
 unknown flags), &nbsp;0.14, 1.38 and 0.13 % would not properly handle unkn=
own TCP header flags in some way, and the remainder were ECN-capable TCP st=
acks. Which, by the way is not the same as stating that these sessions (44,=
35%) would actually constitute actually
 working ECN paths &#8211; only a small fraction of internet paths which ha=
ve ECN-capable TCP stacks on both ends deal properly with the IP-ECN part a=
t this time&#8230; I believe something like 2% of paths (all live hosts) wo=
uld be supportive of AQM/ECN at an congestion
 point at this time overall &#8211; ie. a ECN mark would get through on the=
 IP layer and dealt with properly on the TCP layer.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">That was o=
mitted here, as the ECN-sessions were broken down further during these test=
s, ie. if the path would properly handle IP-ECN flags, mirror
 or normalize IP-ECN flags, keep the &#8220;TOS&#8221; flag of the SYN for =
the session, &nbsp;if ECT(1) and CE would elicit proper responses actually =
etc;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">As Brian s=
tated early on during this campaign, there are many more ways of brokenness=
 than one would think of&#8230;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Richard Scheffenegger</sp=
an><span style=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;;color:#1F497D"><br>
<br>
</span><span style=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quo=
t;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US"><o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><br>
<br>
<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> Bob Briscoe [mailto:bob.briscoe@bt.com]
<br>
<b>Sent:</b> Montag, 10. November 2014 08:13<br>
<b>To:</b> Scheffenegger, Richard<br>
<b>Cc:</b> David Borman; tcpm@ietf.org (tcpm@ietf.org)<br>
<b>Subject:</b> RE: [tcpm] I-D Action: draft-borman-tcpm-tcp4way-00.txt<o:p=
></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Richard,<br>
<br>
Thank you - so useful to have actual data!<br>
<br>
However, I'm having trouble understanding a couple of points about the tabl=
e.<br>
<br>
* I assume the first number in each row is the number of responses observed=
.<br>
<br>
* However, I'm not sure what the second numbers are - the ones in () or {}.=
 They seem to add up to a little over 200, so they can't be percentages. Th=
e ones in {} are 2.11 times the percentage. The one in () is 1.88 times the=
 percentage}.<br>
<br>
* Also, I'm not sure what the difference between the two rows tagged not-EC=
N and SA is meant to be. Surely SA /is/ not-ECN?<br>
<br>
<br>
<br>
Bob<br>
<br>
At 16:38 10/11/2014, Scheffenegger, Richard wrote:<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal">Hi,<br>
&nbsp;<br>
Just to share some (one of the many test run results) of the actual data co=
llected for port 80:<br>
&nbsp;<br>
I tested by sending a &lt;SYN,ECE,CWR,NS&gt; packet; the expected response =
for non-ECN would be a simple &lt;SYN,ACK&gt;, and for ECN-enabled hosts, a=
 &lt;SYN,ACK,ECE&gt;.<br>
&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
Live:&nbsp; 15210 (89.0255)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [targets which ac=
tually established a connection, regular &#43; ECN]<br>
not-ECN: 8465 {55.6542}<br>
SA:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 8214 {54.0039} [a regular SYN=
,ACK, RFC793]<br>
Bad_SAEC:&nbsp; 21 {0.138067}&nbsp; [the Nonce bit was not reflected, the o=
ther ECN bits were]<br>
Bad_SAECN: 210 {1.38067}&nbsp; [all unknown flags are reflected ]<br>
Bad_SAC:&nbsp;&nbsp; 0 {0}&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; [never observed a host with this combination ]<br>
Bad_SACN:&nbsp; 0 {0}&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 [never observed a host with this combination ]<br>
Bad_SAEN:&nbsp; 0 {0}&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 [never observed a host with this combination (RFC 3540 Nonce) ]<br>
Bad_SAN:&nbsp;&nbsp; 20 {0.131492}&nbsp; [ECN flags cleared, but NS bit ref=
lected]<br>
&nbsp;<br>
&nbsp;<br>
So, there are middleboxes that &#8220;understand&#8221; the ECN flags ECE a=
nd CWR (and properly clear them), but still simply reflect the unknown NS f=
lag (SAN).<br>
Other middleboxes (or paths) clear the &#8220;unknown&#8221; NS flag, but r=
eflect both ECE and CWR (SAEC); the majority of broken implementations simp=
ly reflect all unknown flags (SAECN); Finally, SAEN is not strictly a forbi=
dden combination under ECN Nonce, but we have
 never seen ECN Nonce support in the wild anyway.<br>
&nbsp;<br>
Another note to make: On port 80, more broken behavior was seen than on por=
t 21, probably due to a the higher number of paths featuring some kind of w=
eb-enhancing device (load balancer, proxy, &#8230;);
<br>
&nbsp;<br>
Overall, the use of new flags will have to deal with a population of up to =
2.0% of paths/hosts which won&#8217;t deal with unknown flags properly.<br>
&nbsp;<br>
&nbsp;<br>
Richard Scheffenegger<br>
<br>
<br>
<b><u>4) Nit: Reflection of Reserved Flags<br>
<br>
</u></b>Simple reflection of Reserved flags cannot be used to confirm that =
the server agrees with the client. There are TCP implementations that just =
reflect whatever Reserved flags the client sends, whether or not they suppo=
rt them.<br>
<br>
This is fairly easy to work-round. Similar to ECN negotiation [RFC3168, Sec=
 6.1.1.2] you just define the confirmation as a different pattern of flags =
(but this burns more flags).<br>
<br>
Richard Scheffenegger was testing the Alexa top 1M Web servers in Apr with =
flags on SYN as follows:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; CWR ECE SYN <br>
The correct response of ECN-capable servers was:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;&nbsp;&nbsp;&nbsp;&nbs=
p; ECE SYN ACK<br>
I know he found a fair number (sorry I don't have the %) that just dumbly r=
eflected:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt; CWR ECE SYN ACK<br>
And when he set the nonce flag on a SYN sent to these same broken Web serve=
rs, all of them just dumbly reflected back the new pattern:<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; NS CWR ECE SYN <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt; NS CWR ECE SYN ACK<br>
<br>
<br>
&nbsp;<o:p></o:p></p>
<p>________________________________________________________________<br>
Bob Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&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;=
&nbsp; BT<o:p></o:p></p>
</div>
</div>
</body>
</html>

--_000_7cf5e18e16ea4d6186dd6227dab9db5dhioexcmbx05prdhqnetappc_--


From nobody Mon Nov 10 13:24:25 2014
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Alex,

I punted this question to the list, so here's the answer...

See S.4.3. Interaction with the Pre-Existing TCP API
<https://tools.ietf.org/html/draft-briscoe-tcpm-inner-space-01#section-4.3>

Here's the relevant text:
"     Binding to an explicit port:  If an application specifies that it
       wants the TCP client to use a specific port, the Inner Space
       capability MUST be disabled, because the dual handshake has to try
       two ports.  Use of a specific port might be necessary, for example
       in a port-testing application or if the application wants to
       explicitly control all the handshaking logic of the Inner Space
       protocol itself.
"

I realise I need this text is actually too strong. The app doesn't 
need to disable Inner Space. It only needs to disable the dual 
handshake, so that it becomes a serial handshake. I.e. it tries an 
upgraded SYN, and only if the SYN/ACK comes back showing the server 
is legacy does it send a RST. Then it uses the same client port for a 
legacy SYN.

I'll change the text.


Bob


________________________________________________________________
Bob Briscoe,                                                  BT  


From nobody Mon Nov 10 14:34:58 2014
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On 11/9/2014 4:36 PM, Brandon Williams wrote:
> Hi Sujeet,
> 
> I'm still concerned about whether there's enough space in the SYN header
> for this option. My concern is based mostly on a what I think is a flaw
> in the analysis from RFC5925 section 7.2, which leaves out the 4-byte
> MSS option.

Yes, that was an oversight. However, there were 9 SYN option bytes
available with the lengths of the hashes defined at the time, so that's
not a problem.

> As the analysis from RFC6824 indicates, SYN packets
> typically include the MSS option, leaving only 21 bytes available in an
> option packed SYN and 16 in a worst-case unpacked SYN.

Options are not required to be (4-byte) word-aligned; they must *end* on
such a boundary only. If they don't otherwise fit, they should be placed
to fit.

> TCP-AO with SHA1
> works in either case, but SHA2 only works (just barely) in an option
> packed stack.

"barely" means it works. Option word alignment is an optimization that
clearly is inappropriate if the options won't otherwise fit.

> Section 4 of your draft should probably be updated to account for the
> commonality of the MSS option and the related requirement either to drop
> one of the common SYN options or pack the option space.

Pack the options space or (IMO) use a shorter hash. It's not viable for
any one option to declare whether to drop other options.

Joe


From nobody Mon Nov 10 20:32:14 2014
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Joe,

At 19:38 07/11/2014, Joe Touch wrote:
>Hi, Bob,
>
>On 11/7/2014 11:26 AM, Bob Briscoe wrote:
> > Joe,
> >
> > At 00:55 06/11/2014, Joe Touch wrote:
>...
> >> You treat the OOB and SYN as a pair, and treat it with the same time-out
> >> on both ends as far as giving up or retransmitting. How long you keep
> >> queued copies to deal with reordering or other arrival-timing issues is
> >> up to the receiver.
> >
> > Not so IMO. The OOB has to be sent a 'guard time' after the SYN, to
> > minimise the chance of it getting reordered and then getting tangled up
> > in a firewall before the SYN arrives; the firewall will chuck any
> > segment for which it has not seen a prior SYN.
>
>Firewalls aren't the issue. NATs are, and unless reordering is prevalent
>that won't be an issue.

Just because NAT is an issue, doesn't mean a firewall won't be an 
issue as well.


>Reordering would mostly happen on multiple paths, but you'd need
>multiple paths between the client and the NAT - i.e., in the private
>side. That's rare if it happens at all.

Once 2 back-to-back segments have got through a local NAT, they can 
get reordered over an access network (e.g. link aggregation group) 
before arriving at a firewall at the other end.

Also, TCP is used (and might be used in future) in rather more 
circumstances than just where a NAT is in the next room over a small LAN:
- There could be an access network (with link aggregation) then a 
carrier-grade NAT.
- The TCP client might be on a server in a data centre, opening a 
connection to a back-end database in another data centre over a WAN 
with ECMP in between.
- And so on...


Bob



>I thus don't agree that guard times are needed, which means we don't
>need to worry about determining them.
>
> > So although the timeouts might be the same it doesn't seem sensible to
> > start the timers at the same time.
>
>I currently believe that there should be one timer. If you need to
>resend, you resend BOTH the SYN and OOB. The server side can cache to
>deal with reordering that happens there, but that doesn't require
>guard-time at client transmission.
>
>Joe

________________________________________________________________
Bob Briscoe,                                                  BT 


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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: Making TCP robust against reordering - a step forward
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From nobody Tue Nov 11 09:10:13 2014
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From: "Sujeet Nayak A (sua)" <sua@cisco.com>
To: Joe Touch <touch@isi.edu>, Brandon Williams <brandon.williams@akamai.com>,  "tcpm@ietf.org" <tcpm@ietf.org>
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Hi Brandon, Joe,
Thanks a lot for your quick and useful comments.

I will expand section 4 further, with options packing recommendation. If
we cut down on the hash size, that could compromise on the security of
algorithm implementation by increasing the probability of successful
birthday attacks (which could defeat the very purpose of the ask for SHA2).

Regards,

Sujeet



On 11/11/14 4:04 AM, "Joe Touch" <touch@isi.edu> wrote:

>
>
>On 11/9/2014 4:36 PM, Brandon Williams wrote:
>> Hi Sujeet,
>>=20
>> I'm still concerned about whether there's enough space in the SYN header
>> for this option. My concern is based mostly on a what I think is a flaw
>> in the analysis from RFC5925 section 7.2, which leaves out the 4-byte
>> MSS option.
>
>Yes, that was an oversight. However, there were 9 SYN option bytes
>available with the lengths of the hashes defined at the time, so that's
>not a problem.
>
>> As the analysis from RFC6824 indicates, SYN packets
>> typically include the MSS option, leaving only 21 bytes available in an
>> option packed SYN and 16 in a worst-case unpacked SYN.
>
>Options are not required to be (4-byte) word-aligned; they must *end* on
>such a boundary only. If they don't otherwise fit, they should be placed
>to fit.
>
>> TCP-AO with SHA1
>> works in either case, but SHA2 only works (just barely) in an option
>> packed stack.
>
>"barely" means it works. Option word alignment is an optimization that
>clearly is inappropriate if the options won't otherwise fit.
>
>> Section 4 of your draft should probably be updated to account for the
>> commonality of the MSS option and the related requirement either to drop
>> one of the common SYN options or pack the option space.
>
>Pack the options space or (IMO) use a shorter hash. It's not viable for
>any one option to declare whether to drop other options.
>
>Joe
>


From nobody Tue Nov 11 11:07:58 2014
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Subject: [tcpm] Other places for extended TCP options in SYN-only packets
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I=E2=80=99ve been trying to figure out some other place to put extended =
TCP options in a SYN-only packet.  For a standard IPv4/TCP packet, =
because there is only one length field, there is only one place to put =
additional TCP option space that will be ignored by legacy systems: =
encode it into a new IPv4 option.  But I=E2=80=99ll be the first to =
admit this isn=E2=80=99t a great solution:
  1) IPv4 is also limited to just 40 bytes of option space
  2) IPv4 options get looked at by every hop on the path
  3) The presence of IPv4 options would probably cause slow path =
processing in routers.
The slow path processing is mitigated by the fact that this only applies =
to SYN-only packets.  But providing less than 40 additional bytes makes =
this very uninteresting.

If only TCP had an explicit length field, we could put extended option =
space after the TCP data.  And there is a way to do that: encapsulate =
the IPv4/TCP packet, such that the outer IPv4 header had a length that =
included both the inner IPv4/TCP packet, plus the extended options, and =
the inner IPv4/TCP packet length excluded the extended options.  Systems =
that don=E2=80=99t understand the extended options would just see the =
inner IPv4/TCP header, and ignore the trailing data from the outer IPv4 =
packet.  The obvious problem with this is that I wouldn=E2=80=99t expect =
it to work with legacy systems, because it depends on systems already =
having IPv4-in-IPv4 support, which I doubt, since IPv4-in-IPv4 is mainly =
used to create tunnels.

But then I started thinking about IPv6/TCP.  Additional TCP option space =
could be encoded into an IPv6 Destination Options header.  The =
advantages of this over IPv4 are:
  1) There is almost 2K of space; though each option is limited to 255 =
bytes, multiple options could be allowed if that became necessary.
  2) Destination Options headers are not processed at intermediate hops.
  3) With the top 2 bits of the option type set to zero, it will be =
ignored by hosts that don=E2=80=99t understand it.
  4) Hosts that don=E2=80=99t understand it will just see it as a normal =
TCP SYN-only packet.

So for IPv6, it is possible to transmit the expanded TCP option space in =
the same packet in a manner that won't confuse legacy IPv6/TCP =
implementations.

A solution for extended TCP option space in SYN-only packets doesn=E2=80=99=
t need to be the same for IPv4/TCP and IPv6/TCP.  If we can have a clean =
solution for IPv6, we should do that rather than force a less-than =
desireable IPv4 solution on IPv6 connections.

			-David Borman=


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On 11/10/2014 05:34 PM, Joe Touch wrote:
>
>
> On 11/9/2014 4:36 PM, Brandon Williams wrote:
>> Hi Sujeet,
>>
>> I'm still concerned about whether there's enough space in the SYN header
>> for this option. My concern is based mostly on a what I think is a flaw
>> in the analysis from RFC5925 section 7.2, which leaves out the 4-byte
>> MSS option.
>
> Yes, that was an oversight. However, there were 9 SYN option bytes
> available with the lengths of the hashes defined at the time, so that's
> not a problem.
>
>> As the analysis from RFC6824 indicates, SYN packets
>> typically include the MSS option, leaving only 21 bytes available in an
>> option packed SYN and 16 in a worst-case unpacked SYN.
>
> Options are not required to be (4-byte) word-aligned; they must *end* on
> such a boundary only. If they don't otherwise fit, they should be placed
> to fit.

Agreed. But considering how common word alignment was (I don't know 
whether it still _is_ common), it might still be worth pointing out the 
potential compatibility problem.

>
>> TCP-AO with SHA1
>> works in either case, but SHA2 only works (just barely) in an option
>> packed stack.
>
> "barely" means it works. Option word alignment is an optimization that
> clearly is inappropriate if the options won't otherwise fit.
>

"barely" also means that it consumes all of the remaining space. I think 
that's an important consideration.

>> Section 4 of your draft should probably be updated to account for the
>> commonality of the MSS option and the related requirement either to drop
>> one of the common SYN options or pack the option space.
>
> Pack the options space or (IMO) use a shorter hash. It's not viable for
> any one option to declare whether to drop other options.
>

Agreed. Option packing or a shorter hash seem the reasonable choices.

--Brandon

-- 
Brandon Williams; Senior Principal Software Engineer
Emerging Products Engineering; Akamai Technologies Inc.


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David Borman <dab@weston.borman.com> wrote:
> But then I started thinking about IPv6/TCP.  Additional TCP option space could be encoded into an IPv6 Destination Options header.

Yes, but unfortunately ipv6 extension headers do not work.

See e.g.
https://tools.ietf.org/html/draft-gont-v6ops-ipv6-ehs-in-real-world-01

Tables on page 6.


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> On Nov 11, 2014, at 1:41 PM, Florian Westphal <fw@strlen.de> wrote:
>=20
> David Borman <dab@weston.borman.com> wrote:
>> But then I started thinking about IPv6/TCP.  Additional TCP option =
space could be encoded into an IPv6 Destination Options header.
>=20
> Yes, but unfortunately ipv6 extension headers do not work.
>=20
> See e.g.
> https://tools.ietf.org/html/draft-gont-v6ops-ipv6-ehs-in-real-world-01
>=20
> Tables on page 6.

Which doesn=E2=80=99t say that extension headers don=E2=80=99t work, it =
says that for their test, when using a Destination Header with a PadN =
option, roughly 12%-21% of the connections didn=E2=80=99t succeed.

I=E2=80=99m getting tired of the =E2=80=9Cit won=E2=80=99t pass through =
100% of todays middle boxes=E2=80=9D being the first or only thing that =
gets considered.

As I said before:  We need to first decide what we want TCP to look =
like, and then find a way to transition from current implementations to =
the new design, which might include middle boxes having to change, =
rather than letting current middle box implementations dictate how we =
can or can=E2=80=99t change TCP.

		-David Borman


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Subject: Re: [tcpm] Other places for extended TCP options in SYN-only packets
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On 11/11/2014 11:07 AM, David Borman wrote:
> I’ve been trying to figure out some other place to put extended TCP
> options in a SYN-only packet.  For a standard IPv4/TCP packet,
> because there is only one length field, there is only one place to
> put additional TCP option space that will be ignored by legacy
> systems: encode it into a new IPv4 option.  But I’ll be the first to
> admit this isn’t a great solution:
>   1) IPv4 is also limited to just 40 bytes of option space
>   2) IPv4 options get looked at by every hop on the path
>   3) The presence of IPv4 options would probably cause slow path processing in routers.

We've pushed ourselves into a corner by not pushing back on these
optimizations. All our option spaces are getting killed in the name of
profit.

These are the times I wish we had both of the following:
	- an IAB worthy of the name
	- a compliance process

> The slow path processing is mitigated by the fact that this only
> applies to SYN-only packets.  But providing less than 40 additional
> bytes makes this very uninteresting.

+1

AFAICT, the key motivation for larger option space isn't "a little
more", but rather "an order of magnitude more" - e.g., I'd love to have
that so I can use a decent-sized Diffie-Hellman exchange in TCP-AO-encrypt.

> If only TCP had an explicit length field, we could put extended 
> option space after the TCP data.

Agreed. However, the NCP->TCP/IP split design wasn't really a complete
split. TCP isn't completely separate from IP, and the reuse of the
pseudoheader fields from IP in TCP causes us many problems:

	- no length field for TCP
	- forwarding ID == transport connection ID

> And there is a way to do that: encapsulate the IPv4/TCP packet, such
> that the outer IPv4 header had a length that included both the inner
> IPv4/TCP packet, plus the extended options, and the inner IPv4/TCP
> packet length excluded the extended options. 

This is a simpler way to achieve the same result as "Innerspace" - i.e.,
it's going directly towards encapsulation. It's basically TCP-in-TCP as
I suggested.

However, at that point, why do anything so complex? Just run the
simplest, plainest existing features on the outer TCP and use the inner
one to support options that ONLY the endpoint is supposed to know about.

That said, until we're willing to say that some middlebox behavior is
incorrect, we're just pushing off the inevitable - a middlebox that
parses the packets two levels deep, and then messes things up again.

...
> But then I started thinking about IPv6/TCP.  Additional TCP option
> space could be encoded into an IPv6 Destination Options header. 

Same problem. Slow path, and you also have to deal with OPS and SEC folk
who are (IMO) trying to deprecate every "feature" of these protocols
without going through INTAREA. (I've already noted that on both of their
lists).

IMO, a feature that's expensive to deploy is a responsibility - not a
security risk or operational problem. Again, IMO, the companies that
make money on designing Internet products should not be trying to
dumb-down our protocols by this slippery-slope approach; they should
either swim or sink due on the capability of the devices they sell.

However, on that point:
> 1) There is almost 2K of space; though each option is limited to 255
> bytes, multiple options could be allowed if that became necessary.

Maybe, but that was before OPS and SEC got involved. They've been trying
to say that the chain of nested headers should be limited (less work ==
more $$).

>   2) Destination Options headers are not processed at intermediate hops.

TCP isn't supposed to be either, and look where we are with that. So
you'll just be shouting at router vendors who think they can sell
"advanced capabilities" by having their routers process those options
anyway.

> 3) With the top 2 bits of the option type set to zero, it will be
> ignored by hosts that don’t understand it.

As with the "dest opts" values, these too are ignored by router vendors
who want to sell operators "features" they think are useful.

> 4) Hosts that don’t understand it will just see it as a normal TCP
> SYN-only packet.

We like to think that these things are true, but in general, even when a
spec says to ignore such things we already find endpoints that don't. So
we need to validate that first.

> So for IPv6, it is possible to transmit the expanded TCP option
> space in the same packet in a manner that won't confuse legacy IPv6/TCP
> implementations.
> 
> A solution for extended TCP option space in SYN-only packets doesn’t
> need to be the same for IPv4/TCP and IPv6/TCP. If we can have a clean
> solution for IPv6, we should do that rather than force a less-than
> desireable IPv4 solution on IPv6 connections.

In principle, I agree, but I don't think you've found anything but a
different way to have middleboxes mess things up.

Joe


From nobody Wed Nov 12 15:56:13 2014
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Jakob Heitz (jheitz)" <jheitz@cisco.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] FW: New Version Notification for draft-heitzhe-tcpm-vm-rto-00.txt
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Subject: Re: [tcpm] FW: New Version Notification for draft-heitzhe-tcpm-vm-rto-00.txt
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Hi Jakob,

I read your draft; the references section is imho missing a reference to H.=
 Ekstr=F6m, R. Ludwig, "The Peak-Hopper", INFOCOM 2004;=20
http://www.cs.helsinki.fi/u/gurtov/reiner/PeakHopper_draft0207.pdf
as is a detailed discussion why that modification or rfc6298 is not applica=
ble.
=20

A comparison of this new mechanism with the (adjusted) EDWA of RFC6298  or =
other schemes should be given. Also, although the introduction clearly stat=
es that this mechanism is mostly for virtulized environments, a formal appl=
icability statement might be good. Ie. can that algorithm applied safely fo=
r non-VM environments? What is the impact of omitting the RTT variance sign=
al into the RTO calculation?

(And, was the baseline truly RFC6298, or the Linux RTO algorithm, which is =
subtly different and adjust much quicker to lower RTOs once the lower limit=
 is removed)


Technically: the last sentence of section 2 say: "A window of data is the l=
argest ever advertised window of a session." First, I think this definition=
 should be moved to point 2 above; but even then, it's not clear to me if t=
his really is a sensible choice; with many TCPs advertising many 100 kB or =
even MB of window for application-limited flows, the interval (20*advertise=
d window) could be many seconds or even minutes, when the RTO would be adju=
sted down...

In particular, for latency-sensitive, low-bandwidth and application-limited=
 flows, that announce a comparatively large window, this appears to artific=
ially delay the completion time of that kind of transactions. Also, taking =
an spurious RTO, with the changes to the CC state not being unwound by ie. =
Eiffel-reaction, is much less critical in that circumstances than for bulk =
transfers.

The draft is also not clear on its interaction with per-packet RTT measurem=
ent vs. one-per-window (send window, not advertised window) RTT measurement=
.

Best regards,

Richard Scheffenegger


> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Jakob Heitz
> (jheitz)
> Sent: Sonntag, 19. Oktober 2014 14:41
> To: tcpm@ietf.org
> Cc: Chuan He
> Subject: [tcpm] FW: New Version Notification for draft-heitzhe-tcpm-vm-
> rto-00.txt
>=20
> Hi TCPM,
>=20
> We were experimenting with TCP code in a VM setting.
> We reduced the minimum RTO, because average RTT should be much lower than
> 1 second.
> We got lots of spurious retransmissions in a heavily loaded host.
> As a result, we developed an alternative to the exponential weighted
> moving average algorithm to determine the RTO. It operates well without
> any minimum RTO setting.
>=20
> Please take a look and comment.
>=20
> --Jakob
>=20
>=20
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> Sent: Sunday, October 19, 2014 4:52 PM
> To: Jakob Heitz (jheitz); Chuan He; Chuan He; Jakob Heitz (jheitz)
> Subject: New Version Notification for draft-heitzhe-tcpm-vm-rto-00.txt
>=20
>=20
> A new version of I-D, draft-heitzhe-tcpm-vm-rto-00.txt has been
> successfully submitted by Jakob Heitz and posted to the IETF repository.
>=20
> Name:		draft-heitzhe-tcpm-vm-rto
> Revision:	00
> Title:		TCP Retransmission Timer for Virtual Machines
> Document date:	2014-10-19
> Group:		Individual Submission
> Pages:		5
> URL:            http://www.ietf.org/internet-drafts/draft-heitzhe-tcpm-vm=
-
> rto-00.txt
> Status:         https://datatracker.ietf.org/doc/draft-heitzhe-tcpm-vm-
> rto/
> Htmlized:       http://tools.ietf.org/html/draft-heitzhe-tcpm-vm-rto-00
>=20
>=20
> Abstract:
>    A Round Trip Time (RTT) estimate that decays performs badly in a
>    bursty environment. A round trip time estimator that does not decay
>    for a period of time is proposed.
>=20
>    It does not require a minimum value to be configured. It works
>    equally well when the typical RTT is 100uS as when it is 10 seconds.
>=20
>=20
>=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.
>=20
> The IETF Secretariat
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>, Joe Touch <touch@ISI.EDU>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: Comments to draft-ietf-tcpm-tcp-edo-01
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Date: Thu, 13 Nov 2014 01:09:43 +0000
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Date: Wed, 12 Nov 2014 21:20:54 -0800
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Comments to draft-ietf-tcpm-tcp-edo-01
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Hi, Richard,

Thanks for the additional comments. Notes below...

Joe

On 11/12/2014 5:09 PM, Scheffenegger, Richard wrote:
> 
> Another point:
> 
> * Page 4: „A null EDO length option contains the same value as the 
> DO field, i.e., it does not extend the TCP option space“
> 
> "DO field" is not formally mentioned prior to this instance in the
> draft; If the -02 mentiones it in other places, I suggest to introduce
> it somewhere "... the TCP header data offset field (DO field)", or
> alternatively simply refer to it in full words.

Will fix (in the first sentence of the Intro).

> "When the EDO length is invalid, the TCP segment MUST be silently
> dropped"
> 
> When is EDO length invalid? Is that meant to refer to the length 
> field of the EDO option (4), or the "edo length option" as a whole,
> ie. when the Header_length is < DO, (inclusive) or >
> Segment_Length?

There are two different issues for validity, as you point out:

	a well-formed EDO option

	a meaningful EDO length field value

I'll revise and clarify accordingly.

---


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From: Olivier Bonaventure <bonaventure@ieee.org>
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Subject: Re: [tcpm] Other places for extended TCP options in SYN-only packets
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David,


> But then I started thinking about IPv6/TCP.  Additional TCP option space
> could be encoded into an IPv6 Destination Options header.  The advantages
> of this over IPv4 are:
>   1) There is almost 2K of space; though each option is limited to 255
> bytes, multiple options could be allowed if that became necessary.
>   2) Destination Options headers are not processed at intermediate hops.
>   3) With the top 2 bits of the option type set to zero, it will be
> ignored by hosts that don=E2=80=99t understand it.
>   4) Hosts that don=E2=80=99t understand it will just see it as a normal =
TCP
> SYN-only packet.
>
> So for IPv6, it is possible to transmit the expanded TCP option space in
> the same packet in a manner that won't confuse legacy IPv6/TCP
> implementations
>

I think that this is the cleanest possible solution with regular TCP. For
MPTCP, we adopt a similar approach as the control stream draft to encode
options in the payload
https://tools.ietf.org/html/draft-paasch-mptcp-control-stream-00


> A solution for extended TCP option space in SYN-only packets doesn=E2=80=
=99t need
> to be the same for IPv4/TCP and IPv6/TCP.  If we can have a clean solutio=
n
> for IPv6, we should do that rather than force a less-than desireable IPv4
> solution on IPv6 connections.
>

I completely agree. If we can solve the issue for IPv6, we are future
proof. We cannot live forever with all the IPv4 problems and IPv6 is being
widely deployed


Olivier

--f46d043894810223de0507c27943
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<div dir=3D"ltr">David,<br><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-sty=
le:solid;padding-left:1ex"><br>
But then I started thinking about IPv6/TCP.=C2=A0 Additional TCP option spa=
ce could be encoded into an IPv6 Destination Options header.=C2=A0 The adva=
ntages of this over IPv4 are:<br>
=C2=A0 1) There is almost 2K of space; though each option is limited to 255=
 bytes, multiple options could be allowed if that became necessary.<br>
=C2=A0 2) Destination Options headers are not processed at intermediate hop=
s.<br>
=C2=A0 3) With the top 2 bits of the option type set to zero, it will be ig=
nored by hosts that don=E2=80=99t understand it.<br>
=C2=A0 4) Hosts that don=E2=80=99t understand it will just see it as a norm=
al TCP SYN-only packet.<br>
<br>
So for IPv6, it is possible to transmit the expanded TCP option space in th=
e same packet in a manner that won&#39;t confuse legacy IPv6/TCP implementa=
tions<br></blockquote><div><br></div><div>I think that this is the cleanest=
 possible solution with regular TCP. For MPTCP, we adopt a similar approach=
 as the control stream draft to encode options in the payload</div><div><a =
href=3D"https://tools.ietf.org/html/draft-paasch-mptcp-control-stream-00">h=
ttps://tools.ietf.org/html/draft-paasch-mptcp-control-stream-00</a><br></di=
v><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border=
-left-style:solid;padding-left:1ex">
A solution for extended TCP option space in SYN-only packets doesn=E2=80=99=
t need to be the same for IPv4/TCP and IPv6/TCP.=C2=A0 If we can have a cle=
an solution for IPv6, we should do that rather than force a less-than desir=
eable IPv4 solution on IPv6 connections.<br></blockquote><div><br></div><di=
v>I completely agree. If we can solve the issue for IPv6, we are future pro=
of. We cannot live forever with all the IPv4 problems and IPv6 is being wid=
ely deployed</div><div><br></div><div><br></div><div>Olivier</div></div></d=
iv></div>

--f46d043894810223de0507c27943--


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Hi,

Draft TCPM meeting minutes from Monday are available at =
http://www.ietf.org/proceedings/91/minutes/minutes-91-tcpm . Please send =
any corrections to tcpm-chairs.

Many thanks to Mirja and Richard for taking notes!

- Pasi


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From: "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>
To: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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From nobody Fri Nov 14 14:47:47 2014
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Thanks, fixed in my local copy. I will upload a new version after a =
couple of days, giving a bit time for other possible comments.

- Pasi


On 13 Nov 2014, at 20:15, Zimmermann, Alexander =
<Alexander.Zimmermann@netapp.com> wrote:

> in the Cubic Section: high start =97> HyStart
>=20
>> Am 13.11.2014 um 18:25 schrieb Pasi Sarolahti =
<pasi.sarolahti@iki.fi>:
>>=20
>> Hi,
>>=20
>> Draft TCPM meeting minutes from Monday are available at =
http://www.ietf.org/proceedings/91/minutes/minutes-91-tcpm . Please send =
any corrections to tcpm-chairs.
>>=20
>> Many thanks to Mirja and Richard for taking notes!
>>=20
>> - Pasi
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20


From nobody Sun Nov 16 12:16:03 2014
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Date: Sun, 16 Nov 2014 19:24:20 +0000
To: Ted Hardie <ted.ietf@gmail.com>
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Subject: [tcpm] Inner Space and MTU
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Ted,

Thx for the point you raised about when I presented Inner Space in 
tcpinc. As Martin Stiemerling requested, some issues were not 
specific to tcpinc, so I am moving them to tcpm. When resolved, I 
will send a digest to tcpinc.

Regarding the MTU issue you raised, I don't think it requires a 
redesign, but it is a valid concern. Therefore, I propose to add the 
following to the section "Interaction with the Pre-Existing TCP API"

"Some applications interrogate the TCP stack to determine the path 
max transmission unit (PMTU), e.g. in order to optimize application 
message boundaries within the datastream. From the viewpoint of such 
applications, TCP options subtract the same amount from the PMTU 
whether they are Outer or Inner Options. However, the 8 (or 12) octet 
InSpace header and the alignment padding represent extra overhead. 
Therefore, for such applications, the TCP stack as seem through the 
socket API will seem similar to a tunnel that reduces the useful size 
of the PMTU. This could lead to fragmentation until such applications 
are updated. Nonetheless, most such applications already include code 
to adapt to PMTU reduction by tunnels."

Is this an acceptable resolution?


Bob



________________________________________________________________
Bob Briscoe,                                                  BT 


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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 17 Nov 2014 12:30:58 +0800
Message-ID: <CAK6E8=f0E13-e=nAvP-4RJx3EQmAG3YvuLHPK6Y01zkmNpALfQ@mail.gmail.com>
To: Bob Briscoe <bob.briscoe@bt.com>
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Cc: Ted Hardie <ted.ietf@gmail.com>, tcpm IETF list <tcpm@ietf.org>
Subject: Re: [tcpm] Inner Space and MTU
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On Mon, Nov 17, 2014 at 3:24 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> Ted,
>
> Thx for the point you raised about when I presented Inner Space in tcpinc.
> As Martin Stiemerling requested, some issues were not specific to tcpinc, so
> I am moving them to tcpm. When resolved, I will send a digest to tcpinc.
>
> Regarding the MTU issue you raised, I don't think it requires a redesign,
> but it is a valid concern. Therefore, I propose to add the following to the
> section "Interaction with the Pre-Existing TCP API"
>
> "Some applications interrogate the TCP stack to determine the path max
> transmission unit (PMTU), e.g. in order to optimize application message
> boundaries within the datastream. From the viewpoint of such applications,
> TCP options subtract the same amount from the PMTU whether they are Outer or
> Inner Options. However, the 8 (or 12) octet InSpace header and the alignment
> padding represent extra overhead. Therefore, for such applications, the TCP
> stack as seem through the socket API will seem similar to a tunnel that
> reduces the useful size of the PMTU. This could lead to fragmentation until
> such applications are updated. Nonetheless, most such applications already
> include code to adapt to PMTU reduction by tunnels.
Reasonable but I'd remove the last sentence.

IMO it's app's own risk doing these kind of optimizations.
The particular fragmentation issue has motivated the Linux auto-corking feature:
http://lwn.net/Articles/576263/

However an interesting question is for implementation that exports the
MSS to the application (e.g., Linux TCP info tcpi*mss fields): should
it include or exclude the inner-space option?

>
> Is this an acceptable resolution?
>
>
> Bob
>
>
>
> ________________________________________________________________
> Bob Briscoe,                                                  BT
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: John Leslie <john@jlc.net>
Thread-Topic: [tcpm] another basic option question - can TCP enable options on the SYN/ACK?
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Subject: Re: [tcpm] another basic option question - can TCP enable options on the SYN/ACK?
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Return-Path: <ycheng@google.com>
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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 17 Nov 2014 14:48:11 +0800
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To: "Scheffenegger, Richard" <rs@netapp.com>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] another basic option question - can TCP enable options on the SYN/ACK?
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On Mon, Nov 17, 2014 at 2:11 PM, Scheffenegger, Richard <rs@netapp.com> wrote:
> Hi John,
>
> Without any claim of completness or recentness:
>
> liunx (at least used to) start talking / reflecting timestamps when they
> start appearing in mid-session. Simultaneously, a connection will not fail
> when timestamps suddenly disappear (which is, at least in 7323, in violation
> of PAWS. But i did some more extensive tests, which also pointed in the
> direction that paws is actually not active in linux, and the per-segment rtt
> sampling is done using per-segment state in the server. So in conclusion,
> timestamps appear to serve no purpose on linux...
?!

Linux TCP uses RFC1323 PAWS by default if TS is used. RFC7323 is just
published so please don't expect implementation will follow
immediately ok? please bring the specific tests you have to linux
netdev list or send them to me.

Linux also uses TS for RTT measurement if Karn's rule forbids taking
RTTs. TS is also critical for Eifel (undo) operations.

Both have been used for years.

>
> Sorry tried to prune the original
> Mail to the part i respond to, but this client cut that part out instead
> (iphone)
>
> Best regards,
>    Richard Scheffenegger
>
> Am 24.10.2014 um 04:18 schrieb "John Leslie" <john@jlc.net>:
>
> Joe Touch <touch@isi.edu> wrote:
>
> On 10/21/2014 8:54 PM, Yoshifumi Nishida wrote:
>
>
> I'm wondering if all negotiations need to be negotiated.
>
>
> The ones specified in RFC793 do not.
>
>
>   (Of course!)
>
> All others need to be negotiated *if* you care that the other end
>
> supports it.
>
>
>   Quite true. But note that it's easy to imagine options where one
> endpoint _doesn't_ care if the other endpoint supports it.
>
>   In fact, I expect to shortly submit a draft where this is the case.
> An endpoint sending it announces its own support for it; but unless
> the other endpoint also sends it, the first endpoint won't know whether
> it has been acted on in any way.
>
> That negotiation can happen at any point in the connection,
>
>
>   Currently, most options _are_ noted in the originating SYN, for
> several different reasons. First, the SYN and SYN-ACK are the only
> place in a TCP connection that you know the options won't be quietly
> dropped due to congestion. I am not aware
>
> of much evidence either way on that.)
>
> but if it doesn't happen during the opening handshake then the option
>
> needs to be specified with its own 3WHS mechanism that can operate
>
> independently of segment loss and retransmission (which, IMO, is a bad
>
> idea).
>
>
>   I don't agree it's a "bad" idea, but it surely is more complicated.
>
>   Please note, however, that a 3-way HandShake isn't the only way to
> synchronize sender's and receiver's views of an option being enabled.
> (It is usually the best, but not the only...)
>
> Some are confirmed in both directions (such as TS and WS), but others
>
> may need to have a "response" to a specific "request".
>
>
>   That indeed is common; and it becomes complicated because either the
> request or the response can get dropped.
>
> Finally, there are options that aren't as much negotiated as they are
>
> either supported or not, e.g., MD5 and AO. There's no handshake. If the
>
> option is required on that connection and isn't present, segments are
>
> ignored anyway.
>
>
>   True. And, IMHO, this is a rather important case.
>
> I.e., a taxonomy should include, at a minimum:
>
>
>    - always supported
>
>        end-of-list
>
>        NOP
>
>        MSS
>
>
>    - supported if enabled in both directions during the 3WHS
>
>        TS, WS
>
>
>    - request/response
>
>        TFO, MPTCP, SACK
>
>
>    - supported if enabled throughout the connection
>
>        MD5, AO
>
>
>   I'm hesitant to make statements like this: Each option should be
> defined to detail its rules; and we can never hope to cover the entire
> range of possible rules.
>
>   Nonetheless, Joe is correct, so far as this goes.
>
> --
> John Leslie <john@jlc.net>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


From nobody Mon Nov 17 05:46:29 2014
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Yuchung Cheng <ycheng@google.com>, Bob Briscoe <bob.briscoe@bt.com>
Thread-Topic: [tcpm] Inner Space and MTU
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Subject: Re: [tcpm] Inner Space and MTU
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> However an interesting question is for implementation that exports the
> MSS to the application (e.g., Linux TCP info tcpi*mss fields): should
> it include or exclude the inner-space option?

Doesn't the same question apply to all variables that refer to byte stream =
characteristics in bytes? For instance, tcpi_*_ssthresh? (Which is less lik=
ely to be used by an app, but one never knows...)

Another interesting question is what a stack with cwnd in bytes does? Does =
the value include or exclude the inner-space options? What happens if the v=
alue is exposed to an app?

Possibly, other shim layer protocols on top of TCP run into similar questio=
ns.

Michael

=20


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Subject: Re: [tcpm] Inner Space and MTU
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On 11/16/2014 8:30 PM, Yuchung Cheng wrote:
> However an interesting question is for implementation that exports the
> MSS to the application (e.g., Linux TCP info tcpi*mss fields): should
> it include or exclude the inner-space option?

First, the MSS value used in the header does NOT include TCP or IP
options as per rfc6691

However, note that MSS has meaning at the TCP segment level

The user-level shouldn't be involved in this. Sec 4 of that RFC explains
why - the space used by options can vary (e.g., SACK), so there is no
one appropriate MSS to report to the application.

To the user, TCP is a bytestream. Wanting to expose segment boundaries
to the user (or even implementing it in Linux) doesn't make it possible.

Joe


From nobody Tue Nov 18 09:04:21 2014
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--089e0111d1de73c0fe050812ad83
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Hi Bob,

Some thougths in-line.

On Sun, Nov 16, 2014 at 8:30 PM, Yuchung Cheng <ycheng@google.com> wrote:

> On Mon, Nov 17, 2014 at 3:24 AM, Bob Briscoe <bob.briscoe@bt.com> wrote:
> > Ted,
> >
> > Thx for the point you raised about when I presented Inner Space in
> tcpinc.
> > As Martin Stiemerling requested, some issues were not specific to
> tcpinc, so
> > I am moving them to tcpm. When resolved, I will send a digest to tcpinc=
.
> >
> > Regarding the MTU issue you raised, I don't think it requires a redesig=
n,
> > but it is a valid concern. Therefore, I propose to add the following to
> the
> > section "Interaction with the Pre-Existing TCP API"
> >
> > "Some applications interrogate the TCP stack to determine the path max
> > transmission unit (PMTU), e.g. in order to optimize application message
> > boundaries within the datastream. From the viewpoint of such
> applications,
> > TCP options subtract the same amount from the PMTU whether they are
> Outer or
> > Inner Options. However, the 8 (or 12) octet InSpace header and the
> alignment
> > padding represent extra overhead. Therefore, for such applications, the
> TCP
> > stack as seem through the socket API will seem similar to a tunnel that
> > reduces the useful size of the PMTU. This could lead to fragmentation
> until
> > such applications are updated.

=E2=80=8B
So, the best thing to do from my perspective would be to split IP_MTU
into TCP_MTU and UDP_MTU in getsockopt(); if inner headers would be in
use for TCP, then the MTU returned by the new call would be lower than
that returned for UDP_MTU.  If not, they would be the same.  (You'd need to
figure out what to do if IP_MTU were used, but an obvious choice here
would be to return the lower of the two values if they differ.)

In advance of something like that emerging, the conservative thing to do
would be for the application to do in an Inner Space-approved world would
be to assume it's always in use, and to adjust its message boundaries on
that
basis.

regards,

Ted




> Nonetheless, most such applications already
> > include code to adapt to PMTU reduction by tunnels.
> Reasonable but I'd remove the last sentence.
>
> IMO it's app's own risk doing these kind of optimizations.
> The particular fragmentation issue has motivated the Linux auto-corking
> feature:
> http://lwn.net/Articles/576263/
>
> However an interesting question is for implementation that exports the
> MSS to the application (e.g., Linux TCP info tcpi*mss fields): should
> it include or exclude the inner-space option?
>
> >
> > Is this an acceptable resolution?
> >
> >
> > Bob
> >
> >
> >
> > ________________________________________________________________
> > Bob Briscoe,                                                  BT
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
>

--089e0111d1de73c0fe050812ad83
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">Hi Bob,<br><br>Some thougths in-line.<br></div><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, Nov 16, 20=
14 at 8:30 PM, Yuchung Cheng <span dir=3D"ltr">&lt;<a href=3D"mailto:ycheng=
@google.com" target=3D"_blank">ycheng@google.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft:1px solid rgb(204,204,204);padding-left:1ex"><span class=3D"">On Mon, N=
ov 17, 2014 at 3:24 AM, Bob Briscoe &lt;<a href=3D"mailto:bob.briscoe@bt.co=
m">bob.briscoe@bt.com</a>&gt; wrote:<br>
&gt; Ted,<br>
&gt;<br>
&gt; Thx for the point you raised about when I presented Inner Space in tcp=
inc.<br>
&gt; As Martin Stiemerling requested, some issues were not specific to tcpi=
nc, so<br>
&gt; I am moving them to tcpm. When resolved, I will send a digest to tcpin=
c.<br>
&gt;<br>
&gt; Regarding the MTU issue you raised, I don&#39;t think it requires a re=
design,<br>
&gt; but it is a valid concern. Therefore, I propose to add the following t=
o the<br>
&gt; section &quot;Interaction with the Pre-Existing TCP API&quot;<br>
&gt;<br>
&gt; &quot;Some applications interrogate the TCP stack to determine the pat=
h max<br>
&gt; transmission unit (PMTU), e.g. in order to optimize application messag=
e<br>
&gt; boundaries within the datastream. From the viewpoint of such applicati=
ons,<br>
&gt; TCP options subtract the same amount from the PMTU whether they are Ou=
ter or<br>
&gt; Inner Options. However, the 8 (or 12) octet InSpace header and the ali=
gnment<br>
&gt; padding represent extra overhead. Therefore, for such applications, th=
e TCP<br>
&gt; stack as seem through the socket API will seem similar to a tunnel tha=
t<br>
&gt; reduces the useful size of the PMTU. This could lead to fragmentation =
until<br>
&gt; such applications are updated.</span></blockquote><div><div class=3D"g=
mail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">=E2=
=80=8B<br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,sa=
ns-serif;font-size:small">So, the best thing to do from my perspective woul=
d be to split IP_MTU<br>into TCP_MTU and UDP_MTU in getsockopt(); if inner =
headers would be in<br>use for TCP, then the MTU returned by the new call w=
ould be lower than <br>that returned for UDP_MTU.=C2=A0 If not, they would =
be the same.=C2=A0 (You&#39;d need to <br>figure out what to do if IP_MTU w=
ere used, but an obvious choice here <br>would be to return the lower of th=
e two values if they differ.)<br><br></div><div class=3D"gmail_default" sty=
le=3D"font-family:tahoma,sans-serif;font-size:small">In advance of somethin=
g like that emerging, the conservative thing to do<br>would be for the appl=
ication to do in an Inner Space-approved world would<br>be to assume it&#39=
;s always in use, and to adjust its message boundaries on that<br>basis.<br=
><br>regards,<br><br>Ted<br></div><div class=3D"gmail_default" style=3D"fon=
t-family:tahoma,sans-serif;font-size:small"><pre><br></pre></div>=C2=A0</di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left:1px solid rgb(204,204,204);padding-left:1ex"><span class=3D""> Nonet=
heless, most such applications already<br>
&gt; include code to adapt to PMTU reduction by tunnels.<br>
</span>Reasonable but I&#39;d remove the last sentence.<br>
<br>
IMO it&#39;s app&#39;s own risk doing these kind of optimizations.<br>
The particular fragmentation issue has motivated the Linux auto-corking fea=
ture:<br>
<a href=3D"http://lwn.net/Articles/576263/" target=3D"_blank">http://lwn.ne=
t/Articles/576263/</a><br>
<br>
However an interesting question is for implementation that exports the<br>
MSS to the application (e.g., Linux TCP info tcpi*mss fields): should<br>
it include or exclude the inner-space option?<br>
<span class=3D""><br>
&gt;<br>
&gt; Is this an acceptable resolution?<br>
&gt;<br>
&gt;<br>
&gt; Bob<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; ________________________________________________________________<br>
&gt; Bob Briscoe,=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 BT<br>
</span>&gt; _______________________________________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote></div><br></div></div>

--089e0111d1de73c0fe050812ad83--


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To: tcpm IETF list <tcpm@ietf.org>
Thread-Topic: [ipr-announce] IPR Disclosure: Telefonaktiebolaget LM Ericsson (publ)'s Statement about IPR related to draft-heitzhe-tcpm-vm-rto-00
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FYI

-----Original Message-----
From: ipr-announce [mailto:ipr-announce-bounces@ietf.org] On Behalf Of IETF=
 Secretariat
Sent: Tuesday, November 18, 2014 4:38 PM
To: jheitz@cisco.com; chuan.he@ericsson.com
Cc: jari.arkko@piuha.net; ipr-announce@ietf.org
Subject: [ipr-announce] IPR Disclosure: Telefonaktiebolaget LM Ericsson (pu=
bl)'s Statement about IPR related to draft-heitzhe-tcpm-vm-rto-00


Dear Jakob Heitz, Chuan He:

 An IPR disclosure that pertains to your Internet-Draft entitled "TCP
Retransmission Timer for Virtual Machines" (draft-heitzhe-tcpm-vm-rto) was
submitted to the IETF Secretariat on 2014-11-12 and has been posted on the =
"IETF
Page of Intellectual Property Rights Disclosures"
(https://datatracker.ietf.org/ipr/2485/). The title of the IPR disclosure i=
s
"Telefonaktiebolaget LM Ericsson (publ)'s Statement about IPR related to dr=
aft-
heitzhe-tcpm-vm-rto-00."");

The IETF Secretariat

_______________________________________________
ipr-announce mailing list
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Subject: Re: [tcpm] Inner Space and MTU
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On 11/17/2014 11:06 AM, Ted Hardie wrote:
> Hi Bob,
> 
> Some thougths in-line.
...
> So, the best thing to do from my perspective would be to split IP_MTU
> into TCP_MTU and UDP_MTU in getsockopt();

TCP might report an MTU for diagnostics, but it's a byte-oriented
protocol so this value shouldn't be part of what's exposed to the
userland application per se.

It only gives the incorrect impression that socket boundaries are
correlated to TCP segment boundaries.

Additionally, as I've already noted, TCP MSS doesn't take into account
the option space needed by either IP or TCP. TCP is in a position to use
that value to limit how much application data is added to those headers
AFTER they're formed.

At the application API, there's no meaningful TCP MTU. The space that
remains in a segment depends on the size of the options used *for that
segment*, and those option sizes already vary.

Joe


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On 11/18/2014 11:48 AM, Ted Hardie wrote:
> Hi Joe,
> 
> On Tue, Nov 18, 2014 at 11:24 AM, Joe Touch <touch@isi.edu
> <mailto:touch@isi.edu>> wrote:
> 
> 
> 
>     On 11/17/2014 11:06 AM, Ted Hardie wrote:
>     > Hi Bob,
>     >
>     > Some thougths in-line.
>     ...
>     > So, the best thing to do from my perspective would be to split IP_MTU
>     > into TCP_MTU and UDP_MTU in getsockopt();
> 
>     TCP might report an MTU for diagnostics, but it's a byte-oriented
>     protocol so this value shouldn't be part of what's exposed to the
>     userland application per se.
> 
> 
> ​At the moment, you can query the IP_MTU, and some TCP-using applications
> attempt to use that information to craft messages that fit within the MTU. 

I can play PowerBall too, but that doesn't mean I'll be right (or that
the app will either).

IMO, an app that tries to use MTU when interfacing is doing a lot of
work that TCP basically ignores in at least two ways:

	1) the options, which (as noted) aren't considered and can
	vary (and the user can't predict - users don't know how
	the might be aligned, even if they're stable)

	2) TCP grabs what it needs to fill the next segment
	based on what's available in the window. That window
	need not always indicate a complete MSS; its position
	depends on what has been ACKd of what has been sent

Even with Nagle disabled, there's no way to control that interaction. So
a TCP MTU just helps the app operate under a delusion.

TCP is most efficient when the app hands over data as soon as it can.
Anything else is asking for interference, not potentially helping.

> The IP_MTU may well be a foot gun in its common usage, but it does get
> used. 

That makes more sense for raw socket apps and UDP, but makes no sense at
all for TCP for the same reason.

> If those applications could query the socket to discover what the maximum
> effective TCP payload size was, I believe they would use it instead. 

Sure. I might use it to guess a PowerBall ticket too. That doesn't mean
it's actually useful or will ever be.

I agree that apps can, will, and already do think they can do better.
They're wrong. Let's not play into that delusion.

Joe


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On 11/18/2014 12:51 PM, Ted Hardie wrote:
>     TCP is most efficient when the app hands over data as soon as it can.
>     Anything else is asking for interference, not potentially helping.
> 
> 
> ​Applications aren't interested in TCP efficiency in isolation; they
> ​are interested in it in an application context.  Sometimes that
> causes them to make choices different from "hand over the data
> as soon as you can in whatever chunks you have".

When they do, they ought to be using a different transport. It isn't
useful for an app to game its interface to TCP.

...
>     > If those applications could query the socket to discover what the maximum
>     > effective TCP payload size was, I believe they would use it instead.
> 
>     Sure. I might use it to guess a PowerBall ticket too. That doesn't mean
>     it's actually useful or will ever be.
> 
>     I agree that apps can, will, and already do think they can do better.
>     They're wrong. Let's not play into that delusion.
> 
> ​Do you oppose Bob's language as well?

His first sentence, IMO, only serves to support the delusion that apps
can optimize by managing their interaction with the TCP stack.

However, I'm not too interested in wordsmithing when there are other,
(IMO) more fundamental problems with his approach.

Joe


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Hi,

The TCPM meeting last week supported adoption of =
draft-zimmermann-tcpm-undeployed-01 (Moving Undeployed TCP Extensions to =
Historic and Informational Status). This mail is to check that people =
who did not attend the meeting are content with the meeting consensus to =
adopt the draft. Therefore, please let us know by the end of this week, =
if you are against adoption -- preferably on the mailing list.

Thanks!

- Pasi


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From: Ted Hardie <ted.ietf@gmail.com>
To: Joe Touch <touch@isi.edu>
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--001a113e986426bcda0508276215
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Hi Joe,

On Tue, Nov 18, 2014 at 11:24 AM, Joe Touch <touch@isi.edu> wrote:

>
>
> On 11/17/2014 11:06 AM, Ted Hardie wrote:
> > Hi Bob,
> >
> > Some thougths in-line.
> ...
> > So, the best thing to do from my perspective would be to split IP_MTU
> > into TCP_MTU and UDP_MTU in getsockopt();
>
> TCP might report an MTU for diagnostics, but it's a byte-oriented
> protocol so this value shouldn't be part of what's exposed to the
> userland application per se.
>
>
=E2=80=8BAt the moment, you can query the IP_MTU, and some TCP-using applic=
ations
attempt to use that information to craft messages that fit within the MTU.
The IP_MTU may well be a foot gun in its common usage, but it does get
used.

If those applications could query the socket to discover what the maximum
effective TCP payload size was, I believe they would use it instead.  That
might
get reported by taking pessimistic estimates of the variable bits plus the
results
of a path mtu discovery.  How it gets reported may be more important than
the details
of how the stack estimates it, though.

Failing that, those applications will try to discern whether inner space
headers
are in use and will do their own estimates of the impact on available
space.  Until
they have been updated to discern that, the 8 (or 12) octet InSpace header
and the
alignment padding will blow them up, as Bob's text notes.  If the working
group wants
to just note the issue and move on, then something along his text is likely
fine.  If
it wants to suggest more, I still believe recognizing the reality and
providing something
better than the current state is worth considering.

Just my two cents,

Ted





> It only gives the incorrect impression that socket boundaries are
> correlated to TCP segment boundaries.
>
> Additionally, as I've already noted, TCP MSS doesn't take into account
> the option space needed by either IP or TCP. TCP is in a position to use
> that value to limit how much application data is added to those headers
> AFTER they're formed.
>
> At the application API, there's no meaningful TCP MTU. The space that
> remains in a segment depends on the size of the options used *for that
> segment*, and those option sizes already vary.
>
> Joe
>

--001a113e986426bcda0508276215
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">Hi Joe,<br></div><div class=3D"gmail_extra"><br>=
<div class=3D"gmail_quote">On Tue, Nov 18, 2014 at 11:24 AM, Joe Touch <spa=
n 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" style=3D=
"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-le=
ft:1ex"><span class=3D""><br>
<br>
On 11/17/2014 11:06 AM, Ted Hardie wrote:<br>
&gt; Hi Bob,<br>
&gt;<br>
&gt; Some thougths in-line.<br>
</span>...<br>
<span class=3D"">&gt; So, the best thing to do from my perspective would be=
 to split IP_MTU<br>
&gt; into TCP_MTU and UDP_MTU in getsockopt();<br>
<br>
</span>TCP might report an MTU for diagnostics, but it&#39;s a byte-oriente=
d<br>
protocol so this value shouldn&#39;t be part of what&#39;s exposed to the<b=
r>
userland application per se.<br>
<br></blockquote><div><br><div class=3D"gmail_default" style=3D"font-family=
:tahoma,sans-serif;font-size:small">=E2=80=8BAt the moment, you can query t=
he IP_MTU, and some TCP-using applications<br></div><div class=3D"gmail_def=
ault" style=3D"font-family:tahoma,sans-serif;font-size:small">attempt to us=
e that information to craft messages that fit within the MTU.=C2=A0 <br>The=
 IP_MTU may well be a foot gun in its common usage, but it does get<br>used=
.=C2=A0 <br><br>If those applications could query the socket to discover wh=
at the maximum <br>effective TCP payload size was, I believe they would use=
 it instead.=C2=A0 That might <br>get reported by taking pessimistic estima=
tes of the variable bits plus the results<br>of a path mtu discovery.=C2=A0=
 How it gets reported may be more important than the details<br>of how the =
stack estimates it, though.<br><br></div><div class=3D"gmail_default" style=
=3D"font-family:tahoma,sans-serif;font-size:small">Failing that, those appl=
ications will try to discern whether inner space headers<br>are in use and =
will do their own estimates of the impact on available space.=C2=A0 Until<b=
r>they have been updated to discern that, the 8 (or 12) octet InSpace heade=
r and the <br>alignment padding will blow them up, as Bob&#39;s text notes.=
=C2=A0 If the working group wants <br></div><div class=3D"gmail_default" st=
yle=3D"font-family:tahoma,sans-serif;font-size:small">to just note the issu=
e and move on, then something along his text is likely fine.=C2=A0 If<br>it=
 wants to suggest more, I still believe recognizing the reality and providi=
ng something<br>better than the current state is worth considering. <br></d=
iv><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font=
-size:small"><br></div><div class=3D"gmail_default" style=3D"font-family:ta=
homa,sans-serif;font-size:small">Just my two cents,<br><br>Ted<br></div><br=
><br><br>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
It only gives the incorrect impression that socket boundaries are<br>
correlated to TCP segment boundaries.<br>
<br>
Additionally, as I&#39;ve already noted, TCP MSS doesn&#39;t take into acco=
unt<br>
the option space needed by either IP or TCP. TCP is in a position to use<br=
>
that value to limit how much application data is added to those headers<br>
AFTER they&#39;re formed.<br>
<br>
At the application API, there&#39;s no meaningful TCP MTU. The space that<b=
r>
remains in a segment depends on the size of the options used *for that<br>
segment*, and those option sizes already vary.<br>
<span class=3D""><font color=3D"#888888"><br>
Joe<br>
</font></span></blockquote></div><br></div></div>

--001a113e986426bcda0508276215--


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From: Ted Hardie <ted.ietf@gmail.com>
To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Inner Space and MTU
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On Tue, Nov 18, 2014 at 11:58 AM, Joe Touch <touch@isi.edu> wrote:

>
>
> On 11/18/2014 11:48 AM, Ted Hardie wrote:
> > Hi Joe,
> >
> > On Tue, Nov 18, 2014 at 11:24 AM, Joe Touch <touch@isi.edu
> > <mailto:touch@isi.edu>> wrote:
> >
> >
> >
> >     On 11/17/2014 11:06 AM, Ted Hardie wrote:
> >     > Hi Bob,
> >     >
> >     > Some thougths in-line.
> >     ...
> >     > So, the best thing to do from my perspective would be to split
> IP_MTU
> >     > into TCP_MTU and UDP_MTU in getsockopt();
> >
> >     TCP might report an MTU for diagnostics, but it's a byte-oriented
> >     protocol so this value shouldn't be part of what's exposed to the
> >     userland application per se.
> >
> >
> > =E2=80=8BAt the moment, you can query the IP_MTU, and some TCP-using ap=
plications
> > attempt to use that information to craft messages that fit within the
> MTU.
>
> I can play PowerBall too, but that doesn't mean I'll be right (or that
> the app will either).
>
>
=E2=80=8BHmm.  The PowerBall odds are: 1 in 175,223,510 for the Jackpot.  I=
f
those approximate your personal odds of guessing the payload available
to TCP correctly given an MTU, you have one heck of a Path MTU and
some startling variability.  Or possibly you just meant getting your money
back,
which is approximately 1 in 55.  Even that seems pessimistic, since the
actual
problem only occurs if the application's guess is over-generous, resulting
in fragmentation.  But let's take it as read.


IMO, an app that tries to use MTU when interfacing is doing a lot of
> work that TCP basically ignores in at least two ways:
>
>         1) the options, which (as noted) aren't considered and can
>         vary (and the user can't predict - users don't know how
>         the might be aligned, even if they're stable)
>
>         2) TCP grabs what it needs to fill the next segment
>         based on what's available in the window. That window
>         need not always indicate a complete MSS; its position
>         depends on what has been ACKd of what has been sent
>
> Even with Nagle disabled, there's no way to control that interaction. So
> a TCP MTU just helps the app operate under a delusion.
>
> TCP is most efficient when the app hands over data as soon as it can.
> Anything else is asking for interference, not potentially helping.
>
>
=E2=80=8BApplications aren't interested in TCP efficiency in isolation; the=
y
=E2=80=8Bare interested in it in an application context.  Sometimes that
causes them to make choices different from "hand over the data
as soon as you can in whatever chunks you have".


> > The IP_MTU may well be a foot gun in its common usage, but it does get
> > used.
>
> That makes more sense for raw socket apps and UDP, but makes no sense at
> all for TCP for the same reason.
>
> =E2=80=8BIt does make more sense for UDP, in general, and it sometimes dr=
ives
applications to recreate TCP's other characteristics on top of UDP.=E2=80=
=8B



> > If those applications could query the socket to discover what the maxim=
um
> > effective TCP payload size was, I believe they would use it instead.
>
> Sure. I might use it to guess a PowerBall ticket too. That doesn't mean
> it's actually useful or will ever be.
>
> I agree that apps can, will, and already do think they can do better.
> They're wrong. Let's not play into that delusion.
>
>
=E2=80=8BDo you oppose Bob's language as well?

Ted=E2=80=8B





> Joe
>

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<div dir=3D"ltr"><div class=3D"gmail_extra">On Tue, Nov 18, 2014 at 11:58 A=
M, Joe Touch <span dir=3D"ltr">&lt;<a href=3D"mailto:touch@isi.edu" target=
=3D"_blank">touch@isi.edu</a>&gt;</span> wrote:<br><div class=3D"gmail_quot=
e"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bord=
er-left:1px solid rgb(204,204,204);padding-left:1ex"><span class=3D""><br>
<br>
On 11/18/2014 11:48 AM, Ted Hardie wrote:<br>
&gt; Hi Joe,<br>
&gt;<br>
&gt; On Tue, Nov 18, 2014 at 11:24 AM, Joe Touch &lt;<a href=3D"mailto:touc=
h@isi.edu">touch@isi.edu</a><br>
</span><span class=3D"">&gt; &lt;mailto:<a href=3D"mailto:touch@isi.edu">to=
uch@isi.edu</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0On 11/17/2014 11:06 AM, Ted Hardie wrote:<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Hi Bob,<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Some thougths in-line.<br>
&gt;=C2=A0 =C2=A0 =C2=A0...<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; So, the best thing to do from my perspective w=
ould be to split IP_MTU<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; into TCP_MTU and UDP_MTU in getsockopt();<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0TCP might report an MTU for diagnostics, but it&#39=
;s a byte-oriented<br>
&gt;=C2=A0 =C2=A0 =C2=A0protocol so this value shouldn&#39;t be part of wha=
t&#39;s exposed to the<br>
&gt;=C2=A0 =C2=A0 =C2=A0userland application per se.<br>
&gt;<br>
&gt;<br>
&gt; =E2=80=8BAt the moment, you can query the IP_MTU, and some TCP-using a=
pplications<br>
&gt; attempt to use that information to craft messages that fit within the =
MTU.<br>
<br>
</span>I can play PowerBall too, but that doesn&#39;t mean I&#39;ll be righ=
t (or that<br>
the app will either).<br>
<br></blockquote><div><br><div class=3D"gmail_default" style=3D"font-family=
:tahoma,sans-serif;font-size:small">=E2=80=8BHmm.=C2=A0 The PowerBall odds =
are: 1 in 175,223,510 for the Jackpot.=C2=A0 If <br>those approximate your =
personal odds of guessing the payload available<br>to TCP correctly given a=
n MTU, you have one heck of a Path MTU and<br>some startling variability.=
=C2=A0 Or possibly you just meant getting your money back, <br>which is app=
roximately 1 in 55.=C2=A0 Even that seems pessimistic, since the actual<br>=
problem only occurs if the application&#39;s guess is over-generous, result=
ing<br>in fragmentation.=C2=A0 But let&#39;s take it as read.<br></div><br>=
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
IMO, an app that tries to use MTU when interfacing is doing a lot of<br>
work that TCP basically ignores in at least two ways:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 1) the options, which (as noted) aren&#39;t con=
sidered and can<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 vary (and the user can&#39;t predict - users do=
n&#39;t know how<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 the might be aligned, even if they&#39;re stabl=
e)<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 2) TCP grabs what it needs to fill the next seg=
ment<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 based on what&#39;s available in the window. Th=
at window<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 need not always indicate a complete MSS; its po=
sition<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 depends on what has been ACKd of what has been =
sent<br>
<br>
Even with Nagle disabled, there&#39;s no way to control that interaction. S=
o<br>
a TCP MTU just helps the app operate under a delusion.<br>
<br>
TCP is most efficient when the app hands over data as soon as it can.<br>
Anything else is asking for interference, not potentially helping.<br>
<span class=3D""><br></span></blockquote><div><br><div class=3D"gmail_defau=
lt" style=3D"font-family:tahoma,sans-serif;font-size:small">=E2=80=8BApplic=
ations aren&#39;t interested in TCP efficiency in isolation; they<br></div>=
<div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-si=
ze:small">=E2=80=8Bare interested in it in an application context.=C2=A0 So=
metimes that<br></div><div class=3D"gmail_default" style=3D"font-family:tah=
oma,sans-serif;font-size:small">causes them to make choices different from =
&quot;hand over the data<br>as soon as you can in whatever chunks you have&=
quot;.<br></div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1e=
x"><span class=3D"">
&gt; The IP_MTU may well be a foot gun in its common usage, but it does get=
<br>
&gt; used.<br>
<br>
</span>That makes more sense for raw socket apps and UDP, but makes no sens=
e at<br>
all for TCP for the same reason.<br>
<span class=3D""><br></span></blockquote><div><div class=3D"gmail_default" =
style=3D"font-family:tahoma,sans-serif;font-size:small">=E2=80=8BIt does ma=
ke more sense for UDP, in general, and it sometimes drives<br>applications =
to recreate TCP&#39;s other characteristics on top of UDP.=E2=80=8B</div><b=
r>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><span class=
=3D"">
&gt; If those applications could query the socket to discover what the maxi=
mum<br>
&gt; effective TCP payload size was, I believe they would use it instead.<b=
r>
<br>
</span>Sure. I might use it to guess a PowerBall ticket too. That doesn&#39=
;t mean<br>
it&#39;s actually useful or will ever be.<br>
<br>
I agree that apps can, will, and already do think they can do better.<br>
They&#39;re wrong. Let&#39;s not play into that delusion.<br>
<span class=3D""><font color=3D"#888888"><br></font></span></blockquote><di=
v><br><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;f=
ont-size:small">=E2=80=8BDo you oppose Bob&#39;s language as well?<br><br><=
/div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;fo=
nt-size:small">Ted=E2=80=8B</div><br><br><br>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rg=
b(204,204,204);padding-left:1ex"><span class=3D""><font color=3D"#888888">
Joe<br>
</font></span></blockquote></div><br></div></div>

--001a11360240316410050828421b--


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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Joe Touch <touch@isi.edu>, Ted Hardie <ted.ietf@gmail.com>
Thread-Topic: [tcpm] Inner Space and MTU
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Date: Wed, 19 Nov 2014 20:23:08 +0000
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From: "Scheffenegger, Richard" <rs@netapp.com>
To: Pasi Sarolahti <pasi.sarolahti@iki.fi>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Thread-Topic: [tcpm] WG Adoption of draft-zimmermann-tcpm-undeployed-01
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Subject: Re: [tcpm] WG Adoption of draft-zimmermann-tcpm-undeployed-01
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I'm not opposed, but wanted point out that I'm very much in favor of this, =
and similar documents that try match the RFC series with deployment.

Richard Scheffenegger


-----Original Message-----
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Pasi Sarolahti
Sent: Mittwoch, 19. November 2014 12:15
To: tcpm@ietf.org Extensions
Subject: [tcpm] WG Adoption of draft-zimmermann-tcpm-undeployed-01

Hi,

The TCPM meeting last week supported adoption of draft-zimmermann-tcpm-unde=
ployed-01 (Moving Undeployed TCP Extensions to Historic and Informational S=
tatus). This mail is to check that people who did not attend the meeting ar=
e content with the meeting consensus to adopt the draft. Therefore, please =
let us know by the end of this week, if you are against adoption -- prefera=
bly on the mailing list.

Thanks!

- Pasi

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


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Subject: Re: [tcpm] Inner Space and MTU
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On 11/19/2014 12:23 PM, Scheffenegger, Richard wrote:
> Hello Joe,
> 
>>> Applications aren't interested in TCP efficiency in isolation; they 
>>> are interested in it in an application context.  Sometimes that 
>>> causes them to make choices different from "hand over the data as soon 
>>> as you can in whatever chunks you have".
>>
>> When they do, they ought to be using a different transport. 
> 
> I tend to agree if we'd be living in a perfect world...
> 
>> It isn't useful for an app to game its interface to TCP.
> 
> au contraire! In my context (storage) there is a very well-known
> protocol that switched from UDP to TCP (at a time when SCTP wasn't
> available; one can argue, it still isn't in most enterprises...). In
> order to avoid excessive head-of-line blocking, it does make sense for
> an app to feed the sending socket the data in small chunks, to allow
> more critical data to get processed (that is, transported, dealt with on
> the server, and responded to) faster. And that particular piece of
> behavior is actually implemented in one particular user-space NFS
> implementation, fairly recently (2011/12 IIRC).

To be clear - a given implementation can be gamed all day long.

Once the OS or protocol is updated, though, *all bets are off*, and all
of those optimizations can - and will - come back to bite the programmer.

> Or, to use a (in-)famous word, to circumvent bufferbloat a little bit
> at least (tcp send buffer).

Which comes back and bites you once AQM is deployed.

> Also, even though NFS does not explicitly maintain segment
> boundaries (or cares about TCP MSS), the frequent use of PSH serves to align the
> RPC/NFS calls to the extent that many implementations fast-path NFS
> processing by expecting fixed (tcp data) offsets...

Note that PSH is part of the TCP API.

Joe


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On 11/19/2014 3:15 AM, Pasi Sarolahti wrote:
> Hi,
> 
> The TCPM meeting last week supported adoption of
> draft-zimmermann-tcpm-undeployed-01 (Moving Undeployed TCP Extensions to
> Historic and Informational Status). This mail is to check that people
> who did not attend the meeting are content with the meeting consensus to
> adopt the draft. ...

I'm OK with this document as a WG item, but as a point of order, I am not.

The email list is the primary venue for WG decisions. Any events
off-list are at best a reason to ask the list, but it is the list that
decides - not those who attend a meeting.

In particular, the list needs to exhibit a positive "rough consensus" on
this decision. If the list is largely silent, then the decision must be
"no".

Joe

PS - sorry if I'm jumping on a note that wasn't intended to side-step
process, but it's how these queries are worded that participants end up
sensing how the IETF operates.


From nobody Thu Nov 20 09:34:18 2014
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Subject: [tcpm] Last Call: <draft-ietf-tcpm-accecn-reqs-07.txt> (Problem Statement and Requirements for a More Accurate ECN Feedback) to Informational RFC
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The IESG has received a request from the TCP Maintenance and Minor
Extensions WG (tcpm) to consider the following document:
- 'Problem Statement and Requirements for a More Accurate ECN Feedback'
  <draft-ietf-tcpm-accecn-reqs-07.txt> as Informational RFC

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2014-12-04. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   Explicit Congestion Notification (ECN) is a mechanism where network
   nodes can mark IP packets instead of dropping them to indicate
   congestion to the end-points.  An ECN-capable receiver will feed this
   information back to the sender.  ECN is specified for TCP in such a
   way that it can only feed back one congestion signal per Round-Trip
   Time (RTT).  In contrast, ECN for other transport protocols, such as
   RTP/UDP and SCTP, is specified with more accurate ECN feedback.
   Recent new TCP mechanisms (like ConEx or DCTCP) need more accurate
   ECN feedback in the case where more than one marking is received in
   one RTT.  This document specifies requirements for an update to the
   TCP protocol to provide more accurate ECN feedback.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-tcpm-accecn-reqs/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-tcpm-accecn-reqs/ballot/


No IPR declarations have been submitted directly on this I-D.



From nobody Thu Nov 20 12:25:17 2014
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On 20 Nov 2014, at 19:00, Joe Touch <touch@isi.edu> wrote:

> On 11/19/2014 3:15 AM, Pasi Sarolahti wrote:
>> Hi,
>>=20
>> The TCPM meeting last week supported adoption of
>> draft-zimmermann-tcpm-undeployed-01 (Moving Undeployed TCP Extensions =
to
>> Historic and Informational Status). This mail is to check that people
>> who did not attend the meeting are content with the meeting consensus =
to
>> adopt the draft. ...
>=20
> I'm OK with this document as a WG item, but as a point of order, I am =
not.
>=20
> The email list is the primary venue for WG decisions. Any events
> off-list are at best a reason to ask the list, but it is the list that
> decides - not those who attend a meeting.
>=20
> In particular, the list needs to exhibit a positive "rough consensus" =
on
> this decision. If the list is largely silent, then the decision must =
be
> "no=94.

Point taken, the wording in my Email was not all that successful, sorry =
for that. Of course, all opinions for or against adoption are welcome, =
regardless of the meeting events. (But I would note that we discussed =
adoption already before the meeting on this list)

[As far as I could read, RFC 2418 does not specify clear =
primary/secondary decision relationship between meeting and mailing =
list, but of course both should be taken into account. Anyways, I think =
it=92s better to keep further process education off-list, unless there =
is something relevant for this draft.]

- Pasi


From nobody Thu Nov 27 05:32:07 2014
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Subject: Re: [tcpm] WG Adoption of draft-zimmermann-tcpm-undeployed-01
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Hi,

There have been two Emails in support and none against for TCPM WG =
adoption of draft-zimmermann-tcpm-undeployed, in addition to the clear =
consensus for adoption in the Honolulu meeting. Therefore, I would like =
to ask the authors to submit the next version of the document as =
draft-ietf-tcpm-undeployed-00.

Thanks!

- Pasi


On 19 Nov 2014, at 13:15, Pasi Sarolahti <pasi.sarolahti@iki.fi> wrote:

> Hi,
>=20
> The TCPM meeting last week supported adoption of =
draft-zimmermann-tcpm-undeployed-01 (Moving Undeployed TCP Extensions to =
Historic and Informational Status). This mail is to check that people =
who did not attend the meeting are content with the meeting consensus to =
adopt the draft. Therefore, please let us know by the end of this week, =
if you are against adoption -- preferably on the mailing list.
>=20
> Thanks!
>=20
> - Pasi
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri Nov 28 09:00:14 2014
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-undeployed-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.

        Title           : Moving Undeployed TCP Extensions to Historic and Informational Status -- An addition to RFC 6247
        Authors         : Alexander Zimmermann
                          Wesley M. Eddy
                          Lars Eggert
	Filename        : draft-ietf-tcpm-undeployed-00.txt
	Pages           : 4
	Date            : 2014-11-27

Abstract:
   This document reclassifies several TCP extensions that have either
   been superceded or never seen widespread use to Historic status.  The
   affected RFCs are RFC 675, RFC 721, RFC 879, RFC 1078, and RFC 6013.
   Additionally, it reclassifies RFC 813, RFC 814, RFC 816, RFC 817, RFC
   872, RFC 896, and RFC 964 to Informational status.  Most of those
   RFCs are today part of RFC 1122.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-undeployed/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tcpm-undeployed-00


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 nobody Fri Nov 28 09:01:51 2014
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URL: http://datatracker.ietf.org/wg/tcpm/charter/

