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From: Martin Thomson <martin.thomson@gmail.com>
Date: Wed, 3 May 2017 17:44:08 +1000
Message-ID: <CABkgnnV30NOwVSfdtV0AdwXioHUY_1JouOApGaBeTDHnJptvSA@mail.gmail.com>
Subject: Re: Updated Public Reset authentication patch
To: Brian Trammell <ietf@trammell.ch>
Cc: QUIC WG <quic@ietf.org>
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Sorry about not getting back to your essay as quickly as I intended.  As ou

On 28 April 2017 at 19:27, Brian Trammell <ietf@trammell.ch> wrote:
> So unless I'm missing something here, I'm not sure that "privileging just=
 one path" is a thing that happens.

I was primarily talking about migration.  If you move from one network
to another, the path prior to the migration gets the handshake, the
path after the migration gets the close.  The first is privileged in
the sense that it might see a reset verifier.  And yes, multipath
makes this much more obvious a problem.

> (I'm ignoring multipath here, because in the multipath case, I suspect we=
'll end up with an MPTCP-like design [...]

(I don't share your assumption about its design; I want a single
cryptographic context, for one.  FWIW, I'm thinking of multipath as
make before break without the break.  That's also why I want to make
any path signals explicitly path-specific.)

>> We can work around that, but it either requires removing packet
>> numbers completely, or greatly enhancing the information exposed to
>> the path.  For instance, if we had a packet number echo the server
>> would be able to pick a plausible packet number for its reset.
>
> ...I wouldn't say we need to greatly enhance it; this PR enhances it enou=
gh for the purposes of separating endpoint-generated public resets from spo=
ofed ones...

Note that my PR does not do this.  It would require changes to do so.

> I agree completely that it's desirable to separate the discussions about =
end-to-end from end-to-path and path-to-end signaling; but public reset, in=
 particular, counfounds that desire, since it's an end-to-end signal that r=
adiates information to the path. We should focus on, and probably work to m=
inimize, these inseparable effects.

Yes, I agree.  Though I'm not as enthusiastic about a design that
makes public reset indistinguishable from other traffic.  We could do
it but the only way I know how to make that work is to force endpoints
to try hashing every packet of a certain size that fails to decrypt.
We could use a key phase change to reduce that to the points at which
we roll keys, so it's not awful, but it's probably also creating some
nasty externalities (e.g., packets of the magic size cause connections
to fail).

> I don't yet see a way to get better-than-economic defense without allowin=
g a path element to authenticate an endpoint, so that it can verify that ea=
ch header bit carrying a signal indeed came from the endpoint it should hav=
e. This is vastly more expencive, and essentially goes down the path of bui=
lding QUIC atop a multiparty cryptographic protocol -- mcTLS occupies a far=
 corner of this design space. I don't think it's possible to deploy somethi=
ng like this safely in the Internet with the primitives we have available t=
o us, but IANAC.

Nor I, but you are right in observing that it gets funky if you want a
verifiable end-to-middle signal.  One-time passwords like in the
public reset design only work for one bit of information, and I assume
that any generic system here needs to signal more bits, which could
mean public key crypto and I don't think that anyone wants that.
(There's probably something else out there, but nothing I can think of
works for a potentially large audience and arbitrary messages.)


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From: "Eggert, Lars" <lars@netapp.com>
To: IETF QUIC WG <quic@ietf.org>
Subject: Open-source TLS 1.3 with an API suitable for QUIC?
Thread-Topic: Open-source TLS 1.3 with an API suitable for QUIC?
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Hi,

I've been looking at =
https://github.com/quicwg/base-drafts/wiki/First-Implementation-Draft =
and wondering which open-source TLS implementations are suitable for =
providing the required TLS 1.3 functionality over UDP. Specifically, for =
a C implementation.

(There is a list of implementations at =
https://github.com/tlswg/tls13-spec/wiki/Implementations, but it's not =
clear which - if any - expose an API that would be suitable for =
operating over UDP, for use with a QUIC stack.)

Any pointers?

Thanks,
Lars



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From: Kazuho Oku <kazuhooku@gmail.com>
Date: Wed, 3 May 2017 18:06:59 +0900
Message-ID: <CANatvzwRF4gCd6Vsr8vjDzuCNAUGQkdWekFCp7HCfwnn1mSMOQ@mail.gmail.com>
Subject: Re: Open-source TLS 1.3 with an API suitable for QUIC?
To: "Eggert, Lars" <lars@netapp.com>
Cc: IETF QUIC WG <quic@ietf.org>
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Just FYI,

I maintain picotls (one of the pure-C TLS 1.3 implementations listed
on the wiki) which I have designed with QUIC in mind. The API of
picotls is designed like a codec (rather than an abstraction layer on
top of the sockets API) so that it can be naturally integrated with
UDP-based protocols.

Actually, I have started working on to implement QUIC using the
library and have a dedicated branch for adding the missing parts that
are required for QUIC[1]. I believe that the only piece that is
missing at the moment is the Exporter.

[1] https://github.com/h2o/picotls/tree/kazuho/quic


2017-05-03 17:37 GMT+09:00 Eggert, Lars <lars@netapp.com>:
> Hi,
>
> I've been looking at https://github.com/quicwg/base-drafts/wiki/First-Implementation-Draft and wondering which open-source TLS implementations are suitable for providing the required TLS 1.3 functionality over UDP. Specifically, for a C implementation.
>
> (There is a list of implementations at https://github.com/tlswg/tls13-spec/wiki/Implementations, but it's not clear which - if any - expose an API that would be suitable for operating over UDP, for use with a QUIC stack.)
>
> Any pointers?
>
> Thanks,
> Lars
>
>



-- 
Kazuho Oku


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> On 03 May 2017, at 09:44, Martin Thomson <martin.thomson@gmail.com> =
wrote:
>=20
> Sorry about not getting back to your essay as quickly as I intended.  =
As ou
>=20
> On 28 April 2017 at 19:27, Brian Trammell <ietf@trammell.ch> wrote:
>> So unless I'm missing something here, I'm not sure that "privileging =
just one path" is a thing that happens.
>=20
> I was primarily talking about migration.  If you move from one network
> to another, the path prior to the migration gets the handshake, the
> path after the migration gets the close.  The first is privileged in
> the sense that it might see a reset verifier.  And yes, multipath
> makes this much more obvious a problem.

Ah, understood.

>> (I'm ignoring multipath here, because in the multipath case, I =
suspect we'll end up with an MPTCP-like design [...]
>=20
> (I don't share your assumption about its design; I want a single
> cryptographic context, for one.  FWIW, I'm thinking of multipath as
> make before break without the break.  That's also why I want to make
> any path signals explicitly path-specific.)

(Aside: I'm pretty sure we agree on the design here: I intended less by =
"MPTCP-like" than was taken. :) Single cryptographic context is IMO a =
no-brainer, as is designing it with similar semantics to =
make-before-break migration. But let's have that discussion when it =
comes on charter.)

>=20
>>> We can work around that, but it either requires removing packet
>>> numbers completely, or greatly enhancing the information exposed to
>>> the path.  For instance, if we had a packet number echo the server
>>> would be able to pick a plausible packet number for its reset.
>>=20
>> ...I wouldn't say we need to greatly enhance it; this PR enhances it =
enough for the purposes of separating endpoint-generated public resets =
from spoofed ones...
>=20
> Note that my PR does not do this.  It would require changes to do so.

Ah. Yep. Sorry, I carried over some state from PR20 the first time I =
read this.

>> I agree completely that it's desirable to separate the discussions =
about end-to-end from end-to-path and path-to-end signaling; but public =
reset, in particular, counfounds that desire, since it's an end-to-end =
signal that radiates information to the path. We should focus on, and =
probably work to minimize, these inseparable effects.
>=20
> Yes, I agree.  Though I'm not as enthusiastic about a design that
> makes public reset indistinguishable from other traffic.  We could do
> it but the only way I know how to make that work is to force endpoints
> to try hashing every packet of a certain size that fails to decrypt.
> We could use a key phase change to reduce that to the points at which
> we roll keys, so it's not awful, but it's probably also creating some
> nasty externalities (e.g., packets of the magic size cause connections
> to fail).

Ew.

>> I don't yet see a way to get better-than-economic defense without =
allowing a path element to authenticate an endpoint, so that it can =
verify that each header bit carrying a signal indeed came from the =
endpoint it should have. This is vastly more expencive, and essentially =
goes down the path of building QUIC atop a multiparty cryptographic =
protocol -- mcTLS occupies a far corner of this design space. I don't =
think it's possible to deploy something like this safely in the Internet =
with the primitives we have available to us, but IANAC.
>=20
> Nor I, but you are right in observing that it gets funky if you want a
> verifiable end-to-middle signal.  One-time passwords like in the
> public reset design only work for one bit of information, and I assume
> that any generic system here needs to signal more bits, which could
> mean public key crypto and I don't think that anyone wants that.
> (There's probably something else out there, but nothing I can think of
> works for a potentially large audience and arbitrary messages.)

One could probably design a proof verification to signal one of N (for =
small N) conditions, of which one would be reset, such that the =
signaling of a verification would require the production of a new proof =
for further signaling (by providing for one verification value per =
condition, all values of which are simple transforms of the bits hashed =
to create the proof -- verification time scales linearly with the number =
of possible conditions). This isn't completely generic, but it's better =
than one bit.

With respect to close/reset*, there's another possibility on the =
verifiability continuum that we allude to (but don't fully define) in =
the state machine draft: "cancellable". Here, an endpoint can signal a =
close, but so can any on-path device that has access to a valid =
connection ID and packet number. A spoofed close will be detected by the =
endpoint (since it won't decrypt), which will continue sending traffic =
on the same n-tuple and connection ID. The presence of continued traffic =
on the n-tuple is itself a signal that a prior close was spoofed. Spoof =
cancellation semantics only apply to close signaling, and there are =
weird corner cases to consider (what about a spoofed close coincident =
with a real migration? can two collaborating nodes use cancellation to =
keep legitimate firewall pinholes open for illegitimate purposes? etc.), =
but this does represent another possible design...

* We still need to clarify what we mean by reset: is this "all =
connection termination" or "abnormal connection termination" or =
"unrecoverable connectivity error"?

Cheers,

Brian

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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Proposed changes to cleartext packets
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Issue: https://github.com/quicwg/base-drafts/issues/442

I've heard quite a few concerns about the current design for server
selection of the connection ID.  The primary concern is that the
choice of final connection ID is made on the same flight as the server
handshake messages - the same flight where the server is required to
commit non-trivial state and computation resources to the connection.

The current design is actually anathema to certain load balancer
designs.  If the node that receives a packet wants to get rid of it
before sending the final server flight, the load balancer can no
longer route statelessly.

The server folks I've spoken to want to pick a connection ID during
both version negotiation and stateless reject/address validation.
This allows them to do trivial or at least lightweight processing on
the node that receives the first packet AND steer the packet to the
right node.


I'm proposing that we move to a special "first client packet" approach
rather than a "last server handshake packet" approach.  This allows
servers to identify which packet has a client-selected connection ID
and create special logic for that.


Issue: https://github.com/quicwg/base-drafts/pull/482#discussion_r114445062

The second problem is one that arises from stateless rejects.  A few
people observed that there is a disconnect between what the server
sends in a stateless reject and what it subsequently sends.

For instance, the server can't generate a stateless reject then send
subsequent packets with the expected packet numbers.  The reason: it
might need to generate multiple stateless reject packets in response
to multiple handshake attempts from a client.  It can't save the
packet number it chose in the cookie it sends because that would mean
sending different values for the same octets on stream 0 and clients
could get confused by that.

Also, it was previously necessary to resort to shenanigans at the
server after a stateless reject.  The second ClientHello will start at
a non-zero offset on stream 0, but the server will effectively have
never received that data.  That means pretending that the data was
received and delivered already, but then being able to validate the
offset and back out the change if validation fails.


All in all, that's more special handling than is ideal.  Jana proposed
resetting the transport side of things when this happens.  That means
resetting the stream 0 offset and not getting bent out of shape when
the server chooses a non-contiguous packet number.  That seems like a
reasonable idea, and I'm proposing a new cleartext message type for
that so that it can be explicit when the server is doing this.  That
message comes with its own special treatment (it is used for
HelloRetryRequest only, and HelloRetryRequest has to use it, it has to
be one packet).

This externalizes some of the server costs for stateless rejects, and
it adds another special case packet type.  Neither of which I'm
entirely happy about, but it seems like a reasonable compromise.


These are two separate issues, but they are intermingled in annoying
ways, so I have a single draft PR for both changes:

   https://github.com/quicwg/base-drafts/pull/493

There is a question in the PR about two alternative designs:

1. Require the client to send an initial packet after version
negotiation on the basis that version negotiation is a complete
do-over.

2. Maybe also after a stateless reject, which would mean that all
packets containing the ClientHello would use that "initial" type. That
would mean that only the initial type would have packet size
restrictions, which is nice.

I think that these could be an improvement, but I'd like to hear from
server folks.  The cost here is that after version negotiation and
maybe also a stateless reject, you would route the packet from the
client as if it were a client-selected connection ID.  I'm not sure
how much that matters.  What you do get is up to three opportunities
to reroute a client (during version negotiation, optionally during
stateless reject, and during the handshake).


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Wed, 3 May 2017 20:52:14 +1000
Message-ID: <CABkgnnXtLqPTDFj2MjPTWnYydMdhjQpYWgf_VAt5vSq1UPfzVw@mail.gmail.com>
Subject: Re: Updated Public Reset authentication patch
To: "Brian Trammell (IETF)" <ietf@trammell.ch>
Cc: QUIC WG <quic@ietf.org>
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On 3 May 2017 at 20:06, Brian Trammell (IETF) <ietf@trammell.ch> wrote:
> With respect to close/reset*, there's another possibility on the verifiab=
ility continuum that we allude to (but don't fully define) in the state mac=
hine draft: "cancellable". Here, an endpoint can signal a close, but so can=
 any on-path device that has access to a valid connection ID and packet num=
ber. A spoofed close will be detected by the endpoint (since it won't decry=
pt), which will continue sending traffic on the same n-tuple and connection=
 ID. The presence of continued traffic on the n-tuple is itself a signal th=
at a prior close was spoofed. Spoof cancellation semantics only apply to cl=
ose signaling, and there are weird corner cases to consider (what about a s=
poofed close coincident with a real migration? can two collaborating nodes =
use cancellation to keep legitimate firewall pinholes open for illegitimate=
 purposes? etc.), but this does represent another possible design...

Yes, someone suggested that ICMP unreachable might have this property.


From nobody Wed May  3 05:32:00 2017
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From: "Salz, Rich" <rsalz@akamai.com>
To: "Eggert, Lars" <lars@netapp.com>, IETF QUIC WG <quic@ietf.org>
Subject: RE: Open-source TLS 1.3 with an API suitable for QUIC?
Thread-Topic: Open-source TLS 1.3 with an API suitable for QUIC?
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> wondering which open-source TLS
> implementations are suitable for providing the required TLS 1.3 functiona=
lity
> over UDP. Specifically, for a C implementation.

If there are API's that OpenSSL would need to provide, please let me know (=
ideally within a month) and we will work to provide them.


From nobody Wed May  3 07:09:53 2017
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On 5/3/2017 12:44 AM, Martin Thomson wrote:
>> I agree completely that it's desirable to separate the discussions abo=
ut end-to-end from end-to-path and path-to-end signaling; but public rese=
t, in particular, counfounds that desire, since it's an end-to-end signal=
 that radiates information to the path. We should focus on, and probably =
work to minimize, these inseparable effects.
> Yes, I agree.  Though I'm not as enthusiastic about a design that
> makes public reset indistinguishable from other traffic.  We could do
> it but the only way I know how to make that work is to force endpoints
> to try hashing every packet of a certain size that fails to decrypt.
> We could use a key phase change to reduce that to the points at which
> we roll keys, so it's not awful, but it's probably also creating some
> nasty externalities (e.g., packets of the magic size cause connections
> to fail).
We could do better than "hash every packet" if instead of just sending a
reset verifier the server also sent a reset nonce. The public reset
would then contain something like header|reset-nonce|reset-proof, where
proof =3D hash(reset-nonce|reset-verifier). Servers with reduced state
could just derive both nonce and verifier from a static secret and the
connection ID. Clients that remember both nonce and verifier would just
check that the nonce matches the preset value before computing the hash,
thus greatly reducing risk of spending CPU cycles on random garbage.

-- Christian Huitema

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<html>
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    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <p><br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 5/3/2017 12:44 AM, Martin Thomson
      wrote:<br>
    </div>
    <blockquote
cite="mid:CABkgnnV30NOwVSfdtV0AdwXioHUY_1JouOApGaBeTDHnJptvSA@mail.gmail.com"
      type="cite">
      <blockquote type="cite" style="color: #000000;">
        <pre wrap="">I agree completely that it's desirable to separate the discussions about end-to-end from end-to-path and path-to-end signaling; but public reset, in particular, counfounds that desire, since it's an end-to-end signal that radiates information to the path. We should focus on, and probably work to minimize, these inseparable effects.
</pre>
      </blockquote>
      <pre wrap="">Yes, I agree.  Though I'm not as enthusiastic about a design that
makes public reset indistinguishable from other traffic.  We could do
it but the only way I know how to make that work is to force endpoints
to try hashing every packet of a certain size that fails to decrypt.
We could use a key phase change to reduce that to the points at which
we roll keys, so it's not awful, but it's probably also creating some
nasty externalities (e.g., packets of the magic size cause connections
to fail).</pre>
    </blockquote>
    We could do better than "hash every packet" if instead of just
    sending a reset verifier the server also sent a reset nonce. The
    public reset would then contain something like
    header|reset-nonce|reset-proof, where proof =
    hash(reset-nonce|reset-verifier). Servers with reduced state could
    just derive both nonce and verifier from a static secret and the
    connection ID. Clients that remember both nonce and verifier would
    just check that the nonce matches the preset value before computing
    the hash, thus greatly reducing risk of spending CPU cycles on
    random garbage.<br>
    <br>
    -- Christian Huitema<br>
  </body>
</html>

--------------35F7AB14BEE0F0419A1E68A5--


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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 May 2017 07:32:37 -0700
Message-ID: <CABcZeBNkYzM5H2uzgc_a5NAi28=rWRyW+U0o6q+3Oj+KGWoW0A@mail.gmail.com>
Subject: Re: Open-source TLS 1.3 with an API suitable for QUIC?
To: "Eggert, Lars" <lars@netapp.com>
Cc: IETF QUIC WG <quic@ietf.org>
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I believe that NSS already has essentially what you would need, but if it
doesn't,
please file bugs.

-Ekr


On Wed, May 3, 2017 at 1:37 AM, Eggert, Lars <lars@netapp.com> wrote:

> Hi,
>
> I've been looking at https://github.com/quicwg/base-drafts/wiki/First-
> Implementation-Draft and wondering which open-source TLS implementations
> are suitable for providing the required TLS 1.3 functionality over UDP.
> Specifically, for a C implementation.
>
> (There is a list of implementations at https://github.com/tlswg/
> tls13-spec/wiki/Implementations, but it's not clear which - if any -
> expose an API that would be suitable for operating over UDP, for use with a
> QUIC stack.)
>
> Any pointers?
>
> Thanks,
> Lars
>
>
>

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<div dir=3D"ltr">I believe that NSS already has essentially what you would =
need, but if it doesn&#39;t,<div>please file bugs.<br><div><br></div><div>-=
Ekr</div><div><br></div></div></div><div class=3D"gmail_extra"><br><div cla=
ss=3D"gmail_quote">On Wed, May 3, 2017 at 1:37 AM, Eggert, Lars <span dir=
=3D"ltr">&lt;<a href=3D"mailto:lars@netapp.com" target=3D"_blank">lars@neta=
pp.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
I&#39;ve been looking at <a href=3D"https://github.com/quicwg/base-drafts/w=
iki/First-Implementation-Draft" rel=3D"noreferrer" target=3D"_blank">https:=
//github.com/quicwg/<wbr>base-drafts/wiki/First-<wbr>Implementation-Draft</=
a> and wondering which open-source TLS implementations are suitable for pro=
viding the required TLS 1.3 functionality over UDP. Specifically, for a C i=
mplementation.<br>
<br>
(There is a list of implementations at <a href=3D"https://github.com/tlswg/=
tls13-spec/wiki/Implementations" rel=3D"noreferrer" target=3D"_blank">https=
://github.com/tlswg/<wbr>tls13-spec/wiki/<wbr>Implementations</a>, but it&#=
39;s not clear which - if any - expose an API that would be suitable for op=
erating over UDP, for use with a QUIC stack.)<br>
<br>
Any pointers?<br>
<br>
Thanks,<br>
Lars<br>
<br>
<br>
</blockquote></div><br></div>

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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 May 2017 07:40:58 -0700
Message-ID: <CABcZeBP2aZQhU3S=aB62+QOKfxpuss5iRrLL6df=vHAsZi5Oww@mail.gmail.com>
Subject: Re: Updated Public Reset authentication patch
To: Martin Thomson <martin.thomson@gmail.com>
Cc: Brian Trammell <ietf@trammell.ch>, QUIC WG <quic@ietf.org>
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> Yes, I agree.  Though I'm not as enthusiastic about a design that
> makes public reset indistinguishable from other traffic.  We could do
> it but the only way I know how to make that work is to force endpoints
> to try hashing every packet of a certain size that fails to decrypt.
>

It's actually pretty easy. The only reason for the hash preimage trick
is that you want public verifiability.

I'm going to call this "Stateless Reset" to differentiate from "Public
Reset".
Assume the server has a long-term key K (as in the hash design).

In EncryptedExtensions, the client sends V = HKDF(K, <conn-id>).
When the server wants to reset a connection it generates a properly
formatted packet that consists of:

Header || V || <random bytes>

And when the client receives a packet, before trying to decrypt it, it
looks for V as the initial bytes. No hashing required, and it's just
a memcmp.

-Ekr

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><div class=3D"gmail_quote">=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Yes, I agree.=C2=A0 Though I&#39;m not as en=
thusiastic about a design that<br>
makes public reset indistinguishable from other traffic.=C2=A0 We could do<=
br>
it but the only way I know how to make that work is to force endpoints<br>
to try hashing every packet of a certain size that fails to decrypt.<br></b=
lockquote><div><br></div><div>It&#39;s actually pretty easy. The only reaso=
n for the hash preimage trick</div><div>is that you want public verifiabili=
ty.</div><div><br></div><div>I&#39;m going to call this &quot;Stateless Res=
et&quot; to differentiate from &quot;Public Reset&quot;.</div><div>Assume t=
he server has a long-term key K (as in the hash design).</div><div><br></di=
v><div>In EncryptedExtensions, the client sends V =3D HKDF(K, &lt;conn-id&g=
t;).<br></div><div>When the server wants to reset a connection it generates=
 a properly</div><div>formatted packet that consists of:</div><div><br></di=
v><div>Header || V || &lt;random bytes&gt;</div><div><br></div><div>And whe=
n the client receives a packet, before trying to decrypt it, it</div><div>l=
ooks for V as the initial bytes. No hashing required, and it&#39;s just</di=
v><div>a memcmp.</div><div><br></div><div>-Ekr</div><div><br></div></div></=
div></div>

--94eb2c08763a69d582054e9fa79f--


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From: Mike Bishop <Michael.Bishop@microsoft.com>
To: "Salz, Rich" <rsalz@akamai.com>, "Eggert, Lars" <lars@netapp.com>, "IETF QUIC WG" <quic@ietf.org>
Subject: RE: Open-source TLS 1.3 with an API suitable for QUIC?
Thread-Topic: Open-source TLS 1.3 with an API suitable for QUIC?
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The biggest one (and I don't know if OpenSSL has it) is the ability to get =
the actual bytes that TLS wants to send, rather than having the library own=
 the socket and try to send by itself.  With NSS, apparently you need to pa=
ss it a fake socket and grab the bytes that it tries to send, which is a bi=
t of a pain.  SChannel's model just returns byte buffers to the caller, and=
 it's their job to then transport them, which is mostly what's needed here.

Then there's the ability to add an extension for the QUIC parameters.

-----Original Message-----
From: QUIC [mailto:quic-bounces@ietf.org] On Behalf Of Salz, Rich
Sent: Wednesday, May 3, 2017 5:29 AM
To: Eggert, Lars <lars@netapp.com>; IETF QUIC WG <quic@ietf.org>
Subject: RE: Open-source TLS 1.3 with an API suitable for QUIC?

> wondering which open-source TLS
> implementations are suitable for providing the required TLS 1.3=20
> functionality over UDP. Specifically, for a C implementation.

If there are API's that OpenSSL would need to provide, please let me know (=
ideally within a month) and we will work to provide them.


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From: "Salz, Rich" <rsalz@akamai.com>
To: Mike Bishop <Michael.Bishop@microsoft.com>, "Eggert, Lars" <lars@netapp.com>, IETF QUIC WG <quic@ietf.org>
Subject: RE: Open-source TLS 1.3 with an API suitable for QUIC?
Thread-Topic: Open-source TLS 1.3 with an API suitable for QUIC?
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> The biggest one (and I don't know if OpenSSL has it) is the ability to ge=
t the
> actual bytes that TLS wants to send, rather than having the library own t=
he
> socket and try to send by itself.  With NSS, apparently you need to pass =
it a
> fake socket and grab the bytes that it tries to send, which is a bit of a=
 pain.
> SChannel's model just returns byte buffers to the caller, and it's their =
job to
> then transport them, which is mostly what's needed here.

Yes, it  can be done using a "fake BIO" which is the OpenSSL I/O abstractio=
n; you can read/write memory.  It's also not terrible convenient.
=20
> Then there's the ability to add an extension for the QUIC parameters.

This we have, and it's in better shape :)


From nobody Wed May  3 09:10:17 2017
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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 May 2017 09:07:12 -0700
Message-ID: <CABcZeBPVpWMAc+44JcHYhny__a7H3t38858K+vkH7eETq6r_Nw@mail.gmail.com>
Subject: Re: Open-source TLS 1.3 with an API suitable for QUIC?
To: Mike Bishop <Michael.Bishop@microsoft.com>
Cc: "Salz, Rich" <rsalz@akamai.com>, "Eggert, Lars" <lars@netapp.com>, IETF QUIC WG <quic@ietf.org>
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On Wed, May 3, 2017 at 7:47 AM, Mike Bishop <Michael.Bishop@microsoft.com>
wrote:

> The biggest one (and I don't know if OpenSSL has it) is the ability to get
> the actual bytes that TLS wants to send, rather than having the library own
> the socket and try to send by itself.  With NSS, apparently you need to
> pass it a fake socket and grab the bytes that it tries to send, which is a
> bit of a pain.


Yeah, but it's pretty straightforward. See:
http://searchfox.org/nss/source/gtests/ssl_gtest/test_io.h#53


SChannel's model just returns byte buffers to the caller, and it's their
> job to then transport them, which is mostly what's needed here.
>
> Then there's the ability to add an extension for the QUIC parameters.
>

Right. We would need to add that. We haven't added that to NSS, but it's
not cpmplicated

-Ekr


>
> -----Original Message-----
> From: QUIC [mailto:quic-bounces@ietf.org] On Behalf Of Salz, Rich
> Sent: Wednesday, May 3, 2017 5:29 AM
> To: Eggert, Lars <lars@netapp.com>; IETF QUIC WG <quic@ietf.org>
> Subject: RE: Open-source TLS 1.3 with an API suitable for QUIC?
>
> > wondering which open-source TLS
> > implementations are suitable for providing the required TLS 1.3
> > functionality over UDP. Specifically, for a C implementation.
>
> If there are API's that OpenSSL would need to provide, please let me know
> (ideally within a month) and we will work to provide them.
>
>

--001a114fc7d8cfcea8054ea0db31
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Wed, May 3, 2017 at 7:47 AM, Mike Bishop <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:Michael.Bishop@microsoft.com" target=3D"_blank">Michael.Bishop=
@microsoft.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);pad=
ding-left:1ex">The biggest one (and I don&#39;t know if OpenSSL has it) is =
the ability to get the actual bytes that TLS wants to send, rather than hav=
ing the library own the socket and try to send by itself.=C2=A0 With NSS, a=
pparently you need to pass it a fake socket and grab the bytes that it trie=
s to send, which is a bit of a pain.=C2=A0</blockquote><div><br></div><div>=
Yeah, but it&#39;s pretty straightforward. See:</div><div><a href=3D"http:/=
/searchfox.org/nss/source/gtests/ssl_gtest/test_io.h#53">http://searchfox.o=
rg/nss/source/gtests/ssl_gtest/test_io.h#53</a><br></div><div><br></div><di=
v><br></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"> SChannel&#39=
;s model just returns byte buffers to the caller, and it&#39;s their job to=
 then transport them, which is mostly what&#39;s needed here.<br>
<br>
Then there&#39;s the ability to add an extension for the QUIC parameters.<b=
r></blockquote><div><br></div><div>Right. We would need to add that. We hav=
en&#39;t added that to NSS, but it&#39;s not cpmplicated</div><div><br></di=
v><div>-Ekr</div><div>=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">
<div class=3D"gmail-HOEnZb"><div class=3D"gmail-h5"><br>
-----Original Message-----<br>
From: QUIC [mailto:<a href=3D"mailto:quic-bounces@ietf.org">quic-bounces@ie=
tf.org</a>] On Behalf Of Salz, Rich<br>
Sent: Wednesday, May 3, 2017 5:29 AM<br>
To: Eggert, Lars &lt;<a href=3D"mailto:lars@netapp.com">lars@netapp.com</a>=
&gt;; IETF QUIC WG &lt;<a href=3D"mailto:quic@ietf.org">quic@ietf.org</a>&g=
t;<br>
Subject: RE: Open-source TLS 1.3 with an API suitable for QUIC?<br>
<br>
&gt; wondering which open-source TLS<br>
&gt; implementations are suitable for providing the required TLS 1.3<br>
&gt; functionality over UDP. Specifically, for a C implementation.<br>
<br>
If there are API&#39;s that OpenSSL would need to provide, please let me kn=
ow (ideally within a month) and we will work to provide them.<br>
<br>
</div></div></blockquote></div><br></div></div>

--001a114fc7d8cfcea8054ea0db31--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Thu, 4 May 2017 07:56:59 +1000
Message-ID: <CABkgnnVzG9TopFnxcCtPM2ZCr0rP8Gst3DfJDaVO84do2Z6Pig@mail.gmail.com>
Subject: Re: Updated Public Reset authentication patch
To: Eric Rescorla <ekr@rtfm.com>
Cc: Brian Trammell <ietf@trammell.ch>, QUIC WG <quic@ietf.org>
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On 4 May 2017 at 00:40, Eric Rescorla <ekr@rtfm.com> wrote:
> In EncryptedExtensions, the client sends V = HKDF(K, <conn-id>).

I think that Christian's idea works well enough.  This one requires an
impossibility :)


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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 May 2017 14:59:21 -0700
Message-ID: <CABcZeBNOBywJ6miRQzwGW+CwsqvBJ-6fpBTfe3JPCVheWbrAFQ@mail.gmail.com>
Subject: Re: Updated Public Reset authentication patch
To: Martin Thomson <martin.thomson@gmail.com>
Cc: Brian Trammell <ietf@trammell.ch>, QUIC WG <quic@ietf.org>
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--94eb2c087eea30baae054ea5c721
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On Wed, May 3, 2017 at 2:56 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 4 May 2017 at 00:40, Eric Rescorla <ekr@rtfm.com> wrote:
> > In EncryptedExtensions, the client sends V = HKDF(K, <conn-id>).
>
> I think that Christian's idea works well enough.  This one requires an
> impossibility :)
>

Why is this an impossibility?

-Ekr

--94eb2c087eea30baae054ea5c721
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Wed, May 3, 2017 at 2:56 PM, Martin Thomson <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@=
gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span cl=
ass=3D"">On 4 May 2017 at 00:40, Eric Rescorla &lt;<a href=3D"mailto:ekr@rt=
fm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt; In EncryptedExtensions, the client sends V =3D HKDF(K, &lt;conn-id&gt;=
).<br>
<br>
</span>I think that Christian&#39;s idea works well enough.=C2=A0 This one =
requires an<br>
impossibility :)<br></blockquote><div><br></div><div>Why is this an impossi=
bility?</div><div><br></div><div>-Ekr</div><div>=C2=A0</div></div><br></div=
></div>

--94eb2c087eea30baae054ea5c721--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Thu, 4 May 2017 08:45:57 +1000
Message-ID: <CABkgnnV817-JO8vUodGvXN71cnwKEdfdK_wR7r_S=B15p9bbEA@mail.gmail.com>
Subject: Re: Updated Public Reset authentication patch
To: Eric Rescorla <ekr@rtfm.com>
Cc: Brian Trammell <ietf@trammell.ch>, QUIC WG <quic@ietf.org>
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On 4 May 2017 at 07:59, Eric Rescorla <ekr@rtfm.com> wrote:
> On Wed, May 3, 2017 at 2:56 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>>
>> On 4 May 2017 at 00:40, Eric Rescorla <ekr@rtfm.com> wrote:
>> > In EncryptedExtensions, the client sends V = HKDF(K, <conn-id>).
>>
>> I think that Christian's idea works well enough.  This one requires an
>> impossibility :)
>
>
> Why is this an impossibility?

The client doesn't send EncryptedExtensions.  Maybe you meant exactly
what Christian said and the V is sent by the server?


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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 3 May 2017 15:47:20 -0700
Message-ID: <CABcZeBMyHajDC=5VuY545FAjGKeqJjmZ=XWJKyWsxtAU8U-s4w@mail.gmail.com>
Subject: Re: Updated Public Reset authentication patch
To: Martin Thomson <martin.thomson@gmail.com>
Cc: Brian Trammell <ietf@trammell.ch>, QUIC WG <quic@ietf.org>
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--94eb2c09dc36cfe265054ea6729c
Content-Type: text/plain; charset=UTF-8

Yes, I meant the server.

-Ekr


On Wed, May 3, 2017 at 3:45 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 4 May 2017 at 07:59, Eric Rescorla <ekr@rtfm.com> wrote:
> > On Wed, May 3, 2017 at 2:56 PM, Martin Thomson <martin.thomson@gmail.com
> >
> > wrote:
> >>
> >> On 4 May 2017 at 00:40, Eric Rescorla <ekr@rtfm.com> wrote:
> >> > In EncryptedExtensions, the client sends V = HKDF(K, <conn-id>).
> >>
> >> I think that Christian's idea works well enough.  This one requires an
> >> impossibility :)
> >
> >
> > Why is this an impossibility?
>
> The client doesn't send EncryptedExtensions.  Maybe you meant exactly
> what Christian said and the V is sent by the server?
>

--94eb2c09dc36cfe265054ea6729c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Yes, I meant the server.<div><br></div><div>-Ekr</div><div=
><br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Wed, May 3, 2017 at 3:45 PM, Martin Thomson <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmai=
l.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class=
=3D"">On 4 May 2017 at 07:59, Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.=
com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt; On Wed, May 3, 2017 at 2:56 PM, Martin Thomson &lt;<a href=3D"mailto:m=
artin.thomson@gmail.com">martin.thomson@gmail.com</a>&gt;<br>
&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; On 4 May 2017 at 00:40, Eric Rescorla &lt;<a href=3D"mailto:ekr@rt=
fm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;&gt; &gt; In EncryptedExtensions, the client sends V =3D HKDF(K, &lt;co=
nn-id&gt;).<br>
&gt;&gt;<br>
&gt;&gt; I think that Christian&#39;s idea works well enough.=C2=A0 This on=
e requires an<br>
&gt;&gt; impossibility :)<br>
&gt;<br>
&gt;<br>
&gt; Why is this an impossibility?<br>
<br>
</span>The client doesn&#39;t send EncryptedExtensions.=C2=A0 Maybe you mea=
nt exactly<br>
what Christian said and the V is sent by the server?<br>
</blockquote></div><br></div>

--94eb2c09dc36cfe265054ea6729c--


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Hi,

On 2017-5-3, at 16:47, Mike Bishop <Michael.Bishop@microsoft.com> wrote:
> The biggest one (and I don't know if OpenSSL has it) is the ability to =
get the actual bytes that TLS wants to send, rather than having the =
library own the socket and try to send by itself.

exactly. And - since I'm trying to reduce memory copies - I'd like to =
pass TLS the buffers and have it operate on them, ideally in place.

Lars

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From nobody Thu May  4 00:38:01 2017
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From: Jana Iyengar <jri@google.com>
Date: Thu, 4 May 2017 00:37:56 -0700
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Subject: Re: Proposed changes to cleartext packets
To: Martin Thomson <martin.thomson@gmail.com>
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I commented on the PR, but echoing it here as well -- I think this is a
very nice set of improvements.

On your alt designs: The server has no way of ensuring that the Client
Cleartext packet contains the server's chosen connection ID. A bold client
could start a connection with a random (or if nefarious, a targeted)
connection ID with a Client Cleartext packet, defeating one of the purposes
of server-chosen connection IDs.

I'll argue for two things to change. (i) I think we should make both the
changes you propose -- that a Client Initial packet be sent right after
both Version Negotiation and Stateless Retry packets. (ii) 0-RTT packets
MUST contain the same connection ID as in the client's most recent
handshake packet to make it possible for load balancers to work with them.

With these changes, a server treats all Client Initial packets similarly,
in that it responds to them with a new server-chosen connection ID. Load
balancers treat all Client Initial, Client Cleartext, and 0-RTT packets as
containing a client-chosen connection ID, and other packets as containing
server-chosen connection IDs.

One of the purposes of this change is to enable a server to direct a client
towards a particular server via the connection ID in a Version Negotiation
or a Stateless Retry packet. A server can still do that by generating a
connection ID that it knows will hash to the right server. Basically, load
balancers will use two functions for hashing connection ID -- C() to hash
client-chosen connection IDs and S() for server-chosen ones. On a VN or SR
packet, the server needs to generate and send a connection ID so that
C(connection ID) sends the retrying client to the desired server. This
allows a server to redirect while also allowing the server infrastructure
to not trust that a client is doing the right thing.


On Wed, May 3, 2017 at 3:43 AM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> Issue: https://github.com/quicwg/base-drafts/issues/442
>
> I've heard quite a few concerns about the current design for server
> selection of the connection ID.  The primary concern is that the
> choice of final connection ID is made on the same flight as the server
> handshake messages - the same flight where the server is required to
> commit non-trivial state and computation resources to the connection.
>
> The current design is actually anathema to certain load balancer
> designs.  If the node that receives a packet wants to get rid of it
> before sending the final server flight, the load balancer can no
> longer route statelessly.
>
> The server folks I've spoken to want to pick a connection ID during
> both version negotiation and stateless reject/address validation.
> This allows them to do trivial or at least lightweight processing on
> the node that receives the first packet AND steer the packet to the
> right node.
>
>
> I'm proposing that we move to a special "first client packet" approach
> rather than a "last server handshake packet" approach.  This allows
> servers to identify which packet has a client-selected connection ID
> and create special logic for that.
>
>
> Issue: https://github.com/quicwg/base-drafts/pull/482#
> discussion_r114445062
>
> The second problem is one that arises from stateless rejects.  A few
> people observed that there is a disconnect between what the server
> sends in a stateless reject and what it subsequently sends.
>
> For instance, the server can't generate a stateless reject then send
> subsequent packets with the expected packet numbers.  The reason: it
> might need to generate multiple stateless reject packets in response
> to multiple handshake attempts from a client.  It can't save the
> packet number it chose in the cookie it sends because that would mean
> sending different values for the same octets on stream 0 and clients
> could get confused by that.
>
> Also, it was previously necessary to resort to shenanigans at the
> server after a stateless reject.  The second ClientHello will start at
> a non-zero offset on stream 0, but the server will effectively have
> never received that data.  That means pretending that the data was
> received and delivered already, but then being able to validate the
> offset and back out the change if validation fails.
>
>
> All in all, that's more special handling than is ideal.  Jana proposed
> resetting the transport side of things when this happens.  That means
> resetting the stream 0 offset and not getting bent out of shape when
> the server chooses a non-contiguous packet number.  That seems like a
> reasonable idea, and I'm proposing a new cleartext message type for
> that so that it can be explicit when the server is doing this.  That
> message comes with its own special treatment (it is used for
> HelloRetryRequest only, and HelloRetryRequest has to use it, it has to
> be one packet).
>
> This externalizes some of the server costs for stateless rejects, and
> it adds another special case packet type.  Neither of which I'm
> entirely happy about, but it seems like a reasonable compromise.
>
>
> These are two separate issues, but they are intermingled in annoying
> ways, so I have a single draft PR for both changes:
>
>    https://github.com/quicwg/base-drafts/pull/493
>
> There is a question in the PR about two alternative designs:
>
> 1. Require the client to send an initial packet after version
> negotiation on the basis that version negotiation is a complete
> do-over.
>
> 2. Maybe also after a stateless reject, which would mean that all
> packets containing the ClientHello would use that "initial" type. That
> would mean that only the initial type would have packet size
> restrictions, which is nice.
>
> I think that these could be an improvement, but I'd like to hear from
> server folks.  The cost here is that after version negotiation and
> maybe also a stateless reject, you would route the packet from the
> client as if it were a client-selected connection ID.  I'm not sure
> how much that matters.  What you do get is up to three opportunities
> to reroute a client (during version negotiation, optionally during
> stateless reject, and during the handshake).
>
>

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

<div dir=3D"ltr">I commented on the PR, but echoing it here as well -- I th=
ink this is a very nice set of improvements.<div><br></div><div><div>On you=
r alt designs: The server has no way of ensuring that the Client Cleartext =
packet contains the server&#39;s chosen connection ID. A bold client could =
start a connection with a random (or if nefarious, a targeted) connection I=
D with a Client Cleartext packet, defeating one of the purposes of server-c=
hosen connection IDs.</div><div><br></div><div>I&#39;ll argue for two thing=
s to change. (i) I think we should make both the changes you propose -- tha=
t a Client Initial packet be sent right after both Version Negotiation and =
Stateless Retry packets. (ii) 0-RTT packets MUST contain the same connectio=
n ID as in the client&#39;s most recent handshake packet to make it possibl=
e for load balancers to work with them.</div><div><br></div><div>With these=
 changes, a server treats all Client Initial packets similarly, in that it =
responds to them with a new server-chosen connection ID. Load balancers tre=
at all Client Initial, Client Cleartext, and 0-RTT packets as containing a =
client-chosen connection ID, and other packets as containing server-chosen =
connection IDs.</div><div><br></div><div>One of the purposes of this change=
 is to enable a server to direct a client towards a particular server via t=
he connection ID in a Version Negotiation or a Stateless Retry packet. A se=
rver can still do that by generating a connection ID that it knows will has=
h to the right server. Basically, load balancers will use two functions for=
 hashing connection ID -- C() to hash client-chosen connection IDs and S() =
for server-chosen ones. On a VN or SR packet, the server needs to generate =
and send a connection ID so that C(connection ID) sends the retrying client=
 to the desired server. This allows a server to redirect while also allowin=
g the server infrastructure to not trust that a client is doing the right t=
hing.</div></div><div><br></div></div><div class=3D"gmail_extra"><br><div c=
lass=3D"gmail_quote">On Wed, May 3, 2017 at 3:43 AM, Martin Thomson <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank=
">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex">Issue: <a href=3D"https://github.com/quicwg/base-drafts/issues/442"=
 rel=3D"noreferrer" target=3D"_blank">https://github.com/quicwg/<wbr>base-d=
rafts/issues/442</a><br>
<br>
I&#39;ve heard quite a few concerns about the current design for server<br>
selection of the connection ID.=C2=A0 The primary concern is that the<br>
choice of final connection ID is made on the same flight as the server<br>
handshake messages - the same flight where the server is required to<br>
commit non-trivial state and computation resources to the connection.<br>
<br>
The current design is actually anathema to certain load balancer<br>
designs.=C2=A0 If the node that receives a packet wants to get rid of it<br=
>
before sending the final server flight, the load balancer can no<br>
longer route statelessly.<br>
<br>
The server folks I&#39;ve spoken to want to pick a connection ID during<br>
both version negotiation and stateless reject/address validation.<br>
This allows them to do trivial or at least lightweight processing on<br>
the node that receives the first packet AND steer the packet to the<br>
right node.<br>
<br>
<br>
I&#39;m proposing that we move to a special &quot;first client packet&quot;=
 approach<br>
rather than a &quot;last server handshake packet&quot; approach.=C2=A0 This=
 allows<br>
servers to identify which packet has a client-selected connection ID<br>
and create special logic for that.<br>
<br>
<br>
Issue: <a href=3D"https://github.com/quicwg/base-drafts/pull/482#discussion=
_r114445062" rel=3D"noreferrer" target=3D"_blank">https://github.com/quicwg=
/<wbr>base-drafts/pull/482#<wbr>discussion_r114445062</a><br>
<br>
The second problem is one that arises from stateless rejects.=C2=A0 A few<b=
r>
people observed that there is a disconnect between what the server<br>
sends in a stateless reject and what it subsequently sends.<br>
<br>
For instance, the server can&#39;t generate a stateless reject then send<br=
>
subsequent packets with the expected packet numbers.=C2=A0 The reason: it<b=
r>
might need to generate multiple stateless reject packets in response<br>
to multiple handshake attempts from a client.=C2=A0 It can&#39;t save the<b=
r>
packet number it chose in the cookie it sends because that would mean<br>
sending different values for the same octets on stream 0 and clients<br>
could get confused by that.<br>
<br>
Also, it was previously necessary to resort to shenanigans at the<br>
server after a stateless reject.=C2=A0 The second ClientHello will start at=
<br>
a non-zero offset on stream 0, but the server will effectively have<br>
never received that data.=C2=A0 That means pretending that the data was<br>
received and delivered already, but then being able to validate the<br>
offset and back out the change if validation fails.<br>
<br>
<br>
All in all, that&#39;s more special handling than is ideal.=C2=A0 Jana prop=
osed<br>
resetting the transport side of things when this happens.=C2=A0 That means<=
br>
resetting the stream 0 offset and not getting bent out of shape when<br>
the server chooses a non-contiguous packet number.=C2=A0 That seems like a<=
br>
reasonable idea, and I&#39;m proposing a new cleartext message type for<br>
that so that it can be explicit when the server is doing this.=C2=A0 That<b=
r>
message comes with its own special treatment (it is used for<br>
HelloRetryRequest only, and HelloRetryRequest has to use it, it has to<br>
be one packet).<br>
<br>
This externalizes some of the server costs for stateless rejects, and<br>
it adds another special case packet type.=C2=A0 Neither of which I&#39;m<br=
>
entirely happy about, but it seems like a reasonable compromise.<br>
<br>
<br>
These are two separate issues, but they are intermingled in annoying<br>
ways, so I have a single draft PR for both changes:<br>
<br>
=C2=A0 =C2=A0<a href=3D"https://github.com/quicwg/base-drafts/pull/493" rel=
=3D"noreferrer" target=3D"_blank">https://github.com/quicwg/<wbr>base-draft=
s/pull/493</a><br>
<br>
There is a question in the PR about two alternative designs:<br>
<br>
1. Require the client to send an initial packet after version<br>
negotiation on the basis that version negotiation is a complete<br>
do-over.<br>
<br>
2. Maybe also after a stateless reject, which would mean that all<br>
packets containing the ClientHello would use that &quot;initial&quot; type.=
 That<br>
would mean that only the initial type would have packet size<br>
restrictions, which is nice.<br>
<br>
I think that these could be an improvement, but I&#39;d like to hear from<b=
r>
server folks.=C2=A0 The cost here is that after version negotiation and<br>
maybe also a stateless reject, you would route the packet from the<br>
client as if it were a client-selected connection ID.=C2=A0 I&#39;m not sur=
e<br>
how much that matters.=C2=A0 What you do get is up to three opportunities<b=
r>
to reroute a client (during version negotiation, optionally during<br>
stateless reject, and during the handshake).<br>
<br>
</blockquote></div><br></div>

--001a11467bd2f6cc3d054eadd98b--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Thu, 4 May 2017 19:05:46 +1000
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Subject: Re: Proposed changes to cleartext packets
To: Jana Iyengar <jri@google.com>, Subodh Iyengar <subodh@fb.com>,  "Lubashev, Igor" <ilubashe@akamai.com>, Kazuho Oku <kazuhooku@gmail.com>
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On 4 May 2017 at 17:37, Jana Iyengar <jri@google.com> wrote:
> One of the purposes of this change is to enable a server to direct a client
> towards a particular server via the connection ID in a Version Negotiation
> or a Stateless Retry packet. A server can still do that by generating a
> connection ID that it knows will hash to the right server. Basically, load
> balancers will use two functions for hashing connection ID -- C() to hash
> client-chosen connection IDs and S() for server-chosen ones. On a VN or SR
> packet, the server needs to generate and send a connection ID so that
> C(connection ID) sends the retrying client to the desired server. This
> allows a server to redirect while also allowing the server infrastructure to
> not trust that a client is doing the right thing.

This convinced me.  I'd like to hear from Kazuho, Igor, and Subodh
about whether this works for them.  I don't see why it wouldn't, but I
don't know their infrastructure.


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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: Proposed changes to cleartext packets
To: Jana Iyengar <jri@google.com>, Subodh Iyengar <subodh@fb.com>,  "Lubashev, Igor" <ilubashe@akamai.com>, Kazuho Oku <kazuhooku@gmail.com>
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I have written up the proposed change and updated the PR.

   https://github.com/quicwg/base-drafts/pull/493

>From my commit message, here's the summary:

* Client Initial packets are always a complete reset at the transport
layer, always contain a ClientHello, are always padded to the minimum
size, and include a randomized packet number unless they are a
retransmission.  STREAM frames in these packets always start at an
offset of 0, even if the cryptographic handshake is continued.  All
necessary data has to fit into a single packet; it can't span several.
The server can propose a new connection ID in response to Client
Initial packets (giving the server up to three re-routing options).

* Server Stateless Retry packets cause the client to start over,
retaining only cryptographic handshake state.  The contents of this
also can't span two packets.  This includes a server-proposed
connection ID and an echo of the client packet number.  This is
because the server can't be expected to remember what it chose when it
sends subsequent packets.

* Version Negotiation packets also cause a reset.  This has the same
rules for connection ID and packet number as the Server Stateless
Retry.

* Client Cleartext and Server Cleartext contain the other handshake
messages. These aren't really special in any way.

On 4 May 2017 at 19:05, Martin Thomson <martin.thomson@gmail.com> wrote:
> On 4 May 2017 at 17:37, Jana Iyengar <jri@google.com> wrote:
>> One of the purposes of this change is to enable a server to direct a client
>> towards a particular server via the connection ID in a Version Negotiation
>> or a Stateless Retry packet. A server can still do that by generating a
>> connection ID that it knows will hash to the right server. Basically, load
>> balancers will use two functions for hashing connection ID -- C() to hash
>> client-chosen connection IDs and S() for server-chosen ones. On a VN or SR
>> packet, the server needs to generate and send a connection ID so that
>> C(connection ID) sends the retrying client to the desired server. This
>> allows a server to redirect while also allowing the server infrastructure to
>> not trust that a client is doing the right thing.
>
> This convinced me.  I'd like to hear from Kazuho, Igor, and Subodh
> about whether this works for them.  I don't see why it wouldn't, but I
> don't know their infrastructure.


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Subject: Re: Proposed changes to cleartext packets
To: Martin Thomson <martin.thomson@gmail.com>, Jana Iyengar <jri@google.com>,  Subodh Iyengar <subodh@fb.com>, "Lubashev, Igor" <ilubashe@akamai.com>, Kazuho Oku <kazuhooku@gmail.com>
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From: =?UTF-8?Q?Mirja_K=c3=bchlewind?= <mirja.kuehlewind@tik.ee.ethz.ch>
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Quick question, and maybe I lost the context here, but

On 04.05.2017 13:04, Martin Thomson wrote:
> * Client Initial packets are always a complete reset at the transport
> layer, always contain a ClientHello, are always padded to the minimum
> size, and include a randomized packet number unless they are a
> retransmission.

how can you do a retransmit if you cleared all transport state?

Mirja


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From: Ted Hardie <ted.ietf@gmail.com>
Date: Thu, 4 May 2017 13:49:02 -0700
Message-ID: <CA+9kkMBh9K5=bSJRNnLAnMwWeLFYKGQ6SJoMnb4ZoihFbvTq=Q@mail.gmail.com>
Subject: Re: Proposed changes to cleartext packets
To: Jana Iyengar <jri@google.com>
Cc: Martin Thomson <martin.thomson@gmail.com>, QUIC WG <quic@ietf.org>
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On Thu, May 4, 2017 at 12:37 AM, Jana Iyengar <jri@google.com> wrote:

> (ii) 0-RTT packets MUST contain the same connection ID as in the client's
> most recent handshake packet to make it possible for load balancers to work
> with them.
>

I may be misunderstanding something, but I don't see the advantage of using
the same connection ID as in the most recent handshake packet over using
any of the Connection IDs that a server has provided (presuming that the
change that allows a list to be returned for use in connection migration
goes forward).  Presumably, any of the server supplied will work through
load balancers to reach the right server (or the server is implicitly okay
with stateless operation).

To me, at least, it would make for easier client logic if it always used
the next one in the list whether it was migrating a connection because of a
known network change or using 0-RTT because of a period where the transport
wasn't up.  It also might improve privacy, if linking flow state across
0-RTT restarts is a concern.

Am I missing a harm in using one of the later connection IDs, if they were
supplied?

regards,

Ted

--94eb2c033d78f703a3054eb8e8d3
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<div dir=3D"ltr">On Thu, May 4, 2017 at 12:37 AM, Jana Iyengar <span dir=3D=
"ltr">&lt;<a href=3D"mailto:jri@google.com" target=3D"_blank">jri@google.co=
m</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div class=3D"gmail_q=
uote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">(ii) 0-RTT packets MU=
ST contain the same connection ID as in the client&#39;s most recent handsh=
ake packet to make it possible for load balancers to work with them.<br></d=
iv></blockquote></div><br></div><div class=3D"gmail_extra">I may be misunde=
rstanding something, but I don&#39;t see the advantage of using the same co=
nnection ID as in the most recent handshake packet over using any of the Co=
nnection IDs that a server has provided (presuming that the change that all=
ows a list to be returned for use in connection migration goes forward).=C2=
=A0 Presumably, any of the server supplied will work through load balancers=
 to reach the right server (or the server is implicitly okay with stateless=
 operation).<br><br></div><div class=3D"gmail_extra">To me, at least, it wo=
uld make for easier client logic if it always used the next one in the list=
 whether it was migrating a connection because of a known network change or=
 using 0-RTT because of a period where the transport wasn&#39;t up.=C2=A0 I=
t also might improve privacy, if linking flow state across 0-RTT restarts i=
s a concern.<br><br></div><div class=3D"gmail_extra">Am I missing a harm in=
 using one of the later connection IDs, if they were supplied?<br><br></div=
><div class=3D"gmail_extra">regards,<br><br></div><div class=3D"gmail_extra=
">Ted<br></div><div class=3D"gmail_extra"><br></div></div>

--94eb2c033d78f703a3054eb8e8d3--


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From: Jana Iyengar <jri@google.com>
Date: Thu, 4 May 2017 16:23:02 -0700
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Subject: Re: Proposed changes to cleartext packets
To: Ted Hardie <ted.ietf@gmail.com>
Cc: Martin Thomson <martin.thomson@gmail.com>, QUIC WG <quic@ietf.org>
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After some discussion with Ryan and some thinking, I'm going to end up
arguing against my previous self (whee!). I'll also suggest a few
modifications.

1. Sending server-proposed connection ID on only Stateless Retry:

I don't think a new connection ID on a Version Negotiation (VN) packet is
particularly useful. Specifically, the issue that I believe this was
expected to solve (Subodh and/or Kazuho should confirm) was if a server
does not support the client's requested version but knows of a different
server that does, it should be able to redirect the client to the correct
server by sending an appropriate connection ID on the VN packet. This
cannot work because the VN packet would contain the client's chosen version
and would therefore be ignored by the client (as it should be.) A server
can send a Stateless Retry (SR) packet with a server-chosen connection ID
when shedding load to a different server. If this is the only case that
needs supporting, a client's use of the server-chosen ID in a subsequent
Client Cleartext packet can be verified by simply adding the server-chosen
ID in the cookie that is sent with the Retry packet.

2. Co-ordinating connection ID change mechanisms:

This mechanism (and the previous one in the current spec) have both the
server and the client changing connection IDs -- the server does it during
handshake and the client does it after it receives one or more
NEW_CONNECTION_ID frames. I think we can make this simpler (both in
protocol and impl) with only the client initiating this change by sending a
packet with a server-chosen connection ID first. If the client were the
only one initiating the change, this would allow the server to not care
about doing any changes mid-connection, and it could simply use the
connection ID in the largest received packet as the outgoing one.

This implies that the server somehow signals the new connection ID to the
client during handshake, which the client MUST use on all 1-RTT encrypted
packets (since we are designing for a server-side load balancer, we can
allow the server to send 1-RTT encrypted packets with the client-chosen
connection IDs to accomodate the 0.5RTT case.) We can achieve this
signaling in one of two ways:
(ii) The server sends a NEW_CONNECTION_ID frame bundled in the ServerHello
packet, but there are corner cases with and without bundling that might
make it tricky. Additionally, the NEW_CONNECTION_ID frame sent along with
the ServerHello could just be the general one sent for a migration event,
and this would be ambiguous signaling. We could fix that by adding a bit
next to each connection ID, but then this bit is unused for all but the
first connection ID.
(i) The server sends a new connection ID as a transport param in the
ServerHello, instead of splitting NEW_CONNECTION_ID frame elements into two
types. NEW_CONNECTION_ID continues to carry IDs for future use.

Explicitly indicating the new connection ID gives the client an explicit
correlator, which is better than assuming that the 4-tuple will only carry
packets for this connection. Since the server also needs to convey this new
connection ID in a Stateless Retry, an SR packet could carry a
NEW_CONNECTION_ID frame.

This leaves us with two protocol mechanisms for carrying a new connection
ID: transport param and NEW_CONNECTION_ID. If we ditch the transport param,
then it's two types of connection IDs within a NEW_CONNECTION_ID frame,
which seems equivalent.

A nice side-effect is that a server that is happy with client-chosen
connection IDs simply does not send this param / frame during handshake.

If this discussion above makes sense, we would need to:
(i) stick with the packet types in the previous version of this patch, and
(ii) decide and add a mechanism for conveying connection ID to client on
handshake.

Thoughts?

On Thu, May 4, 2017 at 1:49 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> On Thu, May 4, 2017 at 12:37 AM, Jana Iyengar <jri@google.com> wrote:
>
>> (ii) 0-RTT packets MUST contain the same connection ID as in the client's
>> most recent handshake packet to make it possible for load balancers to work
>> with them.
>>
>
> I may be misunderstanding something, but I don't see the advantage of
> using the same connection ID as in the most recent handshake packet over
> using any of the Connection IDs that a server has provided (presuming that
> the change that allows a list to be returned for use in connection
> migration goes forward).  Presumably, any of the server supplied will work
> through load balancers to reach the right server (or the server is
> implicitly okay with stateless operation).
>
> To me, at least, it would make for easier client logic if it always used
> the next one in the list whether it was migrating a connection because of a
> known network change or using 0-RTT because of a period where the transport
> wasn't up.  It also might improve privacy, if linking flow state across
> 0-RTT restarts is a concern.
>
> Am I missing a harm in using one of the later connection IDs, if they were
> supplied?
>
> regards,
>
> Ted
>
>

--001a114d7d58ea43be054ebb0d71
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">After some discussion with Ryan and some thinking, I&#39;m=
 going to end up arguing against my previous self (whee!). I&#39;ll also su=
ggest a few modifications.<div><br></div><div>1. Sending server-proposed co=
nnection ID on only Stateless Retry:=C2=A0</div><div><br></div><div>I don&#=
39;t think a new connection ID on a Version Negotiation (VN) packet is part=
icularly useful. Specifically, the issue that I believe this was expected t=
o solve (Subodh and/or Kazuho should confirm) was if a server does not supp=
ort the client&#39;s requested version but knows of a different server that=
 does, it should be able to redirect the client to the correct server by se=
nding an appropriate connection ID on the VN packet. This cannot work becau=
se the VN packet would contain the client&#39;s chosen version and would th=
erefore be ignored by the client (as it should be.) A server can send a Sta=
teless Retry (SR) packet with a server-chosen connection ID when shedding l=
oad to a different server. If this is the only case that needs supporting, =
a client&#39;s use of the server-chosen ID in a subsequent Client Cleartext=
 packet can be verified by simply adding the server-chosen ID in the cookie=
 that is sent with the Retry packet.=C2=A0</div><div><br></div><div>2. Co-o=
rdinating connection ID change mechanisms:=C2=A0</div><div><br></div><div>T=
his mechanism (and the previous one in the current spec) have both the serv=
er and the client changing connection IDs -- the server does it during hand=
shake and the client does it after it receives one or more NEW_CONNECTION_I=
D frames. I think we can make this simpler (both in protocol and impl) with=
 only the client initiating this change by sending a packet with a server-c=
hosen connection ID first. If the client were the only one initiating the c=
hange, this would allow the server to not care about doing any changes mid-=
connection, and it could simply use the connection ID in the largest receiv=
ed packet as the outgoing one.<br><br>This implies that the server somehow =
signals the new connection ID to the client during handshake, which the cli=
ent MUST use on all 1-RTT encrypted packets (since we are designing for a s=
erver-side load balancer, we can allow the server to send 1-RTT encrypted p=
ackets with the client-chosen connection IDs to accomodate the 0.5RTT case.=
) We can achieve this signaling in one of two ways:</div><div>(ii) The serv=
er sends a NEW_CONNECTION_ID frame bundled in the ServerHello packet, but t=
here are corner cases with and without bundling that might make it tricky. =
Additionally, the NEW_CONNECTION_ID frame sent along with the ServerHello c=
ould just be the general one sent for a migration event, and this would be =
ambiguous signaling. We could fix that by adding a bit next to each connect=
ion ID, but then this bit is unused for all but the first connection ID.</d=
iv><div></div><div>(i) The server sends a new connection ID as a transport =
param in the ServerHello, instead of splitting NEW_CONNECTION_ID frame elem=
ents into two types. NEW_CONNECTION_ID continues to carry IDs for future us=
e.</div><div><br></div><div>Explicitly indicating the new connection ID giv=
es the client an explicit correlator, which is better than assuming that th=
e 4-tuple will only carry packets for this connection. Since the server als=
o needs to convey this new connection ID in a Stateless Retry, an SR packet=
 could carry a NEW_CONNECTION_ID frame.</div><div><br></div><div>This leave=
s us with two protocol mechanisms for carrying a new connection ID: transpo=
rt param and NEW_CONNECTION_ID. If we ditch the transport param, then it&#3=
9;s two types of connection IDs within a NEW_CONNECTION_ID frame, which see=
ms equivalent.</div><div><br></div><div>A nice side-effect is that a server=
 that is happy with client-chosen connection IDs simply does not send this =
param / frame during handshake.</div><div><br></div><div>If this discussion=
 above makes sense, we would need to:</div><div>(i) stick with the packet t=
ypes in the previous version of this patch, and</div><div>(ii) decide and a=
dd a mechanism for conveying connection ID to client on handshake.</div><di=
v><br></div><div>Thoughts?</div></div><div class=3D"gmail_extra"><br><div c=
lass=3D"gmail_quote">On Thu, May 4, 2017 at 1:49 PM, Ted Hardie <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ie=
tf@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div d=
ir=3D"ltr"><span class=3D"">On Thu, May 4, 2017 at 12:37 AM, Jana Iyengar <=
span dir=3D"ltr">&lt;<a href=3D"mailto:jri@google.com" target=3D"_blank">jr=
i@google.com</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">(ii) 0-RTT=
 packets MUST contain the same connection ID as in the client&#39;s most re=
cent handshake packet to make it possible for load balancers to work with t=
hem.<br></div></blockquote></div><br></div></span><div class=3D"gmail_extra=
">I may be misunderstanding something, but I don&#39;t see the advantage of=
 using the same connection ID as in the most recent handshake packet over u=
sing any of the Connection IDs that a server has provided (presuming that t=
he change that allows a list to be returned for use in connection migration=
 goes forward).=C2=A0 Presumably, any of the server supplied will work thro=
ugh load balancers to reach the right server (or the server is implicitly o=
kay with stateless operation).<br><br></div><div class=3D"gmail_extra">To m=
e, at least, it would make for easier client logic if it always used the ne=
xt one in the list whether it was migrating a connection because of a known=
 network change or using 0-RTT because of a period where the transport wasn=
&#39;t up.=C2=A0 It also might improve privacy, if linking flow state acros=
s 0-RTT restarts is a concern.<br><br></div><div class=3D"gmail_extra">Am I=
 missing a harm in using one of the later connection IDs, if they were supp=
lied?<br><br></div><div class=3D"gmail_extra">regards,<br><br></div><div cl=
ass=3D"gmail_extra">Ted<br></div><div class=3D"gmail_extra"><br></div></div=
>
</blockquote></div><br></div>

--001a114d7d58ea43be054ebb0d71--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Fri, 5 May 2017 09:30:16 +1000
Message-ID: <CABkgnnXAvkFT315o2cRhks++3D_FFRvvCe1qRjiBV1E6Gw=OTQ@mail.gmail.com>
Subject: Re: Proposed changes to cleartext packets
To: =?UTF-8?Q?Mirja_K=C3=BChlewind?= <mirja.kuehlewind@tik.ee.ethz.ch>
Cc: Jana Iyengar <jri@google.com>, Subodh Iyengar <subodh@fb.com>,  "Lubashev, Igor" <ilubashe@akamai.com>, Kazuho Oku <kazuhooku@gmail.com>, QUIC WG <quic@ietf.org>
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On 5 May 2017 at 03:23, Mirja K=C3=BChlewind <mirja.kuehlewind@tik.ee.ethz.=
ch> wrote:
> how can you do a retransmit if you cleared all transport state?


The idea here is that you reset transport state when you receive
Version Negotiation or Server Stateless Retry.  When you send the next
Client Initial packet, that needs to be retransmitted.

(Note that you can't reset absolutely everything.  You probably want
to keep the RTT estimate and any congestion signals you have managed
to eke out.)


From nobody Thu May  4 16:41:48 2017
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From: Jana Iyengar <jri@google.com>
Date: Thu, 4 May 2017 16:41:43 -0700
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Subject: Re: Proposed changes to cleartext packets
To: Ted Hardie <ted.ietf@gmail.com>
Cc: Martin Thomson <martin.thomson@gmail.com>, QUIC WG <quic@ietf.org>
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--94eb2c14896cbab2fb054ebb5097
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On Thu, May 4, 2017 at 1:49 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> On Thu, May 4, 2017 at 12:37 AM, Jana Iyengar <jri@google.com> wrote:
>
>> (ii) 0-RTT packets MUST contain the same connection ID as in the client's
>> most recent handshake packet to make it possible for load balancers to work
>> with them.
>>
>
> I may be misunderstanding something, but I don't see the advantage of
> using the same connection ID as in the most recent handshake packet over
> using any of the Connection IDs that a server has provided (presuming that
> the change that allows a list to be returned for use in connection
> migration goes forward).  Presumably, any of the server supplied will work
> through load balancers to reach the right server (or the server is
> implicitly okay with stateless operation).
>
> To me, at least, it would make for easier client logic if it always used
> the next one in the list whether it was migrating a connection because of a
> known network change or using 0-RTT because of a period where the transport
> wasn't up.  It also might improve privacy, if linking flow state across
> 0-RTT restarts is a concern.
>
> Am I missing a harm in using one of the later connection IDs, if they were
> supplied?
>

This design is only for the transition from a client-chosen to a
server-chosen connection ID, and not for subsequent ID changes... does that
make sense? The requirement you highlighted accomodates load balancers that
use different hashing algorithms on client-chosen vs server-chosen
connection IDs. For instance, a load balancer might use simple ECMP hashing
of the client-chosen connection IDs and treat server-chosen connection IDs
as containing structure or routing information. Changing the connection ID
on an externally visible transition allows such load balancers to use the
different hashing algorithms correctly on the connection IDs.

- jana



> regards,
>
> Ted
>
>

--94eb2c14896cbab2fb054ebb5097
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, May 4, 2017 at 1:49 PM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span>=
 wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span class=3D""=
>On Thu, May 4, 2017 at 12:37 AM, Jana Iyengar <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:jri@google.com" target=3D"_blank">jri@google.com</a>&gt;</span>=
 wrote:<br><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><div dir=3D"ltr">(ii) 0-RTT packets MUST contain the s=
ame connection ID as in the client&#39;s most recent handshake packet to ma=
ke it possible for load balancers to work with them.<br></div></blockquote>=
</div><br></div></span><div class=3D"gmail_extra">I may be misunderstanding=
 something, but I don&#39;t see the advantage of using the same connection =
ID as in the most recent handshake packet over using any of the Connection =
IDs that a server has provided (presuming that the change that allows a lis=
t to be returned for use in connection migration goes forward).=C2=A0 Presu=
mably, any of the server supplied will work through load balancers to reach=
 the right server (or the server is implicitly okay with stateless operatio=
n).<br><br></div><div class=3D"gmail_extra">To me, at least, it would make =
for easier client logic if it always used the next one in the list whether =
it was migrating a connection because of a known network change or using 0-=
RTT because of a period where the transport wasn&#39;t up.=C2=A0 It also mi=
ght improve privacy, if linking flow state across 0-RTT restarts is a conce=
rn.<br><br></div><div class=3D"gmail_extra">Am I missing a harm in using on=
e of the later connection IDs, if they were supplied?<br></div></div></bloc=
kquote><div><br></div><div>This design is only for the transition from a cl=
ient-chosen to a server-chosen connection ID, and not for subsequent ID cha=
nges... does that make sense? The requirement you highlighted accomodates l=
oad balancers that use different hashing algorithms on client-chosen vs ser=
ver-chosen connection IDs. For instance, a load balancer might use simple E=
CMP hashing of the client-chosen connection IDs and treat server-chosen con=
nection IDs as containing structure or routing information. Changing the co=
nnection ID on an externally visible transition allows such load balancers =
to use the different hashing algorithms correctly on the connection IDs.=C2=
=A0</div><div><br></div><div>- jana</div><div><br></div><div>=C2=A0</div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"></=
div><div class=3D"gmail_extra">regards,<br><br></div><div class=3D"gmail_ex=
tra">Ted<br></div><div class=3D"gmail_extra"><br></div></div>
</blockquote></div><br></div></div>

--94eb2c14896cbab2fb054ebb5097--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Fri, 5 May 2017 10:09:19 +1000
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Subject: Re: Proposed changes to cleartext packets
To: Jana Iyengar <jri@google.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, QUIC WG <quic@ietf.org>
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I don't think that either suggestion works particularly well.  I'll
explain below.

On 5 May 2017 at 09:23, Jana Iyengar <jri@google.com> wrote:
> I don't think a new connection ID on a Version Negotiation (VN) packet is
> particularly useful. Specifically, the issue that I believe this was
> expected to solve (Subodh and/or Kazuho should confirm) was if a server does
> not support the client's requested version but knows of a different server
> that does, it should be able to redirect the client to the correct server by
> sending an appropriate connection ID on the VN packet.

I believe that this was one of the cases that was identified.  And you
are correct that a stateless reject is the only way to do what was
intended here.  Either the version negotiation packet includes the
client's chosen version and is ignored, or it doesn't match the
version that the other server will ultimately advertise.  BTW, I've
opened an issue for documenting how a server cluster might be upgraded
(and rolled back): https://github.com/quicwg/base-drafts/issues/504

The use case you are missing is one where the server receives an
unsupported version AND it wants to send the client to another server.
One reason you might want to do that is to avoid a second round trip.
I'll concede that is a little unlikely given the potential need for
address validation, so it's a pretty minor gain, but a gain
nonetheless.

Can you demonstrate something that we would lose by allowing the
server to propose a connection ID here?  The gain might be small, but
I'm not seeing any downside to allowing it.

> This mechanism (and the previous one in the current spec) have both the
> server and the client changing connection IDs -- the server does it during
> handshake and the client does it after it receives one or more
> NEW_CONNECTION_ID frames. I think we can make this simpler (both in protocol
> and impl) with only the client initiating this change by sending a packet
> with a server-chosen connection ID first.

Let me try to restate your proposal.

You would have the entire handshake complete with the same connection
ID, then switch to a different connection ID once the handshake
completes.  A stateless reject would be allowed to cause re-routing.

I'm not sure that I understand the advantages that this design has.
You claim that it reduces the number of mechanisms to two, but I think
that you end up with three because you need to use a transport
parameter.  (FWIW, the current writeup essentially only has two
mechanisms: NEW_CONNECTION_ID and server handshake packets, though
I'll concede that there might be three of those).  It has essentially
the same properties as what I've written up, but it's more complicated
in several ways.

The NEW_CONNECTION_ID frame implies a packet number gap.  With client
authentication, the server might receive encrypted packets before the
handshake completes.  If we take ekr's pull request for
NEW_CONNECTION_ID, the server can't know what the gap is without
receiving the handshake packets.

This would create more exceptions for what can be sent in the clear,
which requires more care.  If, as proposed, we allow multiple new
connection IDs to be sent in the same frame, we wouldn't want that to
happen here, so that's another new special case to test.

That suggests a transport parameter might be a better design, but we
don't send transport parameters in the HelloRetryRequest.  We'd lose
confidentiality for those unless we created a new set of rules for
what transport parameters can be sent in a HelloRetryRequest.  That
generates more special rules.

I think that you raise an important point here:

> Changing the connection ID on an externally visible transition allows such load balancers to use the different hashing algorithms correctly on the connection IDs.

The current proposal has equally simple rules:

if packet[0] == 0x82 or packet[0] == 0x86:
  routing = client_selected
else:
  routing = server_selected

That is, Client Initial and 0-RTT get routing as though the client chose the ID.

You would instead have us switch the test to packet[0] & 0x80 == 0x80
and packet[0] != 0x87 and packet[0] != 0x88, that is every long form
packet except the 1-RTT ones.  I don't see any advantage there.


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From: Martin Duke <martin.h.duke@gmail.com>
Date: Thu, 4 May 2017 20:16:58 -0700
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Subject: Re: Updated Public Reset authentication patch
To: Eric Rescorla <ekr@rtfm.com>
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--f4030435b5308f0e60054ebe5280
Content-Type: text/plain; charset=UTF-8

I've finally been able to process this thread. This problem has so many
dimensions that we should definitely discuss in Paris.

TL;DR I think signals to middleboxes are important, and I endorse MT's
mechanism but with the proof sent in the clear. I have concerns about how
this might work without connection IDs or with a changing connection ID, if
some of the MAYs and SHOULDs aren't observed, client-generated public
resets, etc. Once we agree on general approach we can interrogate those
issues.

In my opinion, the logic of this signal extends to also doing #353 and
replacing CONNECTION_CLOSE with a public reset.

*************

There are numerous different tradeoffs we discuss above, but reasoning from
first principles about middleboxes may answer all of these questions. I
propose the following assertions:

1. Middleboxes will do stuff to connections that requires state.

2. There will have to be some sort of idle timeout to clean up that state,
because stuff happens.

3. It is desirable to QUIC endpoints that those timeouts be as long as
possible. Premature state teardown will, at the very least, mess up
whatever the middlebox is trying to optimize, and at worst it will break
the connection (e.g. a firewall may not accept packets if it hasn't seen
the Client Hello.)

4. Middlebox vendors and operators will institute longer timeouts if they
have a *reasonable chance* of being gracefully notified of connection close.

5. On-path observers should not be able to generate (spoof) valid public
resets to middleboxes.

>From these points, we should strive to maximize the likelihood of the
middlebox seeing and validating the public reset, but it's OK if it misses
it in certain corner cases. I propose that MT's mechanism, with the proof
sent in the clear and extended to events where Google-QUIC sends
CONNECTION_CLOSE, meets this test. It's true that path changes and path
failure will prevent delivery of this signal. But this is not terribly
different from the balance of events that we see with TCP.

As for an explicit "path close, connection still open" message, I wouldn't
object to this but I don't see it as necessary, nor do I think it likely
that endpoints would bother to send them.

Martin Duke

On Wed, May 3, 2017 at 3:47 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> Yes, I meant the server.
>
> -Ekr
>
>
> On Wed, May 3, 2017 at 3:45 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
>> On 4 May 2017 at 07:59, Eric Rescorla <ekr@rtfm.com> wrote:
>> > On Wed, May 3, 2017 at 2:56 PM, Martin Thomson <
>> martin.thomson@gmail.com>
>> > wrote:
>> >>
>> >> On 4 May 2017 at 00:40, Eric Rescorla <ekr@rtfm.com> wrote:
>> >> > In EncryptedExtensions, the client sends V = HKDF(K, <conn-id>).
>> >>
>> >> I think that Christian's idea works well enough.  This one requires an
>> >> impossibility :)
>> >
>> >
>> > Why is this an impossibility?
>>
>> The client doesn't send EncryptedExtensions.  Maybe you meant exactly
>> what Christian said and the V is sent by the server?
>>
>
>

--f4030435b5308f0e60054ebe5280
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>I&#39;ve finally been able to process this thread. Th=
is problem has so many dimensions that we should definitely discuss in Pari=
s.</div><div><br></div><div>TL;DR I think signals to middleboxes are import=
ant, and I endorse MT&#39;s mechanism but with the proof sent in=C2=A0the c=
lear. I have concerns about how this might work without connection IDs or w=
ith a changing connection ID, if some of the MAYs and SHOULDs aren&#39;t ob=
served, client-generated public resets, etc. Once we agree on general appro=
ach we can interrogate those issues.</div><div><br></div><div>In my opinion=
, the logic of this signal extends to also doing #353 and replacing CONNECT=
ION_CLOSE with a public reset.</div><div><br></div><div>*************</div>=
<div><br></div><div>There are numerous different tradeoffs we discuss above=
, but reasoning from first principles about middleboxes may answer all of t=
hese questions. I propose the following assertions:</div><div><br></div><di=
v>1. Middleboxes will do stuff to connections that requires state.</div><di=
v><br></div><div>2.=C2=A0There will have to be some sort of idle timeout to=
 clean up that state, because stuff happens.</div><div><br></div><div>3. It=
 is desirable to QUIC endpoints that those timeouts be as long as possible.=
 Premature state teardown=C2=A0will, at the very least, mess up whatever th=
e middlebox is trying to optimize, and at worst it will break the connectio=
n (e.g. a firewall may not accept packets if it hasn&#39;t seen the Client =
Hello.)</div><div><br></div><div>4. Middlebox vendors and operators will in=
stitute longer timeouts if they have a *reasonable chance* of being gracefu=
lly notified of connection close.</div><div><br></div><div>5. On-path obser=
vers should not be able to generate (spoof) valid public resets to middlebo=
xes.</div><div><br></div><div>From these points, we should strive to maximi=
ze the likelihood of the middlebox=C2=A0seeing and=C2=A0validating the publ=
ic reset, but it&#39;s OK if it misses it in certain corner cases. I propos=
e that MT&#39;s mechanism, with the proof sent in the clear and extended to=
 events where Google-QUIC sends CONNECTION_CLOSE, meets this test. It&#39;s=
 true that path changes and path failure will=C2=A0prevent delivery of this=
 signal. But this is not terribly different from the balance of events that=
 we see with TCP.</div><div><br></div><div>As for an explicit &quot;path cl=
ose, connection still open&quot; message, I wouldn&#39;t object to this but=
 I don&#39;t see it as necessary, nor do I think it likely that endpoints w=
ould bother to send them.</div><div><br></div><div>Martin Duke</div></div><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, May 3, 201=
7 at 3:47 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtf=
m.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex"><div dir=3D"ltr">Yes, I meant the server.<div><br></div>=
<div>-Ekr</div><div><br></div></div><div class=3D"HOEnZb"><div class=3D"h5"=
><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, May 3, 2=
017 at 3:45 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:mart=
in.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</s=
pan> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex"><span>On 4 May 2017 at 07:59,=
 Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rt=
fm.com</a>&gt; wrote:<br>
&gt; On Wed, May 3, 2017 at 2:56 PM, Martin Thomson &lt;<a href=3D"mailto:m=
artin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;=
<br>
&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; On 4 May 2017 at 00:40, Eric Rescorla &lt;<a href=3D"mailto:ekr@rt=
fm.com" target=3D"_blank">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;&gt; &gt; In EncryptedExtensions, the client sends V =3D HKDF(K, &lt;co=
nn-id&gt;).<br>
&gt;&gt;<br>
&gt;&gt; I think that Christian&#39;s idea works well enough.=C2=A0 This on=
e requires an<br>
&gt;&gt; impossibility :)<br>
&gt;<br>
&gt;<br>
&gt; Why is this an impossibility?<br>
<br>
</span>The client doesn&#39;t send EncryptedExtensions.=C2=A0 Maybe you mea=
nt exactly<br>
what Christian said and the V is sent by the server?<br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--f4030435b5308f0e60054ebe5280--


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From: Martin Duke <martin.h.duke@gmail.com>
Date: Fri, 5 May 2017 10:53:50 -0700
Message-ID: <CAM4esxQvtcLD60WQj6qBjH0BCJr7Ou4swApmaFy4puCNnjCHgA@mail.gmail.com>
Subject: Re: Updated Public Reset authentication patch
To: Eric Rescorla <ekr@rtfm.com>
Cc: Martin Thomson <martin.thomson@gmail.com>, Brian Trammell <ietf@trammell.ch>, QUIC WG <quic@ietf.org>
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--001a113d164e793c66054eca92e2
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Reading through again, I think what I'm actually arguing for is ekr's PR
#20 instead, since a verifier in the clear is the key distinction between
the two as far as I can tell.

On Thu, May 4, 2017 at 8:16 PM, Martin Duke <martin.h.duke@gmail.com> wrote:

> I've finally been able to process this thread. This problem has so many
> dimensions that we should definitely discuss in Paris.
>
> TL;DR I think signals to middleboxes are important, and I endorse MT's
> mechanism but with the proof sent in the clear. I have concerns about how
> this might work without connection IDs or with a changing connection ID, if
> some of the MAYs and SHOULDs aren't observed, client-generated public
> resets, etc. Once we agree on general approach we can interrogate those
> issues.
>
> In my opinion, the logic of this signal extends to also doing #353 and
> replacing CONNECTION_CLOSE with a public reset.
>
> *************
>
> There are numerous different tradeoffs we discuss above, but reasoning
> from first principles about middleboxes may answer all of these questions.
> I propose the following assertions:
>
> 1. Middleboxes will do stuff to connections that requires state.
>
> 2. There will have to be some sort of idle timeout to clean up that state,
> because stuff happens.
>
> 3. It is desirable to QUIC endpoints that those timeouts be as long as
> possible. Premature state teardown will, at the very least, mess up
> whatever the middlebox is trying to optimize, and at worst it will break
> the connection (e.g. a firewall may not accept packets if it hasn't seen
> the Client Hello.)
>
> 4. Middlebox vendors and operators will institute longer timeouts if they
> have a *reasonable chance* of being gracefully notified of connection close.
>
> 5. On-path observers should not be able to generate (spoof) valid public
> resets to middleboxes.
>
> From these points, we should strive to maximize the likelihood of the
> middlebox seeing and validating the public reset, but it's OK if it misses
> it in certain corner cases. I propose that MT's mechanism, with the proof
> sent in the clear and extended to events where Google-QUIC sends
> CONNECTION_CLOSE, meets this test. It's true that path changes and path
> failure will prevent delivery of this signal. But this is not terribly
> different from the balance of events that we see with TCP.
>
> As for an explicit "path close, connection still open" message, I wouldn't
> object to this but I don't see it as necessary, nor do I think it likely
> that endpoints would bother to send them.
>
> Martin Duke
>
> On Wed, May 3, 2017 at 3:47 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>> Yes, I meant the server.
>>
>> -Ekr
>>
>>
>> On Wed, May 3, 2017 at 3:45 PM, Martin Thomson <martin.thomson@gmail.com>
>> wrote:
>>
>>> On 4 May 2017 at 07:59, Eric Rescorla <ekr@rtfm.com> wrote:
>>> > On Wed, May 3, 2017 at 2:56 PM, Martin Thomson <
>>> martin.thomson@gmail.com>
>>> > wrote:
>>> >>
>>> >> On 4 May 2017 at 00:40, Eric Rescorla <ekr@rtfm.com> wrote:
>>> >> > In EncryptedExtensions, the client sends V = HKDF(K, <conn-id>).
>>> >>
>>> >> I think that Christian's idea works well enough.  This one requires an
>>> >> impossibility :)
>>> >
>>> >
>>> > Why is this an impossibility?
>>>
>>> The client doesn't send EncryptedExtensions.  Maybe you meant exactly
>>> what Christian said and the V is sent by the server?
>>>
>>
>>
>

--001a113d164e793c66054eca92e2
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Reading through again, I think what I&#39;m actually argui=
ng for is ekr&#39;s PR #20 instead, since a verifier in the clear is the ke=
y distinction between the two as far as I can tell.</div><div class=3D"gmai=
l_extra"><br><div class=3D"gmail_quote">On Thu, May 4, 2017 at 8:16 PM, Mar=
tin Duke <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.h.duke@gmail.com" t=
arget=3D"_blank">martin.h.duke@gmail.com</a>&gt;</span> wrote:<br><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex"><div dir=3D"ltr"><div>I&#39;ve finally been able to p=
rocess this thread. This problem has so many dimensions that we should defi=
nitely discuss in Paris.</div><div><br></div><div>TL;DR I think signals to =
middleboxes are important, and I endorse MT&#39;s mechanism but with the pr=
oof sent in=C2=A0the clear. I have concerns about how this might work witho=
ut connection IDs or with a changing connection ID, if some of the MAYs and=
 SHOULDs aren&#39;t observed, client-generated public resets, etc. Once we =
agree on general approach we can interrogate those issues.</div><div><br></=
div><div>In my opinion, the logic of this signal extends to also doing #353=
 and replacing CONNECTION_CLOSE with a public reset.</div><div><br></div><d=
iv>*************</div><div><br></div><div>There are numerous different trad=
eoffs we discuss above, but reasoning from first principles about middlebox=
es may answer all of these questions. I propose the following assertions:</=
div><div><br></div><div>1. Middleboxes will do stuff to connections that re=
quires state.</div><div><br></div><div>2.=C2=A0There will have to be some s=
ort of idle timeout to clean up that state, because stuff happens.</div><di=
v><br></div><div>3. It is desirable to QUIC endpoints that those timeouts b=
e as long as possible. Premature state teardown=C2=A0will, at the very leas=
t, mess up whatever the middlebox is trying to optimize, and at worst it wi=
ll break the connection (e.g. a firewall may not accept packets if it hasn&=
#39;t seen the Client Hello.)</div><div><br></div><div>4. Middlebox vendors=
 and operators will institute longer timeouts if they have a *reasonable ch=
ance* of being gracefully notified of connection close.</div><div><br></div=
><div>5. On-path observers should not be able to generate (spoof) valid pub=
lic resets to middleboxes.</div><div><br></div><div>From these points, we s=
hould strive to maximize the likelihood of the middlebox=C2=A0seeing and=C2=
=A0validating the public reset, but it&#39;s OK if it misses it in certain =
corner cases. I propose that MT&#39;s mechanism, with the proof sent in the=
 clear and extended to events where Google-QUIC sends CONNECTION_CLOSE, mee=
ts this test. It&#39;s true that path changes and path failure will=C2=A0pr=
event delivery of this signal. But this is not terribly different from the =
balance of events that we see with TCP.</div><div><br></div><div>As for an =
explicit &quot;path close, connection still open&quot; message, I wouldn&#3=
9;t object to this but I don&#39;t see it as necessary, nor do I think it l=
ikely that endpoints would bother to send them.</div><span class=3D"HOEnZb"=
><font color=3D"#888888"><div><br></div><div>Martin Duke</div></font></span=
></div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Wed, May 3, 2017 at 3:47 PM, Eric Rescorla=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ek=
r@rtfm.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"ltr">Yes, I meant the server.<div><br></div><div>-Ekr</div><div><br></d=
iv></div><div class=3D"m_8373981483983712245HOEnZb"><div class=3D"m_8373981=
483983712245h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">O=
n Wed, May 3, 2017 at 3:45 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=
=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail=
.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On 4 May=
 2017 at 07:59, Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com" target=3D=
"_blank">ekr@rtfm.com</a>&gt; wrote:<br>
&gt; On Wed, May 3, 2017 at 2:56 PM, Martin Thomson &lt;<a href=3D"mailto:m=
artin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;=
<br>
&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; On 4 May 2017 at 00:40, Eric Rescorla &lt;<a href=3D"mailto:ekr@rt=
fm.com" target=3D"_blank">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;&gt; &gt; In EncryptedExtensions, the client sends V =3D HKDF(K, &lt;co=
nn-id&gt;).<br>
&gt;&gt;<br>
&gt;&gt; I think that Christian&#39;s idea works well enough.=C2=A0 This on=
e requires an<br>
&gt;&gt; impossibility :)<br>
&gt;<br>
&gt;<br>
&gt; Why is this an impossibility?<br>
<br>
</span>The client doesn&#39;t send EncryptedExtensions.=C2=A0 Maybe you mea=
nt exactly<br>
what Christian said and the V is sent by the server?<br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a113d164e793c66054eca92e2--


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From: Jana Iyengar <jri@google.com>
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Subject: Re: Proposed changes to cleartext packets
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On Thu, May 4, 2017 at 5:09 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> I don't think that either suggestion works particularly well.  I'll
> explain below.
>
> On 5 May 2017 at 09:23, Jana Iyengar <jri@google.com> wrote:
> > I don't think a new connection ID on a Version Negotiation (VN) packet is
> > particularly useful. Specifically, the issue that I believe this was
> > expected to solve (Subodh and/or Kazuho should confirm) was if a server
> does
> > not support the client's requested version but knows of a different
> server
> > that does, it should be able to redirect the client to the correct
> server by
> > sending an appropriate connection ID on the VN packet.
>
> I believe that this was one of the cases that was identified.  And you
> are correct that a stateless reject is the only way to do what was
> intended here.  Either the version negotiation packet includes the
> client's chosen version and is ignored, or it doesn't match the
> version that the other server will ultimately advertise.  BTW, I've
> opened an issue for documenting how a server cluster might be upgraded
> (and rolled back): https://github.com/quicwg/base-drafts/issues/504
>
> The use case you are missing is one where the server receives an
> unsupported version AND it wants to send the client to another server.
> One reason you might want to do that is to avoid a second round trip.
> I'll concede that is a little unlikely given the potential need for
> address validation, so it's a pretty minor gain, but a gain
> nonetheless.
>

So an example would be:
Client sends CHLO in QUICv2
Server only speaks QUICv1, and wants to shed load to another that speaks
QUICv1
Server sends VN with Connection ID that leads client to other server on
retry

This will be a super rare case: VN itself should be rare to begin with,
given Alt-Svc. Within that probability, this is a server that wants to shed
load and knows which server to shed to. And for those connections, we're
saving 1 RTT. This one  RTT will be amortized if there are subsequent
connections, since subsequent connections should learn about the new
version and not incur the VN delay. This optimization is super unlikely to
affect performance for applications or for servers.

Can you demonstrate something that we would lose by allowing the
> server to propose a connection ID here?  The gain might be small, but
> I'm not seeing any downside to allowing it.


I don't think we should add complexity to optimize an RTT for super rare
cases. If we could do it for no cost, that's fine. But I don't think it
makes sense to consider a design that gets limited because of this rare
case. The fewer places that we have code doing the same or similar things,
the better, so if there's a mechanism that works well for the rest of the
cases and VN fits in seamlessly, then sure, let's do it.

> This mechanism (and the previous one in the current spec) have both the
> > server and the client changing connection IDs -- the server does it
> during
> > handshake and the client does it after it receives one or more
> > NEW_CONNECTION_ID frames. I think we can make this simpler (both in
> protocol
> > and impl) with only the client initiating this change by sending a packet
> > with a server-chosen connection ID first.
>
> Let me try to restate your proposal.
>
> You would have the entire handshake complete with the same connection
> ID, then switch to a different connection ID once the handshake
> completes.  A stateless reject would be allowed to cause re-routing
>

Yes, that's correct.


> I'm not sure that I understand the advantages that this design has.
> You claim that it reduces the number of mechanisms to two, but I think
> that you end up with three because you need to use a transport
> parameter.  (FWIW, the current writeup essentially only has two
> mechanisms: NEW_CONNECTION_ID and server handshake packets, though
> I'll concede that there might be three of those).  It has essentially
> the same properties as what I've written up, but it's more complicated
> in several ways.



The NEW_CONNECTION_ID frame implies a packet number gap.  With client
> authentication, the server might receive encrypted packets before the
> handshake completes.  If we take ekr's pull request for
> NEW_CONNECTION_ID, the server can't know what the gap is without
> receiving the handshake packets.


That's a single bit of information we can add to the frame.

This would create more exceptions for what can be sent in the clear,
> which requires more care.  If, as proposed, we allow multiple new
> connection IDs to be sent in the same frame, we wouldn't want that to
> happen here, so that's another new special case to test.
>

So you'd send a transport param in the Server Hello (not in the clear) or a
frame.
If you're sending the frame in response to an SR, it will not contain any
information that you need to be confidential.
If you're sending the frame after the ServerHello, it will be encrypted
with 1-RTT keys.
I don't think you need special rules here.


> That suggests a transport parameter might be a better design, but we
> don't send transport parameters in the HelloRetryRequest.  We'd lose
> confidentiality for those unless we created a new set of rules for
> what transport parameters can be sent in a HelloRetryRequest.  That
> generates more special rules.
>

You'd use a frame. Since HRR is a STREAM frame, and should be a small one,
bundling a NEW_CONNECTION_ID frame with it should be relatively easy (in
code.)


> I think that you raise an important point here:
>
> > Changing the connection ID on an externally visible transition allows
> such load balancers to use the different hashing algorithms correctly on
> the connection IDs.
>
> The current proposal has equally simple rules:
>
> if packet[0] == 0x82 or packet[0] == 0x86:
>   routing = client_selected
> else:
>   routing = server_selected
>
> That is, Client Initial and 0-RTT get routing as though the client chose
> the ID.

You would instead have us switch the test to packet[0] & 0x80 == 0x80
> and packet[0] != 0x87 and packet[0] != 0x88, that is every long form
> packet except the 1-RTT ones.  I don't see any advantage there.
>

That's not right. Your logical expression reduces to:
if packet[0] != 0x87 and packet[0] != 0x88:
  routing = client_selected
else:
  routing = server_selected

... which is the same complexity, but it isn't the correct filter anyways.
There's a related problem that I realized with this most recent scheme:
Connection ID in 0-RTT packets are ambiguous. If they follow a Client
Initial, they're client-chosen, but if they follow a Client Cleartext,
they're server-chosen. I think this is a problem in your earlier version
too. So that's a bummer.

You say you see no advantages. I wrote out the advantages in my previous
email, but perhaps you didn't see them?

1. Explicitly indicating the new connection ID gives the client an explicit
correlator from the client's ID to the new server one, which is better than
assuming that the 4-tuple will only carry packets for this connection.
Currently,
the client will receive the first server packet with an unknown Connection
ID and the only correlator is the fact that it was received on the same
4-tuple that the Client Initial was sent on. If you have multiple
connections on the same 4-tuple, there's no explicit correlator.

2. This mechanism (and the previous one in the current spec) have both the
server and the client changing connection IDs -- the server does it during
handshake and the client does it after it receives one or more
NEW_CONNECTION_ID frames. I think we can make this simpler (both in
protocol and impl) with only the client initiating this change by sending a
packet with a server-chosen connection ID first. If the client were the
only one initiating the change, this would allow the server to not care
about doing any changes mid-connection, and it could simply use the
connection ID in the largest received packet as the outgoing one.

These are still valuable if we can have them.

I'll think about this and get back on Monday. I think the current mechanism
in the PR is sound; I'd like to see if we can address the two issues I
mention above.

--f403045ccd5e82aeff054ed09ffb
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, May 4, 2017 at 5:09 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D=
"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.co=
m</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex=
">I don&#39;t think that either suggestion works particularly well.=C2=A0 I=
&#39;ll<br>
explain below.<br>
<span class=3D"gmail-"><br>
On 5 May 2017 at 09:23, Jana Iyengar &lt;<a href=3D"mailto:jri@google.com">=
jri@google.com</a>&gt; wrote:<br>
&gt; I don&#39;t think a new connection ID on a Version Negotiation (VN) pa=
cket is<br>
&gt; particularly useful. Specifically, the issue that I believe this was<b=
r>
&gt; expected to solve (Subodh and/or Kazuho should confirm) was if a serve=
r does<br>
&gt; not support the client&#39;s requested version but knows of a differen=
t server<br>
&gt; that does, it should be able to redirect the client to the correct ser=
ver by<br>
&gt; sending an appropriate connection ID on the VN packet.<br>
<br>
</span>I believe that this was one of the cases that was identified.=C2=A0 =
And you<br>
are correct that a stateless reject is the only way to do what was<br>
intended here.=C2=A0 Either the version negotiation packet includes the<br>
client&#39;s chosen version and is ignored, or it doesn&#39;t match the<br>
version that the other server will ultimately advertise.=C2=A0 BTW, I&#39;v=
e<br>
opened an issue for documenting how a server cluster might be upgraded<br>
(and rolled back): <a href=3D"https://github.com/quicwg/base-drafts/issues/=
504" rel=3D"noreferrer" target=3D"_blank">https://github.com/quicwg/<wbr>ba=
se-drafts/issues/504</a><br>
<br>
The use case you are missing is one where the server receives an<br>
unsupported version AND it wants to send the client to another server.<br>
One reason you might want to do that is to avoid a second round trip.<br>
I&#39;ll concede that is a little unlikely given the potential need for<br>
address validation, so it&#39;s a pretty minor gain, but a gain<br>
nonetheless.<br></blockquote><div><br></div><div>So an example would be:</d=
iv><div>Client sends CHLO in QUICv2</div><div>Server only speaks QUICv1, an=
d wants to shed load to another that speaks QUICv1</div><div>Server sends V=
N with Connection ID that leads client to other server on retry</div><div><=
br></div><div>This will be a super rare case: VN itself should be rare to b=
egin with, given Alt-Svc. Within that probability, this is a server that wa=
nts to shed load and knows which server to shed to. And for those connectio=
ns, we&#39;re saving 1 RTT. This one =C2=A0RTT will be amortized if there a=
re subsequent connections, since subsequent connections should learn about =
the new version and not incur the VN delay. This optimization is super unli=
kely to affect performance for applications or for servers.</div><div><br><=
/div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bo=
rder-left:1px solid rgb(204,204,204);padding-left:1ex">
Can you demonstrate something that we would lose by allowing the<br>
server to propose a connection ID here?=C2=A0 The gain might be small, but<=
br>
I&#39;m not seeing any downside to allowing it.</blockquote><div><br></div>=
<div>I don&#39;t think we should add complexity to optimize an RTT for supe=
r rare cases. If we could do it for no cost, that&#39;s fine. But I don&#39=
;t think it makes sense to consider a design that gets limited because of t=
his rare case. The fewer places that we have code doing the same or similar=
 things, the better, so if there&#39;s a mechanism that works well for the =
rest of the cases and VN fits in seamlessly, then sure, let&#39;s do it.</d=
iv><div><br></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 c=
lass=3D"gmail-">
&gt; This mechanism (and the previous one in the current spec) have both th=
e<br>
&gt; server and the client changing connection IDs -- the server does it du=
ring<br>
&gt; handshake and the client does it after it receives one or more<br>
&gt; NEW_CONNECTION_ID frames. I think we can make this simpler (both in pr=
otocol<br>
&gt; and impl) with only the client initiating this change by sending a pac=
ket<br>
&gt; with a server-chosen connection ID first.<br>
<br>
</span>Let me try to restate your proposal.<br>
<br>
You would have the entire handshake complete with the same connection<br>
ID, then switch to a different connection ID once the handshake<br>
completes.=C2=A0 A stateless reject would be allowed to cause re-routing<br=
></blockquote><div><br></div><div>Yes, that&#39;s correct.</div><div>=C2=A0=
</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;b=
order-left:1px solid rgb(204,204,204);padding-left:1ex">
I&#39;m not sure that I understand the advantages that this design has.<br>
You claim that it reduces the number of mechanisms to two, but I think<br>
that you end up with three because you need to use a transport<br>
parameter.=C2=A0 (FWIW, the current writeup essentially only has two<br>
mechanisms: NEW_CONNECTION_ID and server handshake packets, though<br>
I&#39;ll concede that there might be three of those).=C2=A0 It has essentia=
lly<br>
the same properties as what I&#39;ve written up, but it&#39;s more complica=
ted<br>
in several ways.</blockquote><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1=
ex">=C2=A0</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0p=
x 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
The NEW_CONNECTION_ID frame implies a packet number gap.=C2=A0 With client<=
br>
authentication, the server might receive encrypted packets before the<br>
handshake completes.=C2=A0 If we take ekr&#39;s pull request for<br>
NEW_CONNECTION_ID, the server can&#39;t know what the gap is without<br>
receiving the handshake packets.</blockquote><div><br></div><div>That&#39;s=
 a single bit of information we can add to the frame.</div><div><br></div><=
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">
This would create more exceptions for what can be sent in the clear,<br>
which requires more care.=C2=A0 If, as proposed, we allow multiple new<br>
connection IDs to be sent in the same frame, we wouldn&#39;t want that to<b=
r>
happen here, so that&#39;s another new special case to test.<br></blockquot=
e><div><br></div><div>So you&#39;d send a transport param in the Server Hel=
lo (not in the clear) or a frame.=C2=A0</div><div>If you&#39;re sending the=
 frame in response to an SR, it will not contain any information that you n=
eed to be confidential.</div><div>If you&#39;re sending the frame after the=
 ServerHello, it will be encrypted with 1-RTT keys.</div><div>I don&#39;t t=
hink you need special rules here.</div><div>=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">
That suggests a transport parameter might be a better design, but we<br>
don&#39;t send transport parameters in the HelloRetryRequest.=C2=A0 We&#39;=
d lose<br>
confidentiality for those unless we created a new set of rules for<br>
what transport parameters can be sent in a HelloRetryRequest.=C2=A0 That<br=
>
generates more special rules.<br></blockquote><div><br></div><div>You&#39;d=
 use a frame. Since HRR is a STREAM frame, and should be a small one, bundl=
ing a NEW_CONNECTION_ID frame with it should be relatively easy (in code.)<=
/div><div>=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">
I think that you raise an important point here:<br>
<span class=3D"gmail-"><br>
&gt; Changing the connection ID on an externally visible transition allows =
such load balancers to use the different hashing algorithms correctly on th=
e connection IDs.<br>
<br>
</span>The current proposal has equally simple rules:<br>
<br>
if packet[0] =3D=3D 0x82 or packet[0] =3D=3D 0x86:<br>
=C2=A0 routing =3D client_selected<br>
else:<br>
=C2=A0 routing =3D server_selected<br>
<br>
That is, Client Initial and 0-RTT get routing as though the client chose th=
e ID.</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
You would instead have us switch the test to packet[0] &amp; 0x80 =3D=3D 0x=
80<br>
and packet[0] !=3D 0x87 and packet[0] !=3D 0x88, that is every long form<br=
>
packet except the 1-RTT ones.=C2=A0 I don&#39;t see any advantage there.<br=
>
</blockquote></div></div><div class=3D"gmail_extra"><br></div><div class=3D=
"gmail_extra">That&#39;s not right. Your logical expression reduces to:</di=
v><div class=3D"gmail_extra">if=C2=A0packet[0] !=3D 0x87 and packet[0] !=3D=
 0x88:</div><div class=3D"gmail_extra">=C2=A0 routing =3D client_selected</=
div><div class=3D"gmail_extra">else:</div><div class=3D"gmail_extra">=C2=A0=
 routing =3D server_selected</div><div class=3D"gmail_extra"><br></div><div=
 class=3D"gmail_extra">... which is the same complexity, but it isn&#39;t t=
he correct filter anyways. There&#39;s a related problem that I realized wi=
th this most recent scheme: Connection ID in 0-RTT packets are ambiguous. I=
f they follow a Client Initial, they&#39;re client-chosen, but if they foll=
ow a Client Cleartext, they&#39;re server-chosen. I think this is a problem=
 in your earlier version too. So that&#39;s a bummer.</div><div class=3D"gm=
ail_extra"><br></div><div class=3D"gmail_extra">You say you see no advantag=
es. I wrote out the advantages in my previous email, but perhaps you didn&#=
39;t see them?=C2=A0</div><div class=3D"gmail_extra"><br></div><div class=
=3D"gmail_extra">1.=C2=A0<span style=3D"font-size:12.8px">Explicitly indica=
ting the new connection ID gives the client an explicit correlator from the=
 client&#39;s ID to the new server one, which is better than assuming that =
the 4-tuple will only carry packets for this connection.</span><span style=
=3D"font-size:12.8px">=C2=A0Currently, the client will receive the first se=
rver packet with an unknown Connection ID and the only correlator is the fa=
ct that it was received on the same 4-tuple that the Client Initial was sen=
t on. If you have multiple connections on the same 4-tuple, there&#39;s no =
explicit correlator.</span></div><div class=3D"gmail_extra"><span style=3D"=
font-size:12.8px"><br></span></div><div class=3D"gmail_extra"><span style=
=3D"font-size:12.8px">2.=C2=A0This mechanism (and the previous one in the c=
urrent spec) have both the server and the client changing connection IDs --=
 the server does it during handshake and the client does it after it receiv=
es one or more NEW_CONNECTION_ID frames. I think we can make this simpler (=
both in protocol and impl) with only the client initiating this change by s=
ending a packet with a server-chosen connection ID first. If the client wer=
e the only one initiating the change, this would allow the server to not ca=
re about doing any changes mid-connection, and it could simply use the conn=
ection ID in the largest received packet as the outgoing one.</span></div><=
div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">These are st=
ill valuable if we can have them.=C2=A0</div><div class=3D"gmail_extra"><br=
></div><div class=3D"gmail_extra">I&#39;ll think about this and get back on=
 Monday. I think the current mechanism in the PR is sound; I&#39;d like to =
see if we can address the two issues I mention above.<br></div></div>

--f403045ccd5e82aeff054ed09ffb--


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On 6 May 2017 at 11:06, Jana Iyengar <jri@google.com> wrote:
> I don't think we should add complexity to optimize an RTT for super rare
> cases. If we could do it for no cost, that's fine.

I guess that it's a matter of perspective. I consider the less complex
case to be where the server's connection ID is always correct and the
server can respond with a new connection ID if the client sends Client
Initial.  That means that having Version Negotiation contain a
server-selected connection ID is the cheap option.

>> I'm not sure that I understand the advantages that this design has.
>> You claim that it reduces the number of mechanisms to two, but I think
>> that you end up with three because you need to use a transport
>> parameter.  (FWIW, the current writeup essentially only has two
>> mechanisms: NEW_CONNECTION_ID and server handshake packets, though
>> I'll concede that there might be three of those).  It has essentially
>> the same properties as what I've written up, but it's more complicated
>> in several ways.
>>
>>
>>
>> The NEW_CONNECTION_ID frame implies a packet number gap.  With client
>> authentication, the server might receive encrypted packets before the
>> handshake completes.  If we take ekr's pull request for
>> NEW_CONNECTION_ID, the server can't know what the gap is without
>> receiving the handshake packets.
>
>
> That's a single bit of information we can add to the frame.

Yep, more complexity.

>> You would instead have us switch the test to packet[0] & 0x80 == 0x80>> and packet[0] != 0x87 and packet[0] != 0x88, that is every long form
>> packet except the 1-RTT ones.  I don't see any advantage there.
>
>
> That's not right. Your logical expression reduces to:
> if packet[0] != 0x87 and packet[0] != 0x88:
>   routing = client_selected
> else:
>   routing = server_selected

Read it again, it's not the same.  It says everything with long form
except those two.  Though the point was poorly made: the logic isn't
particularly relevant.

> ... which is the same complexity, but it isn't the correct filter anyways.
> There's a related problem that I realized with this most recent scheme:
> Connection ID in 0-RTT packets are ambiguous. If they follow a Client
> Initial, they're client-chosen, but if they follow a Client Cleartext,
> they're server-chosen. I think this is a problem in your earlier version
> too. So that's a bummer.
>
> You say you see no advantages. I wrote out the advantages in my previous
> email, but perhaps you didn't see them?
>
> 1. Explicitly indicating the new connection ID gives the client an explicit
> correlator from the client's ID to the new server one, which is better than
> assuming that the 4-tuple will only carry packets for this connection.
> Currently, the client will receive the first server packet with an unknown
> Connection ID and the only correlator is the fact that it was received on
> the same 4-tuple that the Client Initial was sent on. If you have multiple
> connections on the same 4-tuple, there's no explicit correlator.

OK, that didn't seem like a real advantage to me, just a rearrangement
of the furniture.  Packets reach the client based on port/IP pairs.

> 2. This mechanism (and the previous one in the current spec) have both the
> server and the client changing connection IDs -- the server does it during
> handshake and the client does it after it receives one or more
> NEW_CONNECTION_ID frames. I think we can make this simpler (both in protocol
> and impl) with only the client initiating this change by sending a packet
> with a server-chosen connection ID first. If the client were the only one
> initiating the change, this would allow the server to not care about doing
> any changes mid-connection, and it could simply use the connection ID in the
> largest received packet as the outgoing one.

Yes, my assertion was that this isn't in fact simpler due to the need
for special rules during the handshake.

If you wanted to talk about how we might, for example, use a different
connection ID in each direction, then I might concede that there are
advantages in using the frame type.


From nobody Mon May  8 18:36:32 2017
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Subject: Re: Consensus Call: A Few More Issues
From: Mark Nottingham <mnot@mnot.net>
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These are now marked as has-consensus.

Cheers,


> On 19 Apr 2017, at 4:08 pm, Mark Nottingham <mnot@mnot.net> wrote:
>=20
> Hello,
>=20
> The editors believe that the issues below have been closed by the -02 =
drafts (or previous ones); these are the ones that I mentioned as being =
missed in the previous round. I *think* they should be =
non-controversial, since they're so old.
>=20
> Please have a look through them; if there are any resolutions you feel =
need more discussion, please request that the issue be reopened.
>=20
> Issues that we need to discuss more will be reopened. The remaining =
ones will be flagged as `has-consensus.`
>=20
> See =
<https://github.com/quicwg/base-drafts/blob/master/CONTRIBUTING.md#resolvi=
ng-issues> for a reminder about the process we're using here. Even when =
we have consensus, we can reopen an issue if new information emerges =
(and that can take a variety of forms).
>=20
> These issues (and a few more recent ones that we won't call consensus =
on just yet) can be found at:
> =
<https://github.com/quicwg/base-drafts/issues?utf8=3D=E2=9C=93&q=3Dis%3Ais=
sue%20is%3Aclosed%20sort%3Acreated-asc%20-label%3Aduplicate%20-label%3Aedi=
torial%20-label%3Ahas-consensus%20>
>=20
> Cheers,
>=20
> #72: Consider eliminating PRIORITY region from HEADERS
> #73: PRIORITY and Stream ID space
> #146: STREAM frame boundaries
> #206: Mismatch between version negotiation in TLS layer and in QUIC =
layer


--
Mark Nottingham   https://www.mnot.net/


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From: Jana Iyengar <jri@google.com>
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Subject: Re: Proposed changes to cleartext packets
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There are two issues that are I brought up; for clarity's sake, I'll limit
this discussion to the immediate one. I'm summarizing the two proposals,
because it helped clarify my thinking, so I hope it helps others as well.
Please correct / add anything you think is incorrect / missing. I'm back in
favor of the newer proposal, FWIW, see below.

*Proposal #1 (first proposal):*
Client sends Client Initial
Server sends ServerHello/Stateless Retry/Version Negotiation with new
connection ID

If server sends SHLO:
Client continues with connection, uses server-supplied connection ID
Server continues using same connection ID

If server sends SR / VN:
Client resets connection
Client sends Client Cleartext with server-supplied connection ID
Server continues using same connection ID.

Load balancer rules: Complicated; 0-RTT packets from client following
Client Initial carry client-chosen connection ID and those following the
Client Cleartext carry server-chosen connection ID.

Issues:
- no rules specified for the server to verify that the client is indeed
using the server's supplied connection ID.
- no rules specified for client to verify that the server saw its Client
Initial before sending SR/VN. (I see now that the packet number is echoed,
but we should turn this into a requirement for the client to verify. This
is what I meant by a "correlator" for the client to use to tie the
client-chosen ID to the server-chosen one.)

*Proposal #2 (newer proposal):*
Client sends Client Initial
Server sends ServerHello/Stateless Retry/Version Negotiation with new
connection ID

If server sends SHLO:
Client continues with connection, uses server-supplied connection ID
Server continues using same connection ID

If server sends SR / VN:
Client resets connection
Client sends Client Initial with server-supplied connection ID #1 (SCID1)
Server sends SHLO with new server-chosen connection ID #2 (SCID2).
Client continues sending Client Cleartext and 0-RTT packets with SCID1
When SHLO is fully received, client switches to 1-RTT packets and SCID2

Load balancer rules: Simple; 1-RTT packets use server-chosen connection
IDs, everything else uses client-chosen IDs (from the load-balancer's point
of view, even though SCID1 is actually server-chosen.)

Issue (repeating for clarity):
- no rules specified for client to verify that the server saw its Client
Initial before sending SR/VN. (I see now that the packet number is echoed,
but we should turn this into a requirement for the client to verify. This
is what I meant by a "correlator" for the client to use to tie the
client-chosen ID to the server-chosen one.)

Proposal #2 additionally has the value that no rules are required for the
server to verify that the client is indeed using the server's supplied
connection ID, since the load balancer treats SCID1 as client-chosen
anyways.

If my understanding is correct, I am back in favor of Proposal #2, because
I cannot resolve the load balancer question of 0-RTT packets with Proposal
#1. Laying it out like this makes it clearer, and I wonder if some of it
can make it into the draft. Currently, the draft does not really specify
how a server might use this degree of freedom of specifying server-chosen
connection ID in SR packets (especially with load balancing rules the way
they are intended to be), and I think it may be valuable to lay out what a
server might want to do. Should this go in the applicability draft perhaps?

- jana



On Sun, May 7, 2017 at 4:31 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 6 May 2017 at 11:06, Jana Iyengar <jri@google.com> wrote:
> > I don't think we should add complexity to optimize an RTT for super rare
> > cases. If we could do it for no cost, that's fine.
>
> I guess that it's a matter of perspective. I consider the less complex
> case to be where the server's connection ID is always correct and the
> server can respond with a new connection ID if the client sends Client
> Initial.  That means that having Version Negotiation contain a
> server-selected connection ID is the cheap option.
>
> >> I'm not sure that I understand the advantages that this design has.
> >> You claim that it reduces the number of mechanisms to two, but I think
> >> that you end up with three because you need to use a transport
> >> parameter.  (FWIW, the current writeup essentially only has two
> >> mechanisms: NEW_CONNECTION_ID and server handshake packets, though
> >> I'll concede that there might be three of those).  It has essentially
> >> the same properties as what I've written up, but it's more complicated
> >> in several ways.
> >>
> >>
> >>
> >> The NEW_CONNECTION_ID frame implies a packet number gap.  With client
> >> authentication, the server might receive encrypted packets before the
> >> handshake completes.  If we take ekr's pull request for
> >> NEW_CONNECTION_ID, the server can't know what the gap is without
> >> receiving the handshake packets.
> >
> >
> > That's a single bit of information we can add to the frame.
>
> Yep, more complexity.
>
> >> You would instead have us switch the test to packet[0] & 0x80 == 0x80>>
> and packet[0] != 0x87 and packet[0] != 0x88, that is every long form
> >> packet except the 1-RTT ones.  I don't see any advantage there.
> >
> >
> > That's not right. Your logical expression reduces to:
> > if packet[0] != 0x87 and packet[0] != 0x88:
> >   routing = client_selected
> > else:
> >   routing = server_selected
>
> Read it again, it's not the same.  It says everything with long form
> except those two.  Though the point was poorly made: the logic isn't
> particularly relevant.
>
> > ... which is the same complexity, but it isn't the correct filter
> anyways.
> > There's a related problem that I realized with this most recent scheme:
> > Connection ID in 0-RTT packets are ambiguous. If they follow a Client
> > Initial, they're client-chosen, but if they follow a Client Cleartext,
> > they're server-chosen. I think this is a problem in your earlier version
> > too. So that's a bummer.
> >
> > You say you see no advantages. I wrote out the advantages in my previous
> > email, but perhaps you didn't see them?
> >
> > 1. Explicitly indicating the new connection ID gives the client an
> explicit
> > correlator from the client's ID to the new server one, which is better
> than
> > assuming that the 4-tuple will only carry packets for this connection.
> > Currently, the client will receive the first server packet with an
> unknown
> > Connection ID and the only correlator is the fact that it was received on
> > the same 4-tuple that the Client Initial was sent on. If you have
> multiple
> > connections on the same 4-tuple, there's no explicit correlator.
>
> OK, that didn't seem like a real advantage to me, just a rearrangement
> of the furniture.  Packets reach the client based on port/IP pairs.
>
> > 2. This mechanism (and the previous one in the current spec) have both
> the
> > server and the client changing connection IDs -- the server does it
> during
> > handshake and the client does it after it receives one or more
> > NEW_CONNECTION_ID frames. I think we can make this simpler (both in
> protocol
> > and impl) with only the client initiating this change by sending a packet
> > with a server-chosen connection ID first. If the client were the only one
> > initiating the change, this would allow the server to not care about
> doing
> > any changes mid-connection, and it could simply use the connection ID in
> the
> > largest received packet as the outgoing one.
>
> Yes, my assertion was that this isn't in fact simpler due to the need
> for special rules during the handshake.
>
> If you wanted to talk about how we might, for example, use a different
> connection ID in each direction, then I might concede that there are
> advantages in using the frame type.
>

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

<div dir=3D"ltr">There are two issues that are I brought up; for clarity&#3=
9;s sake, I&#39;ll limit this discussion to the immediate one. I&#39;m summ=
arizing the two proposals, because it helped clarify my thinking, so I hope=
 it helps others as well. Please correct / add anything you think is incorr=
ect / missing. I&#39;m back in favor of the newer proposal, FWIW, see below=
.<div><br></div><div><b>Proposal #1 (first proposal):</b></div><div>Client =
sends Client Initial</div><div>Server sends ServerHello/Stateless Retry/Ver=
sion Negotiation with new connection ID</div><div><br></div><div>If server =
sends SHLO:</div><div>Client continues with connection, uses server-supplie=
d connection ID</div><div>Server continues using same connection ID</div><d=
iv><br></div><div>If server sends SR / VN:</div><div>Client resets connecti=
on</div><div>Client sends Client Cleartext with server-supplied connection =
ID</div><div>Server continues using same connection ID.<br></div><div><br><=
/div><div>Load balancer rules: Complicated; 0-RTT packets from client follo=
wing Client Initial carry client-chosen connection ID and those following t=
he Client Cleartext carry server-chosen connection ID.</div><div><br></div>=
<div>Issues:<br>- no rules specified for the server to verify that the clie=
nt is indeed using the server&#39;s supplied connection ID.</div><div>- no =
rules specified for client to verify that the server saw its Client Initial=
 before sending SR/VN. (I see now that the packet number is echoed, but we =
should turn this into a requirement for the client to verify. This is what =
I meant by a &quot;correlator&quot; for the client to use to tie the client=
-chosen ID to the server-chosen one.)</div><div><div><br></div><div><b>Prop=
osal #2 (newer proposal):</b></div><div><div>Client sends Client Initial<br=
></div><div>Server sends ServerHello/Stateless Retry/Version Negotiation wi=
th new connection ID</div><div><br></div><div>If server sends SHLO:</div><d=
iv>Client continues with connection, uses server-supplied connection ID</di=
v><div>Server continues using same connection ID</div><div><br></div><div>I=
f server sends SR / VN:</div><div>Client resets connection</div><div>Client=
 sends Client Initial with server-supplied connection ID #1 (SCID1)</div><d=
iv>Server sends SHLO with new server-chosen connection ID #2 (SCID2).<br></=
div></div></div><div>Client continues sending Client Cleartext and 0-RTT pa=
ckets with SCID1</div><div>When SHLO is fully received, client switches to =
1-RTT packets and SCID2</div><div><br></div><div>Load balancer rules: Simpl=
e; 1-RTT packets use server-chosen connection IDs, everything else uses cli=
ent-chosen IDs (from the load-balancer&#39;s point of view, even though SCI=
D1 is actually server-chosen.)</div><div><br></div><div><div>Issue (repeati=
ng for clarity):</div><div>- no rules specified for client to verify that t=
he server saw its Client Initial before sending SR/VN. (I see now that the =
packet number is echoed, but we should turn this into a requirement for the=
 client to verify. This is what I meant by a &quot;correlator&quot; for the=
 client to use to tie the client-chosen ID to the server-chosen one.)<br></=
div><div><br></div><div><div>Proposal #2 additionally has the value that no=
 rules are required for the server to verify that the client is indeed usin=
g the server&#39;s supplied connection ID, since the load balancer treats S=
CID1 as client-chosen anyways.</div></div><div><br></div><div>If my underst=
anding is correct, I am back in favor of Proposal #2, because I cannot reso=
lve the load balancer question of 0-RTT packets with Proposal #1. Laying it=
 out like this makes it clearer, and I wonder if some of it can make it int=
o the draft. Currently, the draft does not really specify how a server migh=
t use this degree of freedom of specifying server-chosen connection ID in S=
R packets (especially with load balancing rules the way they are intended t=
o be), and I think it may be valuable to lay out what a server might want t=
o do. Should this go in the applicability draft perhaps?</div><div><br></di=
v><div>- jana</div><div><br></div><div><br></div><div></div></div></div><di=
v class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, May 7, 2017 =
at 4:31 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.t=
homson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</span>=
 wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex"><span class=3D"">On 6 May 2017 at=
 11:06, Jana Iyengar &lt;<a href=3D"mailto:jri@google.com">jri@google.com</=
a>&gt; wrote:<br>
&gt; I don&#39;t think we should add complexity to optimize an RTT for supe=
r rare<br>
&gt; cases. If we could do it for no cost, that&#39;s fine.<br>
<br>
</span>I guess that it&#39;s a matter of perspective. I consider the less c=
omplex<br>
case to be where the server&#39;s connection ID is always correct and the<b=
r>
server can respond with a new connection ID if the client sends Client<br>
Initial.=C2=A0 That means that having Version Negotiation contain a<br>
server-selected connection ID is the cheap option.<br>
<span class=3D""><br>
&gt;&gt; I&#39;m not sure that I understand the advantages that this design=
 has.<br>
&gt;&gt; You claim that it reduces the number of mechanisms to two, but I t=
hink<br>
&gt;&gt; that you end up with three because you need to use a transport<br>
&gt;&gt; parameter.=C2=A0 (FWIW, the current writeup essentially only has t=
wo<br>
&gt;&gt; mechanisms: NEW_CONNECTION_ID and server handshake packets, though=
<br>
&gt;&gt; I&#39;ll concede that there might be three of those).=C2=A0 It has=
 essentially<br>
&gt;&gt; the same properties as what I&#39;ve written up, but it&#39;s more=
 complicated<br>
&gt;&gt; in several ways.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; The NEW_CONNECTION_ID frame implies a packet number gap.=C2=A0 Wit=
h client<br>
&gt;&gt; authentication, the server might receive encrypted packets before =
the<br>
&gt;&gt; handshake completes.=C2=A0 If we take ekr&#39;s pull request for<b=
r>
&gt;&gt; NEW_CONNECTION_ID, the server can&#39;t know what the gap is witho=
ut<br>
&gt;&gt; receiving the handshake packets.<br>
&gt;<br>
&gt;<br>
&gt; That&#39;s a single bit of information we can add to the frame.<br>
<br>
</span>Yep, more complexity.<br>
<span class=3D""><br>
&gt;&gt; You would instead have us switch the test to packet[0] &amp; 0x80 =
=3D=3D 0x80&gt;&gt; and packet[0] !=3D 0x87 and packet[0] !=3D 0x88, that i=
s every long form<br>
&gt;&gt; packet except the 1-RTT ones.=C2=A0 I don&#39;t see any advantage =
there.<br>
&gt;<br>
&gt;<br>
&gt; That&#39;s not right. Your logical expression reduces to:<br>
&gt; if packet[0] !=3D 0x87 and packet[0] !=3D 0x88:<br>
&gt;=C2=A0 =C2=A0routing =3D client_selected<br>
&gt; else:<br>
&gt;=C2=A0 =C2=A0routing =3D server_selected<br>
<br>
</span>Read it again, it&#39;s not the same.=C2=A0 It says everything with =
long form<br>
except those two.=C2=A0 Though the point was poorly made: the logic isn&#39=
;t<br>
particularly relevant.<br>
<span class=3D""><br>
&gt; ... which is the same complexity, but it isn&#39;t the correct filter =
anyways.<br>
&gt; There&#39;s a related problem that I realized with this most recent sc=
heme:<br>
&gt; Connection ID in 0-RTT packets are ambiguous. If they follow a Client<=
br>
&gt; Initial, they&#39;re client-chosen, but if they follow a Client Cleart=
ext,<br>
&gt; they&#39;re server-chosen. I think this is a problem in your earlier v=
ersion<br>
&gt; too. So that&#39;s a bummer.<br>
&gt;<br>
&gt; You say you see no advantages. I wrote out the advantages in my previo=
us<br>
&gt; email, but perhaps you didn&#39;t see them?<br>
&gt;<br>
&gt; 1. Explicitly indicating the new connection ID gives the client an exp=
licit<br>
&gt; correlator from the client&#39;s ID to the new server one, which is be=
tter than<br>
&gt; assuming that the 4-tuple will only carry packets for this connection.=
<br>
&gt; Currently, the client will receive the first server packet with an unk=
nown<br>
&gt; Connection ID and the only correlator is the fact that it was received=
 on<br>
&gt; the same 4-tuple that the Client Initial was sent on. If you have mult=
iple<br>
&gt; connections on the same 4-tuple, there&#39;s no explicit correlator.<b=
r>
<br>
</span>OK, that didn&#39;t seem like a real advantage to me, just a rearran=
gement<br>
of the furniture.=C2=A0 Packets reach the client based on port/IP pairs.<br=
>
<span class=3D""><br>
&gt; 2. This mechanism (and the previous one in the current spec) have both=
 the<br>
&gt; server and the client changing connection IDs -- the server does it du=
ring<br>
&gt; handshake and the client does it after it receives one or more<br>
&gt; NEW_CONNECTION_ID frames. I think we can make this simpler (both in pr=
otocol<br>
&gt; and impl) with only the client initiating this change by sending a pac=
ket<br>
&gt; with a server-chosen connection ID first. If the client were the only =
one<br>
&gt; initiating the change, this would allow the server to not care about d=
oing<br>
&gt; any changes mid-connection, and it could simply use the connection ID =
in the<br>
&gt; largest received packet as the outgoing one.<br>
<br>
</span>Yes, my assertion was that this isn&#39;t in fact simpler due to the=
 need<br>
for special rules during the handshake.<br>
<br>
If you wanted to talk about how we might, for example, use a different<br>
connection ID in each direction, then I might concede that there are<br>
advantages in using the frame type.<br>
</blockquote></div><br></div>

--94eb2c11ac62e57e5e054f0dc56b--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Tue, 9 May 2017 12:12:37 +1000
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Subject: Re: Proposed changes to cleartext packets
To: Jana Iyengar <jri@google.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, QUIC WG <quic@ietf.org>
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On 9 May 2017 at 12:04, Jana Iyengar <jri@google.com> wrote:
> Load balancer rules: Complicated; 0-RTT packets from client following Client
> Initial carry client-chosen connection ID and those following the Client
> Cleartext carry server-chosen connection ID.


No, this isn't right.  I already agreed to the client using the same
connection ID marking 0-RTT packets, the PR right now explicitly
requires the client to keep the same connection ID.

It looks like the only disagreement here is over a) whether a Version
Negotiation packet can include a new server-chosen connection ID, and
b) whether the server-chosen connection ID is carried in the
connection ID field, in a frame, or in a transport parameter.


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From: Jana Iyengar <jri@google.com>
Date: Mon, 8 May 2017 19:18:14 -0700
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Subject: Re: Proposed changes to cleartext packets
To: Martin Thomson <martin.thomson@gmail.com>
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--94eb2c14896ce9ce5a054f0df73a
Content-Type: text/plain; charset=UTF-8

Did you see proposal #2?

On May 8, 2017 7:12 PM, "Martin Thomson" <martin.thomson@gmail.com> wrote:

> On 9 May 2017 at 12:04, Jana Iyengar <jri@google.com> wrote:
> > Load balancer rules: Complicated; 0-RTT packets from client following
> Client
> > Initial carry client-chosen connection ID and those following the Client
> > Cleartext carry server-chosen connection ID.
>
>
> No, this isn't right.  I already agreed to the client using the same
> connection ID marking 0-RTT packets, the PR right now explicitly
> requires the client to keep the same connection ID.
>
> It looks like the only disagreement here is over a) whether a Version
> Negotiation packet can include a new server-chosen connection ID, and
> b) whether the server-chosen connection ID is carried in the
> connection ID field, in a frame, or in a transport parameter.
>

--94eb2c14896ce9ce5a054f0df73a
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<div dir=3D"auto">Did you see proposal #2?</div><div class=3D"gmail_extra">=
<br><div class=3D"gmail_quote">On May 8, 2017 7:12 PM, &quot;Martin Thomson=
&quot; &lt;<a href=3D"mailto:martin.thomson@gmail.com">martin.thomson@gmail=
.com</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=
On 9 May 2017 at 12:04, Jana Iyengar &lt;<a href=3D"mailto:jri@google.com">=
jri@google.com</a>&gt; wrote:<br>
&gt; Load balancer rules: Complicated; 0-RTT packets from client following =
Client<br>
&gt; Initial carry client-chosen connection ID and those following the Clie=
nt<br>
&gt; Cleartext carry server-chosen connection ID.<br>
<br>
<br>
No, this isn&#39;t right.=C2=A0 I already agreed to the client using the sa=
me<br>
connection ID marking 0-RTT packets, the PR right now explicitly<br>
requires the client to keep the same connection ID.<br>
<br>
It looks like the only disagreement here is over a) whether a Version<br>
Negotiation packet can include a new server-chosen connection ID, and<br>
b) whether the server-chosen connection ID is carried in the<br>
connection ID field, in a frame, or in a transport parameter.<br>
</blockquote></div></div>

--94eb2c14896ce9ce5a054f0df73a--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Tue, 9 May 2017 12:22:32 +1000
Message-ID: <CABkgnnW30MyH9kg9vKbNoYYMaE1u+WmPVoMvcqAMir-D+f=g5w@mail.gmail.com>
Subject: Re: Proposed changes to cleartext packets
To: Jana Iyengar <jri@google.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, IETF QUIC WG <quic@ietf.org>
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On 9 May 2017 at 12:18, Jana Iyengar <jri@google.com> wrote:
> Did you see proposal #2?

Ahh, you changed your mind and I missed the /VN bit.

Ahh, so we only disagree on when the client switches to using the
server selection.  That clearly takes any options away regarding how
the connection ID is carried.


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From: Jana Iyengar <jri@google.com>
Date: Mon, 8 May 2017 19:36:11 -0700
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Subject: Re: Proposed changes to cleartext packets
To: Martin Thomson <martin.thomson@gmail.com>
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--94eb2c13f16c1819bb054f0e3839
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On May 8, 2017 7:22 PM, "Martin Thomson" <martin.thomson@gmail.com> wrote:

On 9 May 2017 at 12:18, Jana Iyengar <jri@google.com> wrote:
> Did you see proposal #2?

Ahh, you changed your mind and I missed the /VN bit.


I did change my mind after thinking about load balancer issues a bit more
as I've tried to articulate, but I think there was a lot of confusion
because of all the different moving parts.

Ahh, so we only disagree on when the client switches to using the
server selection.  That clearly takes any options away regarding how
the connection ID is carried.


Do we? Assuming that you agree on proposal #2, how would you change
proposal #2?

--94eb2c13f16c1819bb054f0e3839
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<div dir=3D"auto"><div><br><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On May 8, 2017 7:22 PM, &quot;Martin Thomson&quot; &lt;<a href=3D=
"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.co=
m</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"m_16888102074=
76135129quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;paddin=
g-left:1ex"><div class=3D"m_1688810207476135129quoted-text">On 9 May 2017 a=
t 12:18, Jana Iyengar &lt;<a href=3D"mailto:jri@google.com" target=3D"_blan=
k">jri@google.com</a>&gt; wrote:<br>
&gt; Did you see proposal #2?<br>
<br>
</div>Ahh, you changed your mind and I missed the /VN bit.<br></blockquote>=
</div></div></div><div dir=3D"auto"><br></div><div dir=3D"auto">I did chang=
e my mind after thinking about load balancer issues a bit more as I&#39;ve =
tried to articulate, but I think there was a lot of confusion because of al=
l the different moving parts.</div><div dir=3D"auto"><br></div><div dir=3D"=
auto"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote cla=
ss=3D"m_1688810207476135129quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Ahh, so we only disagree on when the client switches to using the<br>
server selection.=C2=A0 That clearly takes any options away regarding how<b=
r>
the connection ID is carried.<br></blockquote></div></div></div><div dir=3D=
"auto"><br></div><div dir=3D"auto">Do we? Assuming that you agree on propos=
al #2, how would you change proposal #2?</div><div dir=3D"auto"><div class=
=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"m_16888102=
07476135129quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">
</blockquote></div><br></div></div></div>

--94eb2c13f16c1819bb054f0e3839--


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From: Jana Iyengar <jri@google.com>
Date: Mon, 8 May 2017 19:41:27 -0700
Message-ID: <CAGD1bZYv4=gJAXkjWTK8PKiPbruq4sXe=QBN79d9S-7H6J7=rw@mail.gmail.com>
Subject: Re: Proposed changes to cleartext packets
To: Martin Thomson <martin.thomson@gmail.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, IETF QUIC WG <quic@ietf.org>
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--f403045cc264ec0d3b054f0e4ac0
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On May 8, 2017 7:36 PM, "Jana Iyengar" <jri@google.com> wrote:



On May 8, 2017 7:22 PM, "Martin Thomson" <martin.thomson@gmail.com> wrote:

On 9 May 2017 at 12:18, Jana Iyengar <jri@google.com> wrote:
> Did you see proposal #2?

Ahh, you changed your mind and I missed the /VN bit.


I did change my mind after thinking about load balancer issues a bit more
as I've tried to articulate, but I think there was a lot of confusion
because of all the different moving parts.


Once we agree on this so far, I'd like to bring up the other questions I'm
considering. They may not be worth it, but it'll be easier to discuss them
once we're on the same page.

Ahh, so we only disagree on when the client switches to using the
server selection.  That clearly takes any options away regarding how
the connection ID is carried.


Do we? Assuming that you agree on proposal #2, how would you change
proposal #2?

--f403045cc264ec0d3b054f0e4ac0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"auto"><div><br><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On May 8, 2017 7:36 PM, &quot;Jana Iyengar&quot; &lt;<a href=3D"m=
ailto:jri@google.com">jri@google.com</a>&gt; wrote:<br type=3D"attribution"=
><blockquote class=3D"quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex"><div dir=3D"auto"><div class=3D"quoted-text"><div=
><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On May 8, 20=
17 7:22 PM, &quot;Martin Thomson&quot; &lt;<a href=3D"mailto:martin.thomson=
@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt; wrote:<br ty=
pe=3D"attribution"><blockquote class=3D"m_5171142666479552712m_168881020747=
6135129quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex"><div class=3D"m_5171142666479552712m_1688810207476135129quoted-t=
ext">On 9 May 2017 at 12:18, Jana Iyengar &lt;<a href=3D"mailto:jri@google.=
com" target=3D"_blank">jri@google.com</a>&gt; wrote:<br>
&gt; Did you see proposal #2?<br>
<br>
</div>Ahh, you changed your mind and I missed the /VN bit.<br></blockquote>=
</div></div></div><div dir=3D"auto"><br></div></div><div dir=3D"auto">I did=
 change my mind after thinking about load balancer issues a bit more as I&#=
39;ve tried to articulate, but I think there was a lot of confusion because=
 of all the different moving parts.</div></div></blockquote></div></div></d=
iv><div dir=3D"auto"><br></div><div dir=3D"auto">Once we agree on this so f=
ar, I&#39;d like to bring up the other questions I&#39;m considering. They =
may not be worth it, but it&#39;ll be easier to discuss them once we&#39;re=
 on the same page.</div><div dir=3D"auto"><br></div><div dir=3D"auto"><div =
class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><=
div dir=3D"auto"><div class=3D"quoted-text"><div dir=3D"auto"><div class=3D=
"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"m_51711426664=
79552712m_1688810207476135129quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
Ahh, so we only disagree on when the client switches to using the<br>
server selection.=C2=A0 That clearly takes any options away regarding how<b=
r>
the connection ID is carried.<br></blockquote></div></div></div><div dir=3D=
"auto"><br></div></div><div dir=3D"auto">Do we? Assuming that you agree on =
proposal #2, how would you change proposal #2?</div><div dir=3D"auto"><div =
class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"m_517=
1142666479552712m_1688810207476135129quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">
</blockquote></div><br></div></div></div>
</blockquote></div><br></div></div></div>

--f403045cc264ec0d3b054f0e4ac0--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Tue, 9 May 2017 14:05:01 +1000
Message-ID: <CABkgnnVmU_Zi2CusAf9BFZS_6fbpV8RdsRjoqO4U=NPd1=jhKA@mail.gmail.com>
Subject: Re: Proposed changes to cleartext packets
To: Jana Iyengar <jri@google.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, IETF QUIC WG <quic@ietf.org>
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On 9 May 2017 at 12:36, Jana Iyengar <jri@google.com> wrote:
> Ahh, so we only disagree on when the client switches to using the
> server selection.  That clearly takes any options away regarding how
> the connection ID is carried.
>
>
> Do we? Assuming that you agree on proposal #2, how would you change proposal
> #2?

I don't have enough info, but I think that there might be a problem.

Let's say that the server does bump the client twice, either a
stateless reject or version negotiation (or both).  That way, when the
ClientHello that the server will consume properly will have a
server-selected connection ID.

Then you have the 1-RTT case with rejections:

<some rejections>
Client Initial, CCID, ClientHello ->
  <- Stateless Retry or Version Negotiation, SCID1, [HelloRetryRequest]
Client Initial, SCID1, ClientHello ->
  <- Server Cleartext, SCID2, ServerHello (etc)
Client Cleartext, SCID1**?, Finished ->
1-RTT Data, SCID2 ->

<no rejections>
Client Initial, CCID, ClientHello ->
  <- Server Cleartext, SCID2, ServerHello (etc)
Client Cleartext, SCID2**?, Finished ->
1-RTT Data, SCID2 ->


The change you propose is on the Client Cleartext line in the "some
rejections" case. If I've understood you correctly, rather than use
SCID2 here (as included in what I wrote up), you want to keep using
SCID1.

A few concerns.  If this is right, I'm not sure how the server
determines that a client change is OK in the "no rejections" case.

I'm assuming here that once a server node sends a ServerHello, it
really wants to at least be able to ensure that client packets reach
that node again.  As you said previously, if you have route_c() and
route_s() and you only apply route_c() when the client might have
chosen the connection ID.  Here, you are proposing that we change that
logic to use route_c() for all cleartext packets from a client.

I'm still not sure what you are trying to gain by the change.  Is the
intent to ensure that the handshake all ends up in the same place as
0-RTT data?  I've been thinking that the first priority would be to
ensure that 1-RTT data and the handshake arrive in the same place and
that 0-RTT data might take extra effort to send to the right place
(but since it is bounded in size, you might reasonably be able to
manage that by forwarding or other such horrors).


From nobody Mon May  8 22:01:40 2017
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From: Martin Thomson <martin.thomson@gmail.com>
Date: Tue, 9 May 2017 15:01:34 +1000
Message-ID: <CABkgnnU5gP5S8QygU0JmKo4D0nBum8BzT9Pfvmf=k=i58JuS8Q@mail.gmail.com>
Subject: Re: Revisiting 0-RTT transport parameter negotiation
To: Jana Iyengar <jri@google.com>
Cc: "Lubashev, Igor" <ilubashe@akamai.com>, Victor Vasiliev <vasilvv@google.com>, IETF QUIC WG <quic@ietf.org>
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I've taken the liberty of raising an issue for this, and a PR:

https://github.com/quicwg/base-drafts/issues/513
https://github.com/quicwg/base-drafts/pull/512

I've copied the design that TLS uses: the parameters are implicit,
just as ALPN is.  I don't think that there is any benefit to including
the values explicitly other than an increase to the ClientHello size
and an extra sanity check.


On 18 April 2017 at 15:41, Martin Thomson <martin.thomson@gmail.com> wrote:
> As I said on the other thread, I think that this design isn't quite
> right, though the problem is certainly real.
>
> The TLS design doesn't rely on the client sending the server what it
> thinks the values are, it relies on the server remembering.  That has
> better confidentiality properties, and it is isomorphic with other TLS
> 0-RTT behaviour.
>
> Note that remembering isn't a chore: the client has to remember the
> ticket anyway.  Now it has to remember the ticket and the server
> transport parameters.  The server has a choice: it can remember them
> or stuff them in the session ticket (or it could decide never to
> change them).  The difference being that remembering means that you
> don't spend critical early bytes on stuff that you know both sides
> already know.
>
>
> On 18 April 2017 at 02:32, Jana Iyengar <jri@google.com> wrote:
>> I like this idea, and I support it. Victor -- do you want to generate a PR
>> for this or would you like me to?
>>
>> On Fri, Apr 7, 2017 at 12:57 PM, Lubashev, Igor <ilubashe@akamai.com> wrote:
>>>
>>> Yes, this seems to work and is simple enough.  It would also close issue
>>> 425.
>>>
>>>
>>>
>>> -          Igor
>>>
>>>
>>>
>>> From: Victor Vasiliev [mailto:vasilvv@google.com]
>>> Sent: Friday, April 07, 2017 2:58 PM
>>> To: IETF QUIC WG <quic@ietf.org>
>>> Subject: Revisiting 0-RTT transport parameter negotiation
>>>
>>>
>>>
>>> There was a question of what should the client assume about the server's
>>>
>>> transport parameters in 0-RTT case.  As a result of the discussion at the
>>>
>>> interim, we arrived to the state where client is allowed to assume either
>>> the
>>>
>>> parameters from the previous session, or the default values.  Server may
>>> choose
>>>
>>> whatever parameters it wants, and if that's incompatible with client's
>>>
>>> assumption of server's 0-RTT parameters, the server may reply with an
>>>
>>> appropriate error or RST_STREAM, after which the client is supposed to
>>> retry
>>>
>>> the connection.
>>>
>>>
>>>
>>> I personally find that this solution has "too many joints", as in, it
>>> allows
>>>
>>> implementations a wide variety of behaviors, which in turn requires their
>>> peers
>>>
>>> to accommodate those scenarios.  The situation with the initial flow
>>> control
>>>
>>> window is especially awkward.  Server might choose to excuse flow control
>>>
>>> violations until the 1-RTT phase is reached, which requires client to
>>>
>>> accommodate possible flow control window decrease upon receiving server
>>> hello;
>>>
>>> or it might choose not to excuse them, which requires client to retry by
>>>
>>> special-casing QUIC_FLOW_CONTROL_RECEIVED_TOO_MUCH_DATA shortly after
>>> 0-RTT as
>>>
>>> a retry signal.  All of this adds a complexity burden onto
>>> implementations, and
>>>
>>> makes comprehensive interop testing very hard.
>>>
>>>
>>>
>>> Instead of the current negotiation scheme outlined in the draft, I propose
>>> that
>>>
>>> we adapt a simple scheme similar to what TLS does with cryptographic
>>> parameters
>>>
>>> and 0-RTT:
>>>
>>>   1) Client sends its vision of server's assumed parameters in the
>>> ClientHello,
>>>
>>>      and is expected to follow them in 0-RTT packets.
>>>
>>>   2) Server either accepts those parameters, or declines 0-RTT altogether,
>>>
>>>      by discarding the 0-RTT data and following up with a full 1-RTT
>>> handshake.
>>>
>>> If server accepts 0-RTT, its parameters in the 1-RTT server hello cannot
>>> be
>>>
>>> more limited than what it accepted from the client.
>>>
>>>
>>>
>>> In this scenario, both the client and the server can use their regular
>>>
>>> connection handling logic for 0-RTT data, since both the client and the
>>> server
>>>
>>> have a consistent view of server's parameters.  This works well when the
>>>
>>> parameters match expectations (most of the time), it's simple to
>>> implement, and
>>>
>>> it's simple to test.
>>>
>>>
>>>
>>>   -- Victor.
>>
>>


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Subject: Re: Proposed changes to cleartext packets
To: Martin Thomson <martin.thomson@gmail.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, IETF QUIC WG <quic@ietf.org>, Subodh Iyengar <subodh@fb.com>, Kazuho Oku <kazuhooku@gmail.com>
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(I'd like Subodh and Kazuho to weigh in, since they were interested in
seeing solutions to this problem.)

Ah -- you are right of course! I mis-wrote one part, which I'll correct
below. Apologies for seeming like I'm pushing a change -- I wasn't. I was
partly trying to think through, and partly mis-articulating, and I think
what you just wrote helped clear out the final cobwebs in my addled brain.
So here's my corrected proposal, and below it is what I now understand to
be the proposal in the PR. In summary, either design works for me. I don't
have a strong argument for either.

Now that I understand, the text fits fine. It might just be me, but if
others find this non-trivial to reason about -- not because the text isn't
clear, but because this may just be non-trivial -- I wonder if it will help
to explain this flow of packets somewhere... perhaps in the Applicability
draft?

----------------
*Proposal #2 (newer proposal), my corrections:*
Client sends Client Initial
Server sends ServerHello/Stateless Retry/Version Negotiation with new
connection ID

If server sends SHLO:
Client continues with connection, uses client-chosen connection-ID for
Client Cleartext and 0-RTT packets *(<-- corrected)*
Client switches to server-supplied connection ID for 1-RTT packets. *(<--
corrected)*
Server continues using same connection ID

If server sends SR / VN:
Client resets connection
Client sends Client Initial with server-supplied connection ID #1 (SCID1)
Server sends SHLO with new server-chosen connection ID #2 (SCID2).
Client continues sending Client Cleartext and 0-RTT packets with SCID1
When SHLO is fully received, client switches to 1-RTT packets and SCID2

Load balancer rules: Simple; 1-RTT packets use server-chosen connection
IDs, everything else uses client-chosen IDs (from the load-balancer's point
of view, even though SCID1 is actually server-chosen.)

Issue (repeating for clarity):
- no rules specified for client to verify that the server saw its Client
Initial before sending SR/VN. (I see now that the packet number is echoed,
but we should turn this into a requirement for the client to verify. This
is what I meant by a "correlator" for the client to use to tie the
client-chosen ID to the server-chosen one.)
---------------------


I believe what you're saying is, use client-chosen connection ID on 0-RTT,
but not on Client Cleartext packets, since Client Cleartext packets are
sent only after the client has heard the first ServerHello packet. Which
would change the load balancer rules, as you articulated earlier: Client
Initial and 0-RTT packets use client-chosen connection IDs, everything else
uses server-chosen connection IDs.

----------------
*Proposal in the PR:*
Client sends Client Initial
Server sends ServerHello/Stateless Retry/Version Negotiation with new
connection ID

If server sends SHLO:
Client continues with connection, uses client-chosen connection-ID for
0-RTT packets
Client switches to server-supplied connection ID for Client Cleartext and
1-RTT packets
Server continues using same connection ID

If server sends SR / VN:
Client resets connection
Client sends Client Initial with server-supplied connection ID #1 (SCID1)
Server sends SHLO with new server-chosen connection ID #2 (SCID2).
Client continues sending 0-RTT packets with SCID1
When first packet of SHLO is received, client switches to SCID2 for Client
Cleartext and 1-RTT packets

Load balancer rules: Simple; Client Initial and Client Cleartext packets
use client-chosen connection IDs, everything else uses server-chosen IDs
(from the load-balancer's point of view, even though SCID1 is actually
server-chosen.)

Issue (repeating for clarity):
- no rules specified for client to verify that the server saw its Client
Initial before sending SR/VN. (I see now that the packet number is echoed,
but we should turn this into a requirement for the client to verify. This
is what I meant by a "correlator" for the client to use to tie the
client-chosen ID to the server-chosen one.)
---------------------



On Mon, May 8, 2017 at 9:05 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 9 May 2017 at 12:36, Jana Iyengar <jri@google.com> wrote:
> > Ahh, so we only disagree on when the client switches to using the
> > server selection.  That clearly takes any options away regarding how
> > the connection ID is carried.
> >
> >
> > Do we? Assuming that you agree on proposal #2, how would you change
> proposal
> > #2?
>
> I don't have enough info, but I think that there might be a problem.
>
> Let's say that the server does bump the client twice, either a
> stateless reject or version negotiation (or both).  That way, when the
> ClientHello that the server will consume properly will have a
> server-selected connection ID.
>
> Then you have the 1-RTT case with rejections:
>
> <some rejections>
> Client Initial, CCID, ClientHello ->
>   <- Stateless Retry or Version Negotiation, SCID1, [HelloRetryRequest]
> Client Initial, SCID1, ClientHello ->
>   <- Server Cleartext, SCID2, ServerHello (etc)
> Client Cleartext, SCID1**?, Finished ->
> 1-RTT Data, SCID2 ->
>
> <no rejections>
> Client Initial, CCID, ClientHello ->
>   <- Server Cleartext, SCID2, ServerHello (etc)
> Client Cleartext, SCID2**?, Finished ->
> 1-RTT Data, SCID2 ->
>
>
> The change you propose is on the Client Cleartext line in the "some
> rejections" case. If I've understood you correctly, rather than use
> SCID2 here (as included in what I wrote up), you want to keep using
> SCID1.
>
> A few concerns.  If this is right, I'm not sure how the server
> determines that a client change is OK in the "no rejections" case.
>
> I'm assuming here that once a server node sends a ServerHello, it
> really wants to at least be able to ensure that client packets reach
> that node again.  As you said previously, if you have route_c() and
> route_s() and you only apply route_c() when the client might have
> chosen the connection ID.  Here, you are proposing that we change that
> logic to use route_c() for all cleartext packets from a client.
>
> I'm still not sure what you are trying to gain by the change.  Is the
> intent to ensure that the handshake all ends up in the same place as
> 0-RTT data?  I've been thinking that the first priority would be to
> ensure that 1-RTT data and the handshake arrive in the same place and
> that 0-RTT data might take extra effort to send to the right place
> (but since it is bounded in size, you might reasonably be able to
> manage that by forwarding or other such horrors).
>

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<div dir=3D"ltr"><div><div>(I&#39;d like Subodh and Kazuho to weigh in, sin=
ce they were interested in seeing solutions to this problem.)</div></div><d=
iv><br></div>Ah -- you are right of course! I mis-wrote one part, which I&#=
39;ll correct below. Apologies for seeming like I&#39;m pushing a change --=
 I wasn&#39;t. I was partly trying to think through, and partly mis-articul=
ating, and I think what you just wrote helped clear out the final cobwebs i=
n my addled brain. So here&#39;s my corrected proposal, and below it is wha=
t I now understand to be the proposal in the PR. In summary, either design =
works for me. I don&#39;t have a strong argument for either.<div><br></div>=
<div>Now that I understand, the text fits fine. It might just be me, but if=
 others find this non-trivial to reason about -- not because the text isn&#=
39;t clear, but because this may just be non-trivial -- I wonder if it will=
 help to explain this flow of packets somewhere... perhaps in the Applicabi=
lity draft?<br></div><div><br></div><div><div>----------------</div><div><d=
iv><b>Proposal #2 (newer proposal), my corrections:</b></div><div>Client se=
nds Client Initial</div><div>Server sends ServerHello/Stateless Retry/Versi=
on Negotiation with new connection ID</div><div><br></div><div>If server se=
nds SHLO:</div><div>Client continues with connection, uses client-chosen co=
nnection-ID for Client Cleartext and 0-RTT packets <b><i>(&lt;-- corrected)=
</i></b></div><div>Client switches to server-supplied connection ID for 1-R=
TT packets. <i><b>(&lt;-- corrected)</b></i></div><div>Server continues usi=
ng same connection ID</div><div><br></div><div>If server sends SR / VN:</di=
v><div>Client resets connection</div><div>Client sends Client Initial with =
server-supplied connection ID #1 (SCID1)</div><div>Server sends SHLO with n=
ew server-chosen connection ID #2 (SCID2).</div><div>Client continues sendi=
ng Client Cleartext and 0-RTT packets with SCID1</div><div>When SHLO is ful=
ly received, client switches to 1-RTT packets and SCID2</div><div><br></div=
><div>Load balancer rules: Simple; 1-RTT packets use server-chosen connecti=
on IDs, everything else uses client-chosen IDs (from the load-balancer&#39;=
s point of view, even though SCID1 is actually server-chosen.)</div><div><b=
r></div><div>Issue (repeating for clarity):</div><div>- no rules specified =
for client to verify that the server saw its Client Initial before sending =
SR/VN. (I see now that the packet number is echoed, but we should turn this=
 into a requirement for the client to verify. This is what I meant by a &qu=
ot;correlator&quot; for the client to use to tie the client-chosen ID to th=
e server-chosen one.)</div></div><div>---------------------</div><div><br><=
/div><div><br></div><div>I believe what you&#39;re saying is, use client-ch=
osen connection ID on 0-RTT, but not on Client Cleartext packets, since Cli=
ent Cleartext packets are sent only after the client has heard the first Se=
rverHello packet. Which would change the load balancer rules, as you articu=
lated earlier: Client Initial and 0-RTT packets use client-chosen connectio=
n IDs, everything else uses server-chosen connection IDs.</div><div><br></d=
iv><div><div>----------------</div><div><div><b>Proposal in the PR:</b></di=
v><div>Client sends Client Initial</div><div>Server sends ServerHello/State=
less Retry/Version Negotiation with new connection ID</div><div><br></div><=
div>If server sends SHLO:</div><div>Client continues with connection, uses =
client-chosen connection-ID for 0-RTT packets</div><div>Client switches to =
server-supplied connection ID for Client Cleartext and 1-RTT packets</div><=
div>Server continues using same connection ID</div><div><br></div><div>If s=
erver sends SR / VN:</div><div>Client resets connection</div><div>Client se=
nds Client Initial with server-supplied connection ID #1 (SCID1)</div><div>=
Server sends SHLO with new server-chosen connection ID #2 (SCID2).</div><di=
v>Client continues sending 0-RTT packets with SCID1</div><div>When first pa=
cket of SHLO is received, client switches to SCID2 for Client Cleartext and=
 1-RTT packets</div><div><br></div><div>Load balancer rules: Simple; Client=
 Initial and Client Cleartext packets use client-chosen connection IDs, eve=
rything else uses server-chosen IDs (from the load-balancer&#39;s point of =
view, even though SCID1 is actually server-chosen.)</div><div><br></div><di=
v>Issue (repeating for clarity):</div><div>- no rules specified for client =
to verify that the server saw its Client Initial before sending SR/VN. (I s=
ee now that the packet number is echoed, but we should turn this into a req=
uirement for the client to verify. This is what I meant by a &quot;correlat=
or&quot; for the client to use to tie the client-chosen ID to the server-ch=
osen one.)</div></div><div>---------------------</div></div><div><br></div>=
<div><br></div></div></div><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On Mon, May 8, 2017 at 9:05 PM, Martin Thomson <span dir=3D"ltr">=
&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.th=
omson@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><sp=
an class=3D"">On 9 May 2017 at 12:36, Jana Iyengar &lt;<a href=3D"mailto:jr=
i@google.com">jri@google.com</a>&gt; wrote:<br>
&gt; Ahh, so we only disagree on when the client switches to using the<br>
&gt; server selection.=C2=A0 That clearly takes any options away regarding =
how<br>
&gt; the connection ID is carried.<br>
&gt;<br>
&gt;<br>
&gt; Do we? Assuming that you agree on proposal #2, how would you change pr=
oposal<br>
&gt; #2?<br>
<br>
</span>I don&#39;t have enough info, but I think that there might be a prob=
lem.<br>
<br>
Let&#39;s say that the server does bump the client twice, either a<br>
stateless reject or version negotiation (or both).=C2=A0 That way, when the=
<br>
ClientHello that the server will consume properly will have a<br>
server-selected connection ID.<br>
<br>
Then you have the 1-RTT case with rejections:<br>
<br>
&lt;some rejections&gt;<br>
Client Initial, CCID, ClientHello -&gt;<br>
=C2=A0 &lt;- Stateless Retry or Version Negotiation, SCID1, [HelloRetryRequ=
est]<br>
Client Initial, SCID1, ClientHello -&gt;<br>
=C2=A0 &lt;- Server Cleartext, SCID2, ServerHello (etc)<br>
Client Cleartext, SCID1**?, Finished -&gt;<br>
1-RTT Data, SCID2 -&gt;<br>
<br>
&lt;no rejections&gt;<br>
Client Initial, CCID, ClientHello -&gt;<br>
=C2=A0 &lt;- Server Cleartext, SCID2, ServerHello (etc)<br>
Client Cleartext, SCID2**?, Finished -&gt;<br>
1-RTT Data, SCID2 -&gt;<br>
<br>
<br>
The change you propose is on the Client Cleartext line in the &quot;some<br=
>
rejections&quot; case. If I&#39;ve understood you correctly, rather than us=
e<br>
SCID2 here (as included in what I wrote up), you want to keep using<br>
SCID1.<br>
<br>
A few concerns.=C2=A0 If this is right, I&#39;m not sure how the server<br>
determines that a client change is OK in the &quot;no rejections&quot; case=
.<br>
<br>
I&#39;m assuming here that once a server node sends a ServerHello, it<br>
really wants to at least be able to ensure that client packets reach<br>
that node again.=C2=A0 As you said previously, if you have route_c() and<br=
>
route_s() and you only apply route_c() when the client might have<br>
chosen the connection ID.=C2=A0 Here, you are proposing that we change that=
<br>
logic to use route_c() for all cleartext packets from a client.<br>
<br>
I&#39;m still not sure what you are trying to gain by the change.=C2=A0 Is =
the<br>
intent to ensure that the handshake all ends up in the same place as<br>
0-RTT data?=C2=A0 I&#39;ve been thinking that the first priority would be t=
o<br>
ensure that 1-RTT data and the handshake arrive in the same place and<br>
that 0-RTT data might take extra effort to send to the right place<br>
(but since it is bounded in size, you might reasonably be able to<br>
manage that by forwarding or other such horrors).<br>
</blockquote></div><br></div>

--f403045cc2644127bc054f1125f8--


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From: Jana Iyengar <jri@google.com>
Date: Mon, 8 May 2017 23:21:00 -0700
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Subject: Re: Proposed changes to cleartext packets
To: Martin Thomson <martin.thomson@gmail.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, IETF QUIC WG <quic@ietf.org>, Subodh Iyengar <subodh@fb.com>, Kazuho Oku <kazuhooku@gmail.com>
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--f403045cc2640b95db054f115cf4
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Let me know if we are on the same page.

I'd still like to see if there's anything we can do about localizing
connection ID changes to the client, but I think it's probably best if we
discuss that separately and not on this PR.

On Mon, May 8, 2017 at 11:05 PM, Jana Iyengar <jri@google.com> wrote:

> (I'd like Subodh and Kazuho to weigh in, since they were interested in
> seeing solutions to this problem.)
>
> Ah -- you are right of course! I mis-wrote one part, which I'll correct
> below. Apologies for seeming like I'm pushing a change -- I wasn't. I was
> partly trying to think through, and partly mis-articulating, and I think
> what you just wrote helped clear out the final cobwebs in my addled brain.
> So here's my corrected proposal, and below it is what I now understand to
> be the proposal in the PR. In summary, either design works for me. I don't
> have a strong argument for either.
>
> Now that I understand, the text fits fine. It might just be me, but if
> others find this non-trivial to reason about -- not because the text isn't
> clear, but because this may just be non-trivial -- I wonder if it will help
> to explain this flow of packets somewhere... perhaps in the Applicability
> draft?
>
> ----------------
> *Proposal #2 (newer proposal), my corrections:*
> Client sends Client Initial
> Server sends ServerHello/Stateless Retry/Version Negotiation with new
> connection ID
>
> If server sends SHLO:
> Client continues with connection, uses client-chosen connection-ID for
> Client Cleartext and 0-RTT packets *(<-- corrected)*
> Client switches to server-supplied connection ID for 1-RTT packets. *(<--
> corrected)*
> Server continues using same connection ID
>
> If server sends SR / VN:
> Client resets connection
> Client sends Client Initial with server-supplied connection ID #1 (SCID1)
> Server sends SHLO with new server-chosen connection ID #2 (SCID2).
> Client continues sending Client Cleartext and 0-RTT packets with SCID1
> When SHLO is fully received, client switches to 1-RTT packets and SCID2
>
> Load balancer rules: Simple; 1-RTT packets use server-chosen connection
> IDs, everything else uses client-chosen IDs (from the load-balancer's point
> of view, even though SCID1 is actually server-chosen.)
>
> Issue (repeating for clarity):
> - no rules specified for client to verify that the server saw its Client
> Initial before sending SR/VN. (I see now that the packet number is echoed,
> but we should turn this into a requirement for the client to verify. This
> is what I meant by a "correlator" for the client to use to tie the
> client-chosen ID to the server-chosen one.)
> ---------------------
>
>
> I believe what you're saying is, use client-chosen connection ID on 0-RTT,
> but not on Client Cleartext packets, since Client Cleartext packets are
> sent only after the client has heard the first ServerHello packet. Which
> would change the load balancer rules, as you articulated earlier: Client
> Initial and 0-RTT packets use client-chosen connection IDs, everything else
> uses server-chosen connection IDs.
>
> ----------------
> *Proposal in the PR:*
> Client sends Client Initial
> Server sends ServerHello/Stateless Retry/Version Negotiation with new
> connection ID
>
> If server sends SHLO:
> Client continues with connection, uses client-chosen connection-ID for
> 0-RTT packets
> Client switches to server-supplied connection ID for Client Cleartext and
> 1-RTT packets
> Server continues using same connection ID
>
> If server sends SR / VN:
> Client resets connection
> Client sends Client Initial with server-supplied connection ID #1 (SCID1)
> Server sends SHLO with new server-chosen connection ID #2 (SCID2).
> Client continues sending 0-RTT packets with SCID1
> When first packet of SHLO is received, client switches to SCID2 for Client
> Cleartext and 1-RTT packets
>
> Load balancer rules: Simple; Client Initial and Client Cleartext packets
> use client-chosen connection IDs, everything else uses server-chosen IDs
> (from the load-balancer's point of view, even though SCID1 is actually
> server-chosen.)
>
> Issue (repeating for clarity):
> - no rules specified for client to verify that the server saw its Client
> Initial before sending SR/VN. (I see now that the packet number is echoed,
> but we should turn this into a requirement for the client to verify. This
> is what I meant by a "correlator" for the client to use to tie the
> client-chosen ID to the server-chosen one.)
> ---------------------
>
>
>
> On Mon, May 8, 2017 at 9:05 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
>> On 9 May 2017 at 12:36, Jana Iyengar <jri@google.com> wrote:
>> > Ahh, so we only disagree on when the client switches to using the
>> > server selection.  That clearly takes any options away regarding how
>> > the connection ID is carried.
>> >
>> >
>> > Do we? Assuming that you agree on proposal #2, how would you change
>> proposal
>> > #2?
>>
>> I don't have enough info, but I think that there might be a problem.
>>
>> Let's say that the server does bump the client twice, either a
>> stateless reject or version negotiation (or both).  That way, when the
>> ClientHello that the server will consume properly will have a
>> server-selected connection ID.
>>
>> Then you have the 1-RTT case with rejections:
>>
>> <some rejections>
>> Client Initial, CCID, ClientHello ->
>>   <- Stateless Retry or Version Negotiation, SCID1, [HelloRetryRequest]
>> Client Initial, SCID1, ClientHello ->
>>   <- Server Cleartext, SCID2, ServerHello (etc)
>> Client Cleartext, SCID1**?, Finished ->
>> 1-RTT Data, SCID2 ->
>>
>> <no rejections>
>> Client Initial, CCID, ClientHello ->
>>   <- Server Cleartext, SCID2, ServerHello (etc)
>> Client Cleartext, SCID2**?, Finished ->
>> 1-RTT Data, SCID2 ->
>>
>>
>> The change you propose is on the Client Cleartext line in the "some
>> rejections" case. If I've understood you correctly, rather than use
>> SCID2 here (as included in what I wrote up), you want to keep using
>> SCID1.
>>
>> A few concerns.  If this is right, I'm not sure how the server
>> determines that a client change is OK in the "no rejections" case.
>>
>> I'm assuming here that once a server node sends a ServerHello, it
>> really wants to at least be able to ensure that client packets reach
>> that node again.  As you said previously, if you have route_c() and
>> route_s() and you only apply route_c() when the client might have
>> chosen the connection ID.  Here, you are proposing that we change that
>> logic to use route_c() for all cleartext packets from a client.
>>
>> I'm still not sure what you are trying to gain by the change.  Is the
>> intent to ensure that the handshake all ends up in the same place as
>> 0-RTT data?  I've been thinking that the first priority would be to
>> ensure that 1-RTT data and the handshake arrive in the same place and
>> that 0-RTT data might take extra effort to send to the right place
>> (but since it is bounded in size, you might reasonably be able to
>> manage that by forwarding or other such horrors).
>>
>
>

--f403045cc2640b95db054f115cf4
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Let me know if we are on the same page.<div><br></div><div=
>I&#39;d still like to see if there&#39;s anything we can do about localizi=
ng connection ID changes to the client, but I think it&#39;s probably best =
if we discuss that separately and not on this PR.</div></div><div class=3D"=
gmail_extra"><br><div class=3D"gmail_quote">On Mon, May 8, 2017 at 11:05 PM=
, Jana Iyengar <span dir=3D"ltr">&lt;<a href=3D"mailto:jri@google.com" targ=
et=3D"_blank">jri@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex"><div dir=3D"ltr"><div><div>(I&#39;d like Subodh and Kazuho to wei=
gh in, since they were interested in seeing solutions to this problem.)</di=
v></div><div><br></div>Ah -- you are right of course! I mis-wrote one part,=
 which I&#39;ll correct below. Apologies for seeming like I&#39;m pushing a=
 change -- I wasn&#39;t. I was partly trying to think through, and partly m=
is-articulating, and I think what you just wrote helped clear out the final=
 cobwebs in my addled brain. So here&#39;s my corrected proposal, and below=
 it is what I now understand to be the proposal in the PR. In summary, eith=
er design works for me. I don&#39;t have a strong argument for either.<div>=
<br></div><div>Now that I understand, the text fits fine. It might just be =
me, but if others find this non-trivial to reason about -- not because the =
text isn&#39;t clear, but because this may just be non-trivial -- I wonder =
if it will help to explain this flow of packets somewhere... perhaps in the=
 Applicability draft?<br></div><div><br></div><div><div>----------------</d=
iv><div><div><b>Proposal #2 (newer proposal), my corrections:</b></div><spa=
n class=3D""><div>Client sends Client Initial</div><div>Server sends Server=
Hello/Stateless Retry/Version Negotiation with new connection ID</div><div>=
<br></div><div>If server sends SHLO:</div></span><div>Client continues with=
 connection, uses client-chosen connection-ID for Client Cleartext and 0-RT=
T packets <b><i>(&lt;-- corrected)</i></b></div><div>Client switches to ser=
ver-supplied connection ID for 1-RTT packets. <i><b>(&lt;-- corrected)</b><=
/i></div><span class=3D""><div>Server continues using same connection ID</d=
iv><div><br></div><div>If server sends SR / VN:</div><div>Client resets con=
nection</div><div>Client sends Client Initial with server-supplied connecti=
on ID #1 (SCID1)</div><div>Server sends SHLO with new server-chosen connect=
ion ID #2 (SCID2).</div><div>Client continues sending Client Cleartext and =
0-RTT packets with SCID1</div><div>When SHLO is fully received, client swit=
ches to 1-RTT packets and SCID2</div><div><br></div><div>Load balancer rule=
s: Simple; 1-RTT packets use server-chosen connection IDs, everything else =
uses client-chosen IDs (from the load-balancer&#39;s point of view, even th=
ough SCID1 is actually server-chosen.)</div><div><br></div><div>Issue (repe=
ating for clarity):</div><div>- no rules specified for client to verify tha=
t the server saw its Client Initial before sending SR/VN. (I see now that t=
he packet number is echoed, but we should turn this into a requirement for =
the client to verify. This is what I meant by a &quot;correlator&quot; for =
the client to use to tie the client-chosen ID to the server-chosen one.)</d=
iv></span></div><div>---------------------</div><div><br></div><div><br></d=
iv><div>I believe what you&#39;re saying is, use client-chosen connection I=
D on 0-RTT, but not on Client Cleartext packets, since Client Cleartext pac=
kets are sent only after the client has heard the first ServerHello packet.=
 Which would change the load balancer rules, as you articulated earlier: Cl=
ient Initial and 0-RTT packets use client-chosen connection IDs, everything=
 else uses server-chosen connection IDs.</div><div><br></div><div><div>----=
------------</div><div><div><b>Proposal in the PR:</b></div><span class=3D"=
"><div>Client sends Client Initial</div><div>Server sends ServerHello/State=
less Retry/Version Negotiation with new connection ID</div><div><br></div><=
div>If server sends SHLO:</div></span><div>Client continues with connection=
, uses client-chosen connection-ID for 0-RTT packets</div><div>Client switc=
hes to server-supplied connection ID for Client Cleartext and 1-RTT packets=
</div><span class=3D""><div>Server continues using same connection ID</div>=
<div><br></div><div>If server sends SR / VN:</div><div>Client resets connec=
tion</div><div>Client sends Client Initial with server-supplied connection =
ID #1 (SCID1)</div><div>Server sends SHLO with new server-chosen connection=
 ID #2 (SCID2).</div></span><div>Client continues sending 0-RTT packets wit=
h SCID1</div><div>When first packet of SHLO is received, client switches to=
 SCID2 for Client Cleartext and 1-RTT packets</div><div><br></div><div>Load=
 balancer rules: Simple; Client Initial and Client Cleartext packets use cl=
ient-chosen connection IDs, everything else uses server-chosen IDs (from th=
e load-balancer&#39;s point of view, even though SCID1 is actually server-c=
hosen.)</div><span class=3D""><div><br></div><div>Issue (repeating for clar=
ity):</div><div>- no rules specified for client to verify that the server s=
aw its Client Initial before sending SR/VN. (I see now that the packet numb=
er is echoed, but we should turn this into a requirement for the client to =
verify. This is what I meant by a &quot;correlator&quot; for the client to =
use to tie the client-chosen ID to the server-chosen one.)</div></span></di=
v><div>---------------------</div></div><div><br></div><div><br></div></div=
></div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Mon, May 8, 2017 at 9:05 PM, Martin Thomso=
n <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=
=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex"><span>On 9 May 2017 at 12:36, Jana Iyengar &lt;<a href=3D=
"mailto:jri@google.com" target=3D"_blank">jri@google.com</a>&gt; wrote:<br>
&gt; Ahh, so we only disagree on when the client switches to using the<br>
&gt; server selection.=C2=A0 That clearly takes any options away regarding =
how<br>
&gt; the connection ID is carried.<br>
&gt;<br>
&gt;<br>
&gt; Do we? Assuming that you agree on proposal #2, how would you change pr=
oposal<br>
&gt; #2?<br>
<br>
</span>I don&#39;t have enough info, but I think that there might be a prob=
lem.<br>
<br>
Let&#39;s say that the server does bump the client twice, either a<br>
stateless reject or version negotiation (or both).=C2=A0 That way, when the=
<br>
ClientHello that the server will consume properly will have a<br>
server-selected connection ID.<br>
<br>
Then you have the 1-RTT case with rejections:<br>
<br>
&lt;some rejections&gt;<br>
Client Initial, CCID, ClientHello -&gt;<br>
=C2=A0 &lt;- Stateless Retry or Version Negotiation, SCID1, [HelloRetryRequ=
est]<br>
Client Initial, SCID1, ClientHello -&gt;<br>
=C2=A0 &lt;- Server Cleartext, SCID2, ServerHello (etc)<br>
Client Cleartext, SCID1**?, Finished -&gt;<br>
1-RTT Data, SCID2 -&gt;<br>
<br>
&lt;no rejections&gt;<br>
Client Initial, CCID, ClientHello -&gt;<br>
=C2=A0 &lt;- Server Cleartext, SCID2, ServerHello (etc)<br>
Client Cleartext, SCID2**?, Finished -&gt;<br>
1-RTT Data, SCID2 -&gt;<br>
<br>
<br>
The change you propose is on the Client Cleartext line in the &quot;some<br=
>
rejections&quot; case. If I&#39;ve understood you correctly, rather than us=
e<br>
SCID2 here (as included in what I wrote up), you want to keep using<br>
SCID1.<br>
<br>
A few concerns.=C2=A0 If this is right, I&#39;m not sure how the server<br>
determines that a client change is OK in the &quot;no rejections&quot; case=
.<br>
<br>
I&#39;m assuming here that once a server node sends a ServerHello, it<br>
really wants to at least be able to ensure that client packets reach<br>
that node again.=C2=A0 As you said previously, if you have route_c() and<br=
>
route_s() and you only apply route_c() when the client might have<br>
chosen the connection ID.=C2=A0 Here, you are proposing that we change that=
<br>
logic to use route_c() for all cleartext packets from a client.<br>
<br>
I&#39;m still not sure what you are trying to gain by the change.=C2=A0 Is =
the<br>
intent to ensure that the handshake all ends up in the same place as<br>
0-RTT data?=C2=A0 I&#39;ve been thinking that the first priority would be t=
o<br>
ensure that 1-RTT data and the handshake arrive in the same place and<br>
that 0-RTT data might take extra effort to send to the right place<br>
(but since it is bounded in size, you might reasonably be able to<br>
manage that by forwarding or other such horrors).<br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--f403045cc2640b95db054f115cf4--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Tue, 9 May 2017 17:05:31 +1000
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Subject: Re: Proposed changes to cleartext packets
To: Jana Iyengar <jri@google.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, IETF QUIC WG <quic@ietf.org>, Subodh Iyengar <subodh@fb.com>, Kazuho Oku <kazuhooku@gmail.com>
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On 9 May 2017 at 16:21, Jana Iyengar <jri@google.com> wrote:
> Let me know if we are on the same page.


Yep, I think that's the essence of the choice we have in front of us.
Both alternatives are relatively simple once we get over the hurdle of
actually understanding each other :)


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Subject: Re: Proposed changes to cleartext packets
To: Martin Thomson <martin.thomson@gmail.com>
Cc: Ted Hardie <ted.ietf@gmail.com>, IETF QUIC WG <quic@ietf.org>, Subodh Iyengar <subodh@fb.com>, Kazuho Oku <kazuhooku@gmail.com>
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>
> Both alternatives are relatively simple once we get over the hurdle of
> actually understanding each other :)
>

We usually get there -- just needed some patience and time... and less
bungling :-)

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<div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Both alternatives are relatively simple once we get over the hurdle of<br>
actually understanding each other :)<br>
</blockquote></div><br></div><div class="gmail_extra">We usually get there -- just needed some patience and time... and less bungling :-)</div><div class="gmail_extra"><br></div></div>

--94eb2c14896c32fde9054f120d48--


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Thu, 11 May 2017 10:22:34 +1000
Message-ID: <CABkgnnWyd8fnb5V6a2_b-FL+Rj+po+r_ozXd+jDDpsLVymfgkQ@mail.gmail.com>
Subject: Re: Proposed changes to cleartext packets
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On 3 May 2017 at 20:43, Martin Thomson <martin.thomson@gmail.com> wrote:
> These are two separate issues, but they are intermingled in annoying
> ways, so I have a single draft PR for both changes:
>
>    https://github.com/quicwg/base-drafts/pull/493

After a bunch of discussion, the editors at least agree that this is
an improvement.  I have just merged this change into the editor's
copy.

We will put this on the agenda for the interim to discuss with a larger group.


From nobody Wed May 10 21:54:15 2017
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Subject: Getting to a First Implementation Draft
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Previously, we've mentioned an intention to have a First Implementation =
Draft -- that is, an Internet-Draft that we feel is suitable for =
implementers to write code to, for the purposes of interoperability =
testing and gathering feedback -- shortly after the Paris interim.

Due to the size and complexity of HTTP-over-QUIC, implementing all four =
drafts for this purpose on a reasonable timeline isn't workable.

Instead, the editors have identified a subset of functionality that they =
believe will serve as a suitable starting point. See:
  https://github.com/quicwg/base-drafts/wiki/First-Implementation-Draft

They've also identified the set of issues that we believe will be =
necessary to have proposals for before Paris; see:
  https://github.com/quicwg/base-drafts/milestone/1

If all goes well, the plan is to have a set of drafts out (very) soon =
that do so; then, we can discuss them and the First Implementation Draft =
Candidate in the lead-up to and during the Paris interim.

After Paris, we'll make any necessary adjustments to the documents and =
publish another set, which will be the First Implementation Draft.

Please comment / raise concerns / make suggestions on-list.=20

Regards,

--
Mark Nottingham   https://www.mnot.net/


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From: =?UTF-8?B?SsSBbmlzIMSMb2RlcnM=?= <janis.coders@gmail.com>
Date: Thu, 11 May 2017 14:57:11 +0300
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Subject: Amplification attack using the same connection id?
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Hi, doesn't the "free to change source-ip and port" feature make
server vulnerable to amplification attacks? Imagine that there is
ongoing session and client spoofs source address, but sends the same
connection-id - and because it knows all the internal states of QUIC,
server should accept and respond to the "new spoofed" address (of
course it should be in a state where server's response is bigger than
request).
NOTE that I am talking about state after the ClientInitial has been verified.
Are my assumptions correct? If yes, then how could this be prevented?
Maybe by re-verifying that client really changed IP/port, but maybe
easier would be allowing to switch IP/port only every X seconds.


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Subject: Re: Amplification attack using the same connection id?
To: =?UTF-8?B?SsSBbmlzIMSMb2RlcnM=?= <janis.coders@gmail.com>, quic@ietf.org
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On 05/11/2017 06:57 AM, Jānis Čoders wrote:
> Hi, doesn't the "free to change source-ip and port" feature make
> server vulnerable to amplification attacks? Imagine that there is
> ongoing session and client spoofs source address, but sends the same
> connection-id - and because it knows all the internal states of QUIC,
> server should accept and respond to the "new spoofed" address (of
> course it should be in a state where server's response is bigger than
> request).
> NOTE that I am talking about state after the ClientInitial has been verified.
> Are my assumptions correct? If yes, then how could this be prevented?
> Maybe by re-verifying that client really changed IP/port, but maybe
> easier would be allowing to switch IP/port only every X seconds.
>

If the received message does not cryptographically verify as being sent
by the peer, it is dropped and no reply is sent.

-Ben

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    On 05/11/2017 06:57 AM, Jānis Čoders wrote:<br>
    <blockquote
cite="mid:CA+tEvRRGrqwGdXve8eaBSr2T9Wkb6u+gNOq=uMpwW73-T6tDtA@mail.gmail.com"
      type="cite">
      <pre wrap="">Hi, doesn't the "free to change source-ip and port" feature make
server vulnerable to amplification attacks? Imagine that there is
ongoing session and client spoofs source address, but sends the same
connection-id - and because it knows all the internal states of QUIC,
server should accept and respond to the "new spoofed" address (of
course it should be in a state where server's response is bigger than
request).
NOTE that I am talking about state after the ClientInitial has been verified.
Are my assumptions correct? If yes, then how could this be prevented?
Maybe by re-verifying that client really changed IP/port, but maybe
easier would be allowing to switch IP/port only every X seconds.

</pre>
    </blockquote>
    <br>
    If the received message does not cryptographically verify as being
    sent by the peer, it is dropped and no reply is sent.<br>
    <br>
    -Ben<br>
  </body>
</html>

--------------97128E7BE9C4D9FF9983A5BF--


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Subject: Re: Amplification attack using the same connection id?
To: =?UTF-8?B?SsSBbmlzIMSMb2RlcnM=?= <janis.coders@gmail.com>, quic@ietf.org
References: <CA+tEvRRGrqwGdXve8eaBSr2T9Wkb6u+gNOq=uMpwW73-T6tDtA@mail.gmail.com> <2a740401-8fb8-4b8a-a8d8-9fe89971f99c@akamai.com>
From: Benjamin Kaduk <bkaduk@akamai.com>
Message-ID: <bc3ed316-0494-b1d1-5a0b-79c0707b7d09@akamai.com>
Date: Thu, 11 May 2017 11:52:26 -0500
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On 05/11/2017 11:42 AM, Benjamin Kaduk wrote:
> On 05/11/2017 06:57 AM, Jānis Čoders wrote:
>> Hi, doesn't the "free to change source-ip and port" feature make
>> server vulnerable to amplification attacks? Imagine that there is
>> ongoing session and client spoofs source address, but sends the same
>> connection-id - and because it knows all the internal states of QUIC,
>> server should accept and respond to the "new spoofed" address (of
>> course it should be in a state where server's response is bigger than
>> request).
>> NOTE that I am talking about state after the ClientInitial has been verified.
>> Are my assumptions correct? If yes, then how could this be prevented?
>> Maybe by re-verifying that client really changed IP/port, but maybe
>> easier would be allowing to switch IP/port only every X seconds.
>>
>
> If the received message does not cryptographically verify as being
> sent by the peer, it is dropped and no reply is sent.

Rereading your message more carefully, maybe the "internal states of
QUIC" is supposed to include the relevant cryptographic material?

Some rate limiting of switching (maybe slightly clever to allow
multipath or similar schemes where a small set of routes is used
interchangeably) is probably advisable for implementations.

-Ben

--------------85FB89650875371EAC3F08FE
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<html>
  <head>
    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    On 05/11/2017 11:42 AM, Benjamin Kaduk wrote:<br>
    <blockquote
      cite="mid:2a740401-8fb8-4b8a-a8d8-9fe89971f99c@akamai.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html; charset=utf-8">
      On 05/11/2017 06:57 AM, Jānis Čoders wrote:<br>
      <blockquote
cite="mid:CA+tEvRRGrqwGdXve8eaBSr2T9Wkb6u+gNOq=uMpwW73-T6tDtA@mail.gmail.com"
        type="cite">
        <pre wrap="">Hi, doesn't the "free to change source-ip and port" feature make
server vulnerable to amplification attacks? Imagine that there is
ongoing session and client spoofs source address, but sends the same
connection-id - and because it knows all the internal states of QUIC,
server should accept and respond to the "new spoofed" address (of
course it should be in a state where server's response is bigger than
request).
NOTE that I am talking about state after the ClientInitial has been verified.
Are my assumptions correct? If yes, then how could this be prevented?
Maybe by re-verifying that client really changed IP/port, but maybe
easier would be allowing to switch IP/port only every X seconds.

</pre>
      </blockquote>
      <br>
      If the received message does not cryptographically verify as being
      sent by the peer, it is dropped and no reply is sent.<br>
    </blockquote>
    <br>
    Rereading your message more carefully, maybe the "internal states of
    QUIC" is supposed to include the relevant cryptographic material?<br>
    <br>
    Some rate limiting of switching (maybe slightly clever to allow
    multipath or similar schemes where a small set of routes is used
    interchangeably) is probably advisable for implementations.<br>
    <br>
    -Ben<br>
  </body>
</html>

--------------85FB89650875371EAC3F08FE--


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From: "Lubashev, Igor" <ilubashe@akamai.com>
To: "Kaduk, Ben" <bkaduk@akamai.com>, =?utf-8?B?SsSBbmlzIMSMb2RlcnM=?= <janis.coders@gmail.com>, "quic@ietf.org" <quic@ietf.org>
Subject: RE: Amplification attack using the same connection id?
Thread-Topic: Amplification attack using the same connection id?
Thread-Index: AQHSynTxuyuEZN72KUe2JZ4J3A/+iKHvmS2AgAAC5gD//8mdcA==
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--_000_0be35d73b1044e20b68018db3470760eusma1exdag1mb5msgcorpak_--


From nobody Thu May 11 12:06:18 2017
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Subject: Re: Amplification attack using the same connection id?
To: "Lubashev, Igor" <ilubashe@akamai.com>, =?UTF-8?B?SsSBbmlzIMSMb2RlcnM=?= <janis.coders@gmail.com>, "quic@ietf.org" <quic@ietf.org>
References: <CA+tEvRRGrqwGdXve8eaBSr2T9Wkb6u+gNOq=uMpwW73-T6tDtA@mail.gmail.com> <2a740401-8fb8-4b8a-a8d8-9fe89971f99c@akamai.com> <bc3ed316-0494-b1d1-5a0b-79c0707b7d09@akamai.com> <0be35d73b1044e20b68018db3470760e@usma1ex-dag1mb5.msg.corp.akamai.com>
From: Benjamin Kaduk <bkaduk@akamai.com>
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Date: Thu, 11 May 2017 13:59:31 -0500
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Would ICMP port unreachable kill the entire connection?  I thought we
were leaning towards treating that as just a path signal.

-Ben

On 05/11/2017 12:55 PM, Lubashev, Igor wrote:
>
> So this attack supposes that an attacker has established a QUIC
> connection and is using that connection’s cryptographic material for a
> reflection attack on a victim.  However, would not the victim simply
> reply with a Public Reset (or ICMP port unreachable), and that would
> kill this QUIC connection?  In the end, the attacker would had done a
> lot of work establishing the connection just to get a single reflected
> QUIC packet to the victim.
>
>  
>
> -          Igor
>
>  
>
> *From:*Benjamin Kaduk [mailto:bkaduk@akamai.com]
> *Sent:* Thursday, May 11, 2017 12:52 PM
> *To:* Jānis Čoders <janis.coders@gmail.com>; quic@ietf.org
> *Subject:* Re: Amplification attack using the same connection id?
>
>  
>
> On 05/11/2017 11:42 AM, Benjamin Kaduk wrote:
>
>     On 05/11/2017 06:57 AM, Jānis Čoders wrote:
>
>         Hi, doesn't the "free to change source-ip and port" feature make
>
>         server vulnerable to amplification attacks? Imagine that there is
>
>         ongoing session and client spoofs source address, but sends the same
>
>         connection-id - and because it knows all the internal states of QUIC,
>
>         server should accept and respond to the "new spoofed" address (of
>
>         course it should be in a state where server's response is bigger than
>
>         request).
>
>         NOTE that I am talking about state after the ClientInitial has been verified.
>
>         Are my assumptions correct? If yes, then how could this be prevented?
>
>         Maybe by re-verifying that client really changed IP/port, but maybe
>
>         easier would be allowing to switch IP/port only every X seconds.
>
>          
>
>
>     If the received message does not cryptographically verify as being
>     sent by the peer, it is dropped and no reply is sent.
>
>
> Rereading your message more carefully, maybe the "internal states of
> QUIC" is supposed to include the relevant cryptographic material?
>
> Some rate limiting of switching (maybe slightly clever to allow
> multipath or similar schemes where a small set of routes is used
> interchangeably) is probably advisable for implementations.
>
> -Ben
>


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    <tt>Would ICMP port unreachable kill the entire connection?  I
      thought we were leaning towards treating that as just a path
      signal.<br>
      <br>
      -Ben<br>
    </tt><br>
    <div class="moz-cite-prefix">On 05/11/2017 12:55 PM, Lubashev, Igor
      wrote:<br>
    </div>
    <blockquote
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">So
            this attack supposes that an attacker has established a QUIC
            connection and is using that connection’s cryptographic
            material for a reflection attack on a victim.  However,
            would not the victim simply reply with a Public Reset (or
            ICMP port unreachable), and that would kill this QUIC
            connection?  In the end, the attacker would had done a lot
            of work establishing the connection just to get a single
            reflected QUIC packet to the victim.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext"><o:p> </o:p></span></p>
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              </span></span></span><!--[endif]--><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">Igor<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext"><o:p> </o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #E1E1E1
            1.0pt;padding:3.0pt 0in 0in 0in">
            <p class="MsoNormal"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">From:</span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">
                Benjamin Kaduk [<a class="moz-txt-link-freetext" href="mailto:bkaduk@akamai.com">mailto:bkaduk@akamai.com</a>]
                <br>
                <b>Sent:</b> Thursday, May 11, 2017 12:52 PM<br>
                <b>To:</b> Jānis Čoders <a class="moz-txt-link-rfc2396E" href="mailto:janis.coders@gmail.com">&lt;janis.coders@gmail.com&gt;</a>;
                <a class="moz-txt-link-abbreviated" href="mailto:quic@ietf.org">quic@ietf.org</a><br>
                <b>Subject:</b> Re: Amplification attack using the same
                connection id?<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p> </o:p></p>
        <p class="MsoNormal">On 05/11/2017 11:42 AM, Benjamin Kaduk
          wrote:<br>
          <br>
          <o:p></o:p></p>
        <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
          <p class="MsoNormal">On 05/11/2017 06:57 AM, Jānis Čoders
            wrote:<br>
            <br>
            <o:p></o:p></p>
          <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
            <pre>Hi, doesn't the "free to change source-ip and port" feature make<o:p></o:p></pre>
            <pre>server vulnerable to amplification attacks? Imagine that there is<o:p></o:p></pre>
            <pre>ongoing session and client spoofs source address, but sends the same<o:p></o:p></pre>
            <pre>connection-id - and because it knows all the internal states of QUIC,<o:p></o:p></pre>
            <pre>server should accept and respond to the "new spoofed" address (of<o:p></o:p></pre>
            <pre>course it should be in a state where server's response is bigger than<o:p></o:p></pre>
            <pre>request).<o:p></o:p></pre>
            <pre>NOTE that I am talking about state after the ClientInitial has been verified.<o:p></o:p></pre>
            <pre>Are my assumptions correct? If yes, then how could this be prevented?<o:p></o:p></pre>
            <pre>Maybe by re-verifying that client really changed IP/port, but maybe<o:p></o:p></pre>
            <pre>easier would be allowing to switch IP/port only every X seconds.<o:p></o:p></pre>
            <pre><o:p> </o:p></pre>
          </blockquote>
          <p class="MsoNormal"><br>
            If the received message does not cryptographically verify as
            being sent by the peer, it is dropped and no reply is sent.<o:p></o:p></p>
        </blockquote>
        <p class="MsoNormal"><br>
          Rereading your message more carefully, maybe the "internal
          states of QUIC" is supposed to include the relevant
          cryptographic material?<br>
          <br>
          Some rate limiting of switching (maybe slightly clever to
          allow multipath or similar schemes where a small set of routes
          is used interchangeably) is probably advisable for
          implementations.<br>
          <br>
          -Ben<o:p></o:p></p>
      </div>
    </blockquote>
    <br>
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Subject: Re: Amplification attack using the same connection id?
To: quic@ietf.org
References: <CA+tEvRRGrqwGdXve8eaBSr2T9Wkb6u+gNOq=uMpwW73-T6tDtA@mail.gmail.com> <2a740401-8fb8-4b8a-a8d8-9fe89971f99c@akamai.com> <bc3ed316-0494-b1d1-5a0b-79c0707b7d09@akamai.com> <0be35d73b1044e20b68018db3470760e@usma1ex-dag1mb5.msg.corp.akamai.com>
From: Roland Zink <roland@zinks.de>
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The victim has no state about the connection and probably can't send an 
authenticated public reset. Does this mean that public resets shouldn't 
be authenticated?

Roland



Am 11.05.2017 um 19:55 schrieb Lubashev, Igor:
>
> So this attack supposes that an attacker has established a QUIC 
> connection and is using that connection’s cryptographic material for a 
> reflection attack on a victim.  However, would not the victim simply 
> reply with a Public Reset (or ICMP port unreachable), and that would 
> kill this QUIC connection?  In the end, the attacker would had done a 
> lot of work establishing the connection just to get a single reflected 
> QUIC packet to the victim.
>
> -Igor
>
> *From:*Benjamin Kaduk [mailto:bkaduk@akamai.com]
> *Sent:* Thursday, May 11, 2017 12:52 PM
> *To:* Jānis Čoders <janis.coders@gmail.com>; quic@ietf.org
> *Subject:* Re: Amplification attack using the same connection id?
>
> On 05/11/2017 11:42 AM, Benjamin Kaduk wrote:
>
>     On 05/11/2017 06:57 AM, Jānis Čoders wrote:
>
>         Hi, doesn't the "free to change source-ip and port" feature make
>
>         server vulnerable to amplification attacks? Imagine that there is
>
>         ongoing session and client spoofs source address, but sends the same
>
>         connection-id - and because it knows all the internal states of QUIC,
>
>         server should accept and respond to the "new spoofed" address (of
>
>         course it should be in a state where server's response is bigger than
>
>         request).
>
>         NOTE that I am talking about state after the ClientInitial has been verified.
>
>         Are my assumptions correct? If yes, then how could this be prevented?
>
>         Maybe by re-verifying that client really changed IP/port, but maybe
>
>         easier would be allowing to switch IP/port only every X seconds.
>
>
>     If the received message does not cryptographically verify as being
>     sent by the peer, it is dropped and no reply is sent.
>
>
> Rereading your message more carefully, maybe the "internal states of 
> QUIC" is supposed to include the relevant cryptographic material?
>
> Some rate limiting of switching (maybe slightly clever to allow 
> multipath or similar schemes where a small set of routes is used 
> interchangeably) is probably advisable for implementations.
>
> -Ben
>


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    <p>The victim has no state about the connection and probably can't
      send an authenticated public reset. Does this mean that public
      resets shouldn't be authenticated?</p>
    <p>Roland</p>
    <p><br>
    </p>
    <br>
    <div class="moz-cite-prefix">Am 11.05.2017 um 19:55 schrieb
      Lubashev, Igor:<br>
    </div>
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">So
            this attack supposes that an attacker has established a QUIC
            connection and is using that connection’s cryptographic
            material for a reflection attack on a victim.  However,
            would not the victim simply reply with a Public Reset (or
            ICMP port unreachable), and that would kill this QUIC
            connection?  In the end, the attacker would had done a lot
            of work establishing the connection just to get a single
            reflected QUIC packet to the victim.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext"><o:p> </o:p></span></p>
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              </span></span></span><!--[endif]--><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">Igor<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext"><o:p> </o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #E1E1E1
            1.0pt;padding:3.0pt 0in 0in 0in">
            <p class="MsoNormal"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">From:</span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:windowtext">
                Benjamin Kaduk [<a class="moz-txt-link-freetext" href="mailto:bkaduk@akamai.com">mailto:bkaduk@akamai.com</a>]
                <br>
                <b>Sent:</b> Thursday, May 11, 2017 12:52 PM<br>
                <b>To:</b> Jānis Čoders <a class="moz-txt-link-rfc2396E" href="mailto:janis.coders@gmail.com">&lt;janis.coders@gmail.com&gt;</a>;
                <a class="moz-txt-link-abbreviated" href="mailto:quic@ietf.org">quic@ietf.org</a><br>
                <b>Subject:</b> Re: Amplification attack using the same
                connection id?<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p> </o:p></p>
        <p class="MsoNormal">On 05/11/2017 11:42 AM, Benjamin Kaduk
          wrote:<br>
          <br>
          <o:p></o:p></p>
        <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
          <p class="MsoNormal">On 05/11/2017 06:57 AM, Jānis Čoders
            wrote:<br>
            <br>
            <o:p></o:p></p>
          <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
            <pre>Hi, doesn't the "free to change source-ip and port" feature make<o:p></o:p></pre>
            <pre>server vulnerable to amplification attacks? Imagine that there is<o:p></o:p></pre>
            <pre>ongoing session and client spoofs source address, but sends the same<o:p></o:p></pre>
            <pre>connection-id - and because it knows all the internal states of QUIC,<o:p></o:p></pre>
            <pre>server should accept and respond to the "new spoofed" address (of<o:p></o:p></pre>
            <pre>course it should be in a state where server's response is bigger than<o:p></o:p></pre>
            <pre>request).<o:p></o:p></pre>
            <pre>NOTE that I am talking about state after the ClientInitial has been verified.<o:p></o:p></pre>
            <pre>Are my assumptions correct? If yes, then how could this be prevented?<o:p></o:p></pre>
            <pre>Maybe by re-verifying that client really changed IP/port, but maybe<o:p></o:p></pre>
            <pre>easier would be allowing to switch IP/port only every X seconds.<o:p></o:p></pre>
            <pre><o:p> </o:p></pre>
          </blockquote>
          <p class="MsoNormal"><br>
            If the received message does not cryptographically verify as
            being sent by the peer, it is dropped and no reply is sent.<o:p></o:p></p>
        </blockquote>
        <p class="MsoNormal"><br>
          Rereading your message more carefully, maybe the "internal
          states of QUIC" is supposed to include the relevant
          cryptographic material?<br>
          <br>
          Some rate limiting of switching (maybe slightly clever to
          allow multipath or similar schemes where a small set of routes
          is used interchangeably) is probably advisable for
          implementations.<br>
          <br>
          -Ben<o:p></o:p></p>
      </div>
    </blockquote>
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From: =?UTF-8?B?SsSBbmlzIMSMb2RlcnM=?= <janis.coders@gmail.com>
Date: Thu, 11 May 2017 22:03:03 +0300
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Subject: Re: Amplification attack using the same connection id?
To: "Lubashev, Igor" <ilubashe@akamai.com>
Cc: "Kaduk, Ben" <bkaduk@akamai.com>, "quic@ietf.org" <quic@ietf.org>
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Well the attack will most probably be made on device/host that doesnt
use QUIC or doesnt respond with ICMP because of firewall - but the
attacks goal is to saturate target's network. What about situation
where a protocol which runs over QUIC responds with big data payload
(e.g. client asks for HTTP GET and in the last QUIC packet, changes
IP. The HTTP layer doesn't know(care) about transport layer and just
sends multiple packets to target - probably until non anknowledget
window size is full - and as I understand QUIC should resend the
packets, because they are not acknowledget.

On 11 May 2017 at 20:55, Lubashev, Igor <ilubashe@akamai.com> wrote:
> So this attack supposes that an attacker has established a QUIC connectio=
n
> and is using that connection=E2=80=99s cryptographic material for a refle=
ction
> attack on a victim.  However, would not the victim simply reply with a
> Public Reset (or ICMP port unreachable), and that would kill this QUIC
> connection?  In the end, the attacker would had done a lot of work
> establishing the connection just to get a single reflected QUIC packet to
> the victim.
>
>
>
> -          Igor
>
>
>
> From: Benjamin Kaduk [mailto:bkaduk@akamai.com]
> Sent: Thursday, May 11, 2017 12:52 PM
> To: J=C4=81nis =C4=8Coders <janis.coders@gmail.com>; quic@ietf.org
> Subject: Re: Amplification attack using the same connection id?
>
>
>
> On 05/11/2017 11:42 AM, Benjamin Kaduk wrote:
>
> On 05/11/2017 06:57 AM, J=C4=81nis =C4=8Coders wrote:
>
> Hi, doesn't the "free to change source-ip and port" feature make
>
> server vulnerable to amplification attacks? Imagine that there is
>
> ongoing session and client spoofs source address, but sends the same
>
> connection-id - and because it knows all the internal states of QUIC,
>
> server should accept and respond to the "new spoofed" address (of
>
> course it should be in a state where server's response is bigger than
>
> request).
>
> NOTE that I am talking about state after the ClientInitial has been
> verified.
>
> Are my assumptions correct? If yes, then how could this be prevented?
>
> Maybe by re-verifying that client really changed IP/port, but maybe
>
> easier would be allowing to switch IP/port only every X seconds.
>
>
>
>
> If the received message does not cryptographically verify as being sent b=
y
> the peer, it is dropped and no reply is sent.
>
>
> Rereading your message more carefully, maybe the "internal states of QUIC=
"
> is supposed to include the relevant cryptographic material?
>
> Some rate limiting of switching (maybe slightly clever to allow multipath=
 or
> similar schemes where a small set of routes is used interchangeably) is
> probably advisable for implementations.
>
> -Ben



--=20
Ar cie=C5=86u,
J=C4=81nis =C4=8Coders


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From: "Lubashev, Igor" <ilubashe@akamai.com>
To: Roland Zink <roland@zinks.de>, "quic@ietf.org" <quic@ietf.org>
Subject: RE: Amplification attack using the same connection id?
Thread-Topic: Amplification attack using the same connection id?
Thread-Index: AQHSynTxuyuEZN72KUe2JZ4J3A/+iKHvmS2AgAAC5gD//8mdcIAAWjwA///AXzA=
Date: Thu, 11 May 2017 19:18:08 +0000
Message-ID: <ef1de6f866a040d9af815d362c8eb735@usma1ex-dag1mb5.msg.corp.akamai.com>
References: <CA+tEvRRGrqwGdXve8eaBSr2T9Wkb6u+gNOq=uMpwW73-T6tDtA@mail.gmail.com> <2a740401-8fb8-4b8a-a8d8-9fe89971f99c@akamai.com> <bc3ed316-0494-b1d1-5a0b-79c0707b7d09@akamai.com> <0be35d73b1044e20b68018db3470760e@usma1ex-dag1mb5.msg.corp.akamai.com> <aa224187-f073-9437-3482-222334c164f0@zinks.de>
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From: "Lubashev, Igor" <ilubashe@akamai.com>
To: "Kaduk, Ben" <bkaduk@akamai.com>, =?utf-8?B?SsSBbmlzIMSMb2RlcnM=?= <janis.coders@gmail.com>, "quic@ietf.org" <quic@ietf.org>
Subject: RE: Amplification attack using the same connection id?
Thread-Topic: Amplification attack using the same connection id?
Thread-Index: AQHSynTxuyuEZN72KUe2JZ4J3A/+iKHvmS2AgAAC5gD//8mdcIAAWeWA///CMmA=
Date: Thu, 11 May 2017 19:21:08 +0000
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From: Martin Thomson <martin.thomson@gmail.com>
Date: Fri, 12 May 2017 10:40:19 +1000
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Subject: Re: Amplification attack using the same connection id?
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On 11 May 2017 at 21:57, J=C4=81nis =C4=8Coders <janis.coders@gmail.com> wr=
ote:
> Hi, doesn't the "free to change source-ip and port" feature make
> server vulnerable to amplification attacks? Imagine that there is
> ongoing session and client spoofs source address, but sends the same
> connection-id - and because it knows all the internal states of QUIC,
> server should accept and respond to the "new spoofed" address (of
> course it should be in a state where server's response is bigger than
> request).

FYI, this is an issue we are aware of:
https://github.com/quicwg/base-drafts/issues/161

First, the server resets its congestion state when it detects a new
path and this is obviously a new path.  That isn't sufficient in
practice because asymmetric traffic patterns could eventually lead to
some severe amplification.  Think video downloads for an example.

The main fix is that the server needs to verify that the client can
receive at this new address before it sends (significantly) more than
it receives *on this path*. This is not currently guaranteed, though
there is some discussion about using packet number skipping for this
in the draft.  I'm personally not especially happy about that
approach.

We've discussed several ideas for this, because it's the important one
for handling amplification.  A simple example of what has been
suggested: the server sends a ping that contains entropy and requires
an echo of that ping.


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From: =?UTF-8?B?SsSBbmlzIMSMb2RlcnM=?= <janis.coders@gmail.com>
Date: Fri, 12 May 2017 09:09:57 +0300
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Subject: Re: Amplification attack using the same connection id?
To: Martin Thomson <martin.thomson@gmail.com>
Cc: QUIC WG <quic@ietf.org>
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Thanks, so there is already discussion about this. I'll try to follow
it, but currently for me a "challenge ping" seems more reasonable
solution - clearer intention.

On 12 May 2017 at 03:40, Martin Thomson <martin.thomson@gmail.com> wrote:
> On 11 May 2017 at 21:57, J=C4=81nis =C4=8Coders <janis.coders@gmail.com> =
wrote:
>> Hi, doesn't the "free to change source-ip and port" feature make
>> server vulnerable to amplification attacks? Imagine that there is
>> ongoing session and client spoofs source address, but sends the same
>> connection-id - and because it knows all the internal states of QUIC,
>> server should accept and respond to the "new spoofed" address (of
>> course it should be in a state where server's response is bigger than
>> request).
>
> FYI, this is an issue we are aware of:
> https://github.com/quicwg/base-drafts/issues/161
>
> First, the server resets its congestion state when it detects a new
> path and this is obviously a new path.  That isn't sufficient in
> practice because asymmetric traffic patterns could eventually lead to
> some severe amplification.  Think video downloads for an example.
>
> The main fix is that the server needs to verify that the client can
> receive at this new address before it sends (significantly) more than
> it receives *on this path*. This is not currently guaranteed, though
> there is some discussion about using packet number skipping for this
> in the draft.  I'm personally not especially happy about that
> approach.
>
> We've discussed several ideas for this, because it's the important one
> for handling amplification.  A simple example of what has been
> suggested: the server sends a ping that contains entropy and requires
> an echo of that ping.



--=20
Ar cie=C5=86u,
J=C4=81nis =C4=8Coders


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Subject: Re: Getting to a First Implementation Draft
To: Mark Nottingham <mnot@mnot.net>
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--94eb2c14d834898b6a054f6c5cd7
Content-Type: text/plain; charset="UTF-8"

A few comments on this proposed list:

> Integration with TLS 1.3 handshake - The basic 1-RTT mode must be
> supported. TLS exporters are not needed, nor are session
> tickets. Basic key exchange is sufficient and implementations can
> use any certificate. All MTI algorithms listed in TLS 1.3 are
> expected.

I note that this and the transport parameters and HRR stuff below
necessitate the generic changes to the TLS library to permit adding
extra extensions. I wonder if it might make life better to just have a
canned set of parameters for now. It's not that it's a lot of work,
it's just that it's not critical path otherwise.


> Packet protection - All post handshake packets must be sent with
> 1RTT keys and packet protection.

This is inconsistent with not requiring exporters above. As far
as I can tell, if you're not doing NST, you don't need post-handshake
packets anyway here, so you can probably skip this.



On Wed, May 10, 2017 at 9:53 PM, Mark Nottingham <mnot@mnot.net> wrote:

> Previously, we've mentioned an intention to have a First Implementation
> Draft -- that is, an Internet-Draft that we feel is suitable for
> implementers to write code to, for the purposes of interoperability testing
> and gathering feedback -- shortly after the Paris interim.
>
> Due to the size and complexity of HTTP-over-QUIC, implementing all four
> drafts for this purpose on a reasonable timeline isn't workable.
>
> Instead, the editors have identified a subset of functionality that they
> believe will serve as a suitable starting point. See:
>   https://github.com/quicwg/base-drafts/wiki/First-Implementation-Draft
>
> They've also identified the set of issues that we believe will be
> necessary to have proposals for before Paris; see:
>   https://github.com/quicwg/base-drafts/milestone/1
>
> If all goes well, the plan is to have a set of drafts out (very) soon that
> do so; then, we can discuss them and the First Implementation Draft
> Candidate in the lead-up to and during the Paris interim.
>
> After Paris, we'll make any necessary adjustments to the documents and
> publish another set, which will be the First Implementation Draft.
>
> Please comment / raise concerns / make suggestions on-list.
>
> Regards,
>
> --
> Mark Nottingham   https://www.mnot.net/
>
>

--94eb2c14d834898b6a054f6c5cd7
Content-Type: text/html; charset="UTF-8"
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<div dir=3D"ltr">A few comments on this proposed list:<div><br></div><div><=
div>&gt; Integration with TLS 1.3 handshake - The basic 1-RTT mode must be<=
/div><div>&gt; supported. TLS exporters are not needed, nor are session</di=
v><div>&gt; tickets. Basic key exchange is sufficient and implementations c=
an</div><div>&gt; use any certificate. All MTI algorithms listed in TLS 1.3=
 are</div><div>&gt; expected.</div><div><br></div><div>I note that this and=
 the transport parameters and HRR stuff below</div><div>necessitate the gen=
eric changes to the TLS library to permit adding</div><div>extra extensions=
. I wonder if it might make life better to just have a</div><div>canned set=
 of parameters for now. It&#39;s not that it&#39;s a lot of work,</div><div=
>it&#39;s just that it&#39;s not critical path otherwise.</div><div><br></d=
iv><div><br></div><div>&gt; Packet protection - All post handshake packets =
must be sent with</div><div>&gt; 1RTT keys and packet protection.</div><div=
><br></div><div>This is inconsistent with not requiring exporters above. As=
 far</div><div>as I can tell, if you&#39;re not doing NST, you don&#39;t ne=
ed post-handshake</div><div>packets anyway here, so you can probably skip t=
his.</div><div><br></div></div><div><br></div></div><div class=3D"gmail_ext=
ra"><br><div class=3D"gmail_quote">On Wed, May 10, 2017 at 9:53 PM, Mark No=
ttingham <span dir=3D"ltr">&lt;<a href=3D"mailto:mnot@mnot.net" target=3D"_=
blank">mnot@mnot.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
">Previously, we&#39;ve mentioned an intention to have a First Implementati=
on Draft -- that is, an Internet-Draft that we feel is suitable for impleme=
nters to write code to, for the purposes of interoperability testing and ga=
thering feedback -- shortly after the Paris interim.<br>
<br>
Due to the size and complexity of HTTP-over-QUIC, implementing all four dra=
fts for this purpose on a reasonable timeline isn&#39;t workable.<br>
<br>
Instead, the editors have identified a subset of functionality that they be=
lieve will serve as a suitable starting point. See:<br>
=C2=A0 <a href=3D"https://github.com/quicwg/base-drafts/wiki/First-Implemen=
tation-Draft" rel=3D"noreferrer" target=3D"_blank">https://github.com/quicw=
g/<wbr>base-drafts/wiki/First-<wbr>Implementation-Draft</a><br>
<br>
They&#39;ve also identified the set of issues that we believe will be neces=
sary to have proposals for before Paris; see:<br>
=C2=A0 <a href=3D"https://github.com/quicwg/base-drafts/milestone/1" rel=3D=
"noreferrer" target=3D"_blank">https://github.com/quicwg/<wbr>base-drafts/m=
ilestone/1</a><br>
<br>
If all goes well, the plan is to have a set of drafts out (very) soon that =
do so; then, we can discuss them and the First Implementation Draft Candida=
te in the lead-up to and during the Paris interim.<br>
<br>
After Paris, we&#39;ll make any necessary adjustments to the documents and =
publish another set, which will be the First Implementation Draft.<br>
<br>
Please comment / raise concerns / make suggestions on-list.<br>
<br>
Regards,<br>
<br>
--<br>
Mark Nottingham=C2=A0 =C2=A0<a href=3D"https://www.mnot.net/" rel=3D"norefe=
rrer" target=3D"_blank">https://www.mnot.net/</a><br>
<br>
</blockquote></div><br></div>

--94eb2c14d834898b6a054f6c5cd7--


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Subject: Re: Getting to a First Implementation Draft
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--001a114a80fc1076d3054f7fdedb
Content-Type: text/plain; charset="UTF-8"

I don't really know how I feel about this wiki page.. without ratholing too
deeply on it I think that while it tries to facilitate testing and interop,
it treads pretty significantly into the land of project management which is
squarely out of bounds for the working group. I'm willing to see how it
goes but that's my concern - I would rather the WG focuses on the complete
picture while providing more of a forum for pair (or more) wise testing.

anyhow wrt specifics -  ekr points out the 2 major inconsistencies.. tbh I
would require exporters and the ability to do at least one non-0 stream. At
that point you have a hello world milestone, which is always the first
stage worthy of a toast. Its ok by me if the transport params are just
profiled.

On Sat, May 13, 2017 at 2:54 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> A few comments on this proposed list:
>
> > Integration with TLS 1.3 handshake - The basic 1-RTT mode must be
> > supported. TLS exporters are not needed, nor are session
> > tickets. Basic key exchange is sufficient and implementations can
> > use any certificate. All MTI algorithms listed in TLS 1.3 are
> > expected.
>
> I note that this and the transport parameters and HRR stuff below
> necessitate the generic changes to the TLS library to permit adding
> extra extensions. I wonder if it might make life better to just have a
> canned set of parameters for now. It's not that it's a lot of work,
> it's just that it's not critical path otherwise.
>
>
> > Packet protection - All post handshake packets must be sent with
> > 1RTT keys and packet protection.
>
> This is inconsistent with not requiring exporters above. As far
> as I can tell, if you're not doing NST, you don't need post-handshake
> packets anyway here, so you can probably skip this.
>
>
>
> On Wed, May 10, 2017 at 9:53 PM, Mark Nottingham <mnot@mnot.net> wrote:
>
>> Previously, we've mentioned an intention to have a First Implementation
>> Draft -- that is, an Internet-Draft that we feel is suitable for
>> implementers to write code to, for the purposes of interoperability testing
>> and gathering feedback -- shortly after the Paris interim.
>>
>> Due to the size and complexity of HTTP-over-QUIC, implementing all four
>> drafts for this purpose on a reasonable timeline isn't workable.
>>
>> Instead, the editors have identified a subset of functionality that they
>> believe will serve as a suitable starting point. See:
>>   https://github.com/quicwg/base-drafts/wiki/First-Implementation-Draft
>>
>> They've also identified the set of issues that we believe will be
>> necessary to have proposals for before Paris; see:
>>   https://github.com/quicwg/base-drafts/milestone/1
>>
>> If all goes well, the plan is to have a set of drafts out (very) soon
>> that do so; then, we can discuss them and the First Implementation Draft
>> Candidate in the lead-up to and during the Paris interim.
>>
>> After Paris, we'll make any necessary adjustments to the documents and
>> publish another set, which will be the First Implementation Draft.
>>
>> Please comment / raise concerns / make suggestions on-list.
>>
>> Regards,
>>
>> --
>> Mark Nottingham   https://www.mnot.net/
>>
>>
>

--001a114a80fc1076d3054f7fdedb
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>I don&#39;t really know how I feel about this wiki pa=
ge.. without ratholing too deeply on it I think that while it tries to faci=
litate testing and interop, it treads pretty significantly into the land of=
 project management which is squarely out of bounds for the working group. =
I&#39;m willing to see how it goes but that&#39;s my concern - I would rath=
er the WG focuses on the complete picture while providing more of a forum f=
or pair (or more) wise testing.<br></div><div><br></div><div>anyhow wrt spe=
cifics -=C2=A0 ekr points out the 2 major inconsistencies.. tbh I would req=
uire exporters and the ability to do at least one non-0 stream. At that poi=
nt you have a hello world milestone, which is always the first stage worthy=
 of a toast. Its ok by me if the transport params are just profiled.<br></d=
iv></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sat, =
May 13, 2017 at 2:54 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex"><div dir=3D"ltr">A few comments on this propo=
sed list:<div><br></div><div><div>&gt; Integration with TLS 1.3 handshake -=
 The basic 1-RTT mode must be</div><div>&gt; supported. TLS exporters are n=
ot needed, nor are session</div><div>&gt; tickets. Basic key exchange is su=
fficient and implementations can</div><div>&gt; use any certificate. All MT=
I algorithms listed in TLS 1.3 are</div><div>&gt; expected.</div><div><br><=
/div><div>I note that this and the transport parameters and HRR stuff below=
</div><div>necessitate the generic changes to the TLS library to permit add=
ing</div><div>extra extensions. I wonder if it might make life better to ju=
st have a</div><div>canned set of parameters for now. It&#39;s not that it&=
#39;s a lot of work,</div><div>it&#39;s just that it&#39;s not critical pat=
h otherwise.</div><div><br></div><div><br></div><div>&gt; Packet protection=
 - All post handshake packets must be sent with</div><div>&gt; 1RTT keys an=
d packet protection.</div><div><br></div><div>This is inconsistent with not=
 requiring exporters above. As far</div><div>as I can tell, if you&#39;re n=
ot doing NST, you don&#39;t need post-handshake</div><div>packets anyway he=
re, so you can probably skip this.</div><div><br></div></div><div><br></div=
></div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Wed, May 10, 2017 at 9:53 PM, Mark Notting=
ham <span dir=3D"ltr">&lt;<a href=3D"mailto:mnot@mnot.net" target=3D"_blank=
">mnot@mnot.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Pre=
viously, we&#39;ve mentioned an intention to have a First Implementation Dr=
aft -- that is, an Internet-Draft that we feel is suitable for implementers=
 to write code to, for the purposes of interoperability testing and gatheri=
ng feedback -- shortly after the Paris interim.<br>
<br>
Due to the size and complexity of HTTP-over-QUIC, implementing all four dra=
fts for this purpose on a reasonable timeline isn&#39;t workable.<br>
<br>
Instead, the editors have identified a subset of functionality that they be=
lieve will serve as a suitable starting point. See:<br>
=C2=A0 <a href=3D"https://github.com/quicwg/base-drafts/wiki/First-Implemen=
tation-Draft" rel=3D"noreferrer" target=3D"_blank">https://github.com/quicw=
g/base<wbr>-drafts/wiki/First-Implementat<wbr>ion-Draft</a><br>
<br>
They&#39;ve also identified the set of issues that we believe will be neces=
sary to have proposals for before Paris; see:<br>
=C2=A0 <a href=3D"https://github.com/quicwg/base-drafts/milestone/1" rel=3D=
"noreferrer" target=3D"_blank">https://github.com/quicwg/base<wbr>-drafts/m=
ilestone/1</a><br>
<br>
If all goes well, the plan is to have a set of drafts out (very) soon that =
do so; then, we can discuss them and the First Implementation Draft Candida=
te in the lead-up to and during the Paris interim.<br>
<br>
After Paris, we&#39;ll make any necessary adjustments to the documents and =
publish another set, which will be the First Implementation Draft.<br>
<br>
Please comment / raise concerns / make suggestions on-list.<br>
<br>
Regards,<br>
<br>
--<br>
Mark Nottingham=C2=A0 =C2=A0<a href=3D"https://www.mnot.net/" rel=3D"norefe=
rrer" target=3D"_blank">https://www.mnot.net/</a><br>
<br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a114a80fc1076d3054f7fdedb--


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Subject: Re: Getting to a First Implementation Draft
To: Patrick McManus <pmcmanus@mozilla.com>
Cc: Eric Rescorla <ekr@rtfm.com>, Mark Nottingham <mnot@mnot.net>, IETF QUIC WG <quic@ietf.org>, Lars Eggert <lars@netapp.com>
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--001a113f3cbef13b75054f81b513
Content-Type: text/plain; charset="UTF-8"

Fixing the transport parameters sounds like a good simplification to me.

I pushed for transferring some information with the 1RTT keys in a late
draft, which likely resulted in the inconsistency.  My logic is that you
haven't really shown the handshake works unless you use the output of it.

Transferring data on a non-0 stream requires an application of some sort.
Do you have any suggestions for what that would be?  Or were you thinking
something of simple like "ping" and "pong"?

On Sun, May 14, 2017 at 2:11 PM, Patrick McManus <pmcmanus@mozilla.com>
wrote:

> I don't really know how I feel about this wiki page.. without ratholing
> too deeply on it I think that while it tries to facilitate testing and
> interop, it treads pretty significantly into the land of project management
> which is squarely out of bounds for the working group. I'm willing to see
> how it goes but that's my concern - I would rather the WG focuses on the
> complete picture while providing more of a forum for pair (or more) wise
> testing.
>
> anyhow wrt specifics -  ekr points out the 2 major inconsistencies.. tbh I
> would require exporters and the ability to do at least one non-0 stream. At
> that point you have a hello world milestone, which is always the first
> stage worthy of a toast. Its ok by me if the transport params are just
> profiled.
>
> On Sat, May 13, 2017 at 2:54 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>> A few comments on this proposed list:
>>
>> > Integration with TLS 1.3 handshake - The basic 1-RTT mode must be
>> > supported. TLS exporters are not needed, nor are session
>> > tickets. Basic key exchange is sufficient and implementations can
>> > use any certificate. All MTI algorithms listed in TLS 1.3 are
>> > expected.
>>
>> I note that this and the transport parameters and HRR stuff below
>> necessitate the generic changes to the TLS library to permit adding
>> extra extensions. I wonder if it might make life better to just have a
>> canned set of parameters for now. It's not that it's a lot of work,
>> it's just that it's not critical path otherwise.
>>
>>
>> > Packet protection - All post handshake packets must be sent with
>> > 1RTT keys and packet protection.
>>
>> This is inconsistent with not requiring exporters above. As far
>> as I can tell, if you're not doing NST, you don't need post-handshake
>> packets anyway here, so you can probably skip this.
>>
>>
>>
>> On Wed, May 10, 2017 at 9:53 PM, Mark Nottingham <mnot@mnot.net> wrote:
>>
>>> Previously, we've mentioned an intention to have a First Implementation
>>> Draft -- that is, an Internet-Draft that we feel is suitable for
>>> implementers to write code to, for the purposes of interoperability testing
>>> and gathering feedback -- shortly after the Paris interim.
>>>
>>> Due to the size and complexity of HTTP-over-QUIC, implementing all four
>>> drafts for this purpose on a reasonable timeline isn't workable.
>>>
>>> Instead, the editors have identified a subset of functionality that they
>>> believe will serve as a suitable starting point. See:
>>>   https://github.com/quicwg/base-drafts/wiki/First-Implementation-Draft
>>>
>>> They've also identified the set of issues that we believe will be
>>> necessary to have proposals for before Paris; see:
>>>   https://github.com/quicwg/base-drafts/milestone/1
>>>
>>> If all goes well, the plan is to have a set of drafts out (very) soon
>>> that do so; then, we can discuss them and the First Implementation Draft
>>> Candidate in the lead-up to and during the Paris interim.
>>>
>>> After Paris, we'll make any necessary adjustments to the documents and
>>> publish another set, which will be the First Implementation Draft.
>>>
>>> Please comment / raise concerns / make suggestions on-list.
>>>
>>> Regards,
>>>
>>> --
>>> Mark Nottingham   https://www.mnot.net/
>>>
>>>
>>
>

--001a113f3cbef13b75054f81b513
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Fixing the transport parameters sounds like a good simplif=
ication to me.<div><br></div><div>I pushed for transferring some informatio=
n with the 1RTT keys in a late draft, which likely resulted in the inconsis=
tency.=C2=A0 My logic is that you haven&#39;t really shown the handshake wo=
rks unless you use the output of it.</div><div><br></div><div>Transferring =
data on a non-0 stream requires an application of some sort.=C2=A0 Do you h=
ave any suggestions for what that would be?=C2=A0 Or were you thinking some=
thing of simple like &quot;ping&quot; and &quot;pong&quot;?</div></div><div=
 class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, May 14, 2017 =
at 2:11 PM, Patrick McManus <span dir=3D"ltr">&lt;<a href=3D"mailto:pmcmanu=
s@mozilla.com" target=3D"_blank">pmcmanus@mozilla.com</a>&gt;</span> wrote:=
<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-lef=
t:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>I don&#39;t really=
 know how I feel about this wiki page.. without ratholing too deeply on it =
I think that while it tries to facilitate testing and interop, it treads pr=
etty significantly into the land of project management which is squarely ou=
t of bounds for the working group. I&#39;m willing to see how it goes but t=
hat&#39;s my concern - I would rather the WG focuses on the complete pictur=
e while providing more of a forum for pair (or more) wise testing.<br></div=
><div><br></div><div>anyhow wrt specifics -=C2=A0 ekr points out the 2 majo=
r inconsistencies.. tbh I would require exporters and the ability to do at =
least one non-0 stream. At that point you have a hello world milestone, whi=
ch is always the first stage worthy of a toast. Its ok by me if the transpo=
rt params are just profiled.<br></div></div><div class=3D"HOEnZb"><div clas=
s=3D"h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sat, =
May 13, 2017 at 2:54 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex"><div dir=3D"ltr">A few comments on this propo=
sed list:<div><br></div><div><div>&gt; Integration with TLS 1.3 handshake -=
 The basic 1-RTT mode must be</div><div>&gt; supported. TLS exporters are n=
ot needed, nor are session</div><div>&gt; tickets. Basic key exchange is su=
fficient and implementations can</div><div>&gt; use any certificate. All MT=
I algorithms listed in TLS 1.3 are</div><div>&gt; expected.</div><div><br><=
/div><div>I note that this and the transport parameters and HRR stuff below=
</div><div>necessitate the generic changes to the TLS library to permit add=
ing</div><div>extra extensions. I wonder if it might make life better to ju=
st have a</div><div>canned set of parameters for now. It&#39;s not that it&=
#39;s a lot of work,</div><div>it&#39;s just that it&#39;s not critical pat=
h otherwise.</div><div><br></div><div><br></div><div>&gt; Packet protection=
 - All post handshake packets must be sent with</div><div>&gt; 1RTT keys an=
d packet protection.</div><div><br></div><div>This is inconsistent with not=
 requiring exporters above. As far</div><div>as I can tell, if you&#39;re n=
ot doing NST, you don&#39;t need post-handshake</div><div>packets anyway he=
re, so you can probably skip this.</div><div><br></div></div><div><br></div=
></div><div class=3D"m_-2442783920927145455HOEnZb"><div class=3D"m_-2442783=
920927145455h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">O=
n Wed, May 10, 2017 at 9:53 PM, Mark Nottingham <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:mnot@mnot.net" target=3D"_blank">mnot@mnot.net</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">Previously, we&#39;ve mentioned an=
 intention to have a First Implementation Draft -- that is, an Internet-Dra=
ft that we feel is suitable for implementers to write code to, for the purp=
oses of interoperability testing and gathering feedback -- shortly after th=
e Paris interim.<br>
<br>
Due to the size and complexity of HTTP-over-QUIC, implementing all four dra=
fts for this purpose on a reasonable timeline isn&#39;t workable.<br>
<br>
Instead, the editors have identified a subset of functionality that they be=
lieve will serve as a suitable starting point. See:<br>
=C2=A0 <a href=3D"https://github.com/quicwg/base-drafts/wiki/First-Implemen=
tation-Draft" rel=3D"noreferrer" target=3D"_blank">https://github.com/quicw=
g/base<wbr>-drafts/wiki/First-Implementat<wbr>ion-Draft</a><br>
<br>
They&#39;ve also identified the set of issues that we believe will be neces=
sary to have proposals for before Paris; see:<br>
=C2=A0 <a href=3D"https://github.com/quicwg/base-drafts/milestone/1" rel=3D=
"noreferrer" target=3D"_blank">https://github.com/quicwg/base<wbr>-drafts/m=
ilestone/1</a><br>
<br>
If all goes well, the plan is to have a set of drafts out (very) soon that =
do so; then, we can discuss them and the First Implementation Draft Candida=
te in the lead-up to and during the Paris interim.<br>
<br>
After Paris, we&#39;ll make any necessary adjustments to the documents and =
publish another set, which will be the First Implementation Draft.<br>
<br>
Please comment / raise concerns / make suggestions on-list.<br>
<br>
Regards,<br>
<br>
--<br>
Mark Nottingham=C2=A0 =C2=A0<a href=3D"https://www.mnot.net/" rel=3D"norefe=
rrer" target=3D"_blank">https://www.mnot.net/</a><br>
<br>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a113f3cbef13b75054f81b513--


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Subject: Re: Getting to a First Implementation Draft
To: Ian Swett <ianswett@google.com>
Cc: Patrick McManus <pmcmanus@mozilla.com>, Eric Rescorla <ekr@rtfm.com>, Mark Nottingham <mnot@mnot.net>,  IETF QUIC WG <quic@ietf.org>, Lars Eggert <lars@netapp.com>
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On Sun, May 14, 2017 at 4:23 PM, Ian Swett <ianswett@google.com> wrote:

> Or were you thinking something of simple like "ping" and "pong"?
>


yep.

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Sun, May 14, 2017 at 4:23 PM, Ian Swett <span dir=3D"ltr">&lt;<a href=3D=
"mailto:ianswett@google.com" target=3D"_blank">ianswett@google.com</a>&gt;<=
/span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div> Or w=
ere you thinking something of simple like &quot;ping&quot; and &quot;pong&q=
uot;?</div></div><div class=3D"yj6qo ajU"><div id=3D":18e" class=3D"ajR" ta=
bindex=3D"0"></div></div></blockquote></div><br><br></div><div class=3D"gma=
il_extra">yep.</div></div>

--001a11404b9ca34b0f054f81eff9--


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Subject: Re: Getting to a First Implementation Draft
To: Patrick McManus <pmcmanus@mozilla.com>
Cc: Ian Swett <ianswett@google.com>, Mark Nottingham <mnot@mnot.net>, Eric Rescorla <ekr@rtfm.com>,  IETF QUIC WG <quic@ietf.org>, Lars Eggert <lars@netapp.com>
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--001a11456f4a4ffda9054f82d5bd
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Adding a PING frame + ACK seems like a good idea.

On Sun, May 14, 2017 at 1:39 PM, Patrick McManus <pmcmanus@mozilla.com>
wrote:

>
> On Sun, May 14, 2017 at 4:23 PM, Ian Swett <ianswett@google.com> wrote:
>
>> Or were you thinking something of simple like "ping" and "pong"?
>>
>
>
> yep.
>

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<div dir=3D"ltr">Adding a PING frame + ACK seems like a good idea.</div><di=
v class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, May 14, 2017=
 at 1:39 PM, Patrick McManus <span dir=3D"ltr">&lt;<a href=3D"mailto:pmcman=
us@mozilla.com" target=3D"_blank">pmcmanus@mozilla.com</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span class=3D""><div =
class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Sun, May 14, 2017 a=
t 4:23 PM, Ian Swett <span dir=3D"ltr">&lt;<a href=3D"mailto:ianswett@googl=
e.com" target=3D"_blank">ianswett@google.com</a>&gt;</span> wrote:<br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex"><div dir=3D"ltr"><div> Or were you thinking somet=
hing of simple like &quot;ping&quot; and &quot;pong&quot;?</div></div><div =
class=3D"m_453359946080365841yj6qo m_453359946080365841ajU"><div id=3D"m_45=
3359946080365841:18e" class=3D"m_453359946080365841ajR"></div></div></block=
quote></div><br><br></div></span><div class=3D"gmail_extra">yep.</div></div=
>
</blockquote></div><br></div>

--001a11456f4a4ffda9054f82d5bd--


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From: Mark Nottingham <mnot@mnot.net>
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Subject: Remote participation in the June QUIC Interim meeting
Date: Mon, 15 May 2017 14:11:17 +1000
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Cc: Lars Eggert <lars@netapp.com>
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If you registered to participate in the June interim meeting remotely, =
you should now have an e-mail with the details from WebEx.=20

If you did but don't receive such an e-mail in the next few hours, =
please reply to this e-mail (just to me) and we'll sort it out.

Regards,

--
Mark Nottingham   https://www.mnot.net/


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From: Lucas Clemente <luke.clemente@gmail.com>
Date: Mon, 15 May 2017 08:02:03 +0000
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Subject: Re: Getting to a First Implementation Draft
To: Jana Iyengar <jri@google.com>, Patrick McManus <pmcmanus@mozilla.com>
Cc: Ian Swett <ianswett@google.com>, IETF QUIC WG <quic@ietf.org>, Lars Eggert <lars@netapp.com>,  Mark Nottingham <mnot@mnot.net>, Eric Rescorla <ekr@rtfm.com>, Marten Seemann <martenseemann@gmail.com>
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--001a11490122166929054f8b798f
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Another idea would be to just have a single stream and "emulate" TCP+TLS
using that. We used something similar
<https://github.com/marten-seemann/quic-conn> in the past to refine our
API, and this gave us a lot of input for that.

This would probably add flow control to the scope, unless we just manually
set the window to something huge for these initial tests.

On Sun, 14 May 2017 at 23:47 Jana Iyengar <jri@google.com> wrote:

> Adding a PING frame + ACK seems like a good idea.
>
> On Sun, May 14, 2017 at 1:39 PM, Patrick McManus <pmcmanus@mozilla.com>
> wrote:
>
>>
>> On Sun, May 14, 2017 at 4:23 PM, Ian Swett <ianswett@google.com> wrote:
>>
>>> Or were you thinking something of simple like "ping" and "pong"?
>>>
>>
>>
>> yep.
>>
>
>

--001a11490122166929054f8b798f
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Another idea would be to just have a single stream and &qu=
ot;emulate&quot; TCP+TLS using that. We used something <a href=3D"https://g=
ithub.com/marten-seemann/quic-conn">similar</a> in the past to refine our A=
PI, and this gave us a lot of input for that.<div><br></div><div>This would=
 probably add flow control to the scope, unless we just manually set the wi=
ndow to something huge for these initial tests.</div><div><br><div class=3D=
"gmail_quote"><div dir=3D"ltr">On Sun, 14 May 2017 at 23:47 Jana Iyengar &l=
t;<a href=3D"mailto:jri@google.com">jri@google.com</a>&gt; wrote:<br></div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">Adding a PING frame + ACK s=
eems like a good idea.</div><div class=3D"gmail_extra"><br><div class=3D"gm=
ail_quote">On Sun, May 14, 2017 at 1:39 PM, Patrick McManus <span dir=3D"lt=
r">&lt;<a href=3D"mailto:pmcmanus@mozilla.com" target=3D"_blank">pmcmanus@m=
ozilla.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"ltr"><span><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On=
 Sun, May 14, 2017 at 4:23 PM, Ian Swett <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:ianswett@google.com" target=3D"_blank">ianswett@google.com</a>&gt;</s=
pan> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div> Or wer=
e you thinking something of simple like &quot;ping&quot; and &quot;pong&quo=
t;?</div></div><div class=3D"m_5359792189250674884m_453359946080365841yj6qo=
 m_5359792189250674884m_453359946080365841ajU"><div id=3D"m_535979218925067=
4884m_453359946080365841:18e" class=3D"m_5359792189250674884m_4533599460803=
65841ajR"></div></div></blockquote></div><br><br></div></span><div class=3D=
"gmail_extra">yep.</div></div>
</blockquote></div><br></div>
</blockquote></div></div></div>

--001a11490122166929054f8b798f--


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Subject: Potential conflict between draft-ietf-quic-transport and RFC 7983
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There appears to be a potential conflict between draft-ietf-quic-transport
and RFC 7983. Looking at draft-ietf-quic-transport, both the short and the
long headers appear to conflict with the de-multiplexing scheme defined in
RFC 7983, which is based on the value of the first byte of multi-plexed
protocols:

   The process for demultiplexing a packet is as follows.  The receiver
   looks at the first byte of the packet.  If the value of this byte is
   in between 0 and 3 (inclusive), then the packet is STUN.  If the
   value is between 16 and 19 (inclusive), then the packet is ZRTP.  If
   the value is between 20 and 63 (inclusive), then the packet is DTLS.
   If the value is between 64 and 79 (inclusive), then the packet is
   TURN Channel.  If the value is in between 128 and 191 (inclusive),
   then the packet is RTP (or RTCP, if both RTCP and RTP are being
   multiplexed over the same destination port).  If the value does not
   match any known range, then the packet MUST be dropped and an alert
   MAY be logged.  This process is summarized in Figure 3.

                    +----------------+
                    |        [0..3] -+--> forward to STUN
                    |                |
                    |      [16..19] -+--> forward to ZRTP
                    |                |
        packet -->  |      [20..63] -+--> forward to DTLS
                    |                |
                    |      [64..79] -+--> forward to TURN Channel
                    |                |
                    |    [128..191] -+--> forward to RTP/RTCP
                    +----------------+

     Figure 3: The DTLS-SRTP receiver's packet demultiplexing algorithm.

--94eb2c14970c037594054fbf17dd
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<div dir=3D"ltr">There appears to be a potential conflict between draft-iet=
f-quic-transport and=C2=A0RFC 7983. Looking at draft-ietf-quic-transport, b=
oth the short and the long headers appear to conflict with the de-multiplex=
ing scheme defined in RFC 7983, which is based on the value of the first by=
te of multi-plexed protocols:=C2=A0<div><div><br></div><div><pre class=3D"g=
mail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px=
;color:rgb(0,0,0)">   The process for demultiplexing a packet is as follows=
.  The receiver
   looks at the first byte of the packet.  If the value of this byte is
   in between 0 and 3 (inclusive), then the packet is STUN.  If the
   value is between 16 and 19 (inclusive), then the packet is ZRTP.  If
   the value is between 20 and 63 (inclusive), then the packet is DTLS.
   If the value is between 64 and 79 (inclusive), then the packet is
   TURN Channel.  If the value is in between 128 and 191 (inclusive),
   then the packet is RTP (or RTCP, if both RTCP and RTP are being
   multiplexed over the same destination port).  If the value does not
   match any known range, then the packet MUST be dropped and an alert
   MAY be logged.  This process is summarized in Figure 3.

                    +----------------+
                    |        [0..3] -+--&gt; forward to STUN
                    |                |
                    |      [16..19] -+--&gt; forward to ZRTP
                    |                |
        packet --&gt;  |      [20..63] -+--&gt; forward to DTLS
                    |                |
                    |      [64..79] -+--&gt; forward to TURN Channel
                    |                |
                    |    [128..191] -+--&gt; forward to RTP/RTCP
                    +----------------+

     Figure 3: The DTLS-SRTP receiver&#39;s packet demultiplexing algorithm=
.</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top=
:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"gmail-newp=
age" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rg=
b(0,0,0)"><br></pre></div></div></div>

--94eb2c14970c037594054fbf17dd--


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Subject: RE: Potential conflict between draft-ietf-quic-transport and RFC 7983
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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Wed, 17 May 2017 15:31:21 -0700
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Subject: Re: Potential conflict between draft-ietf-quic-transport and RFC 7983
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Mike asked:

"Is it a goal for QUIC and DTLS-SRTP to coexist on the same port"?

[BA] RFC 7983 enables DTLS, SRTP/SRTCP, ZRTP, STUN and TURN to co-exist on
the same port.  Within the scheme, DTLS is used for transport of WebRTC's
DTLS/SCTP/UDP data channel and DTLS-SRTP is used for key management of
SRTP/SRTCP.

While within WebRTC, DTLS currently provides both data channel transport as
well as SRTP key management, it is not clear that initial QUIC
implementations within WebRTC will attempt to provide alternatives for both
of these functions.

For example, experimental implementations of a QUIC data channel may wish
to focus solely on that objective without having to provide an alternative
to DTLS-SRTP.  In such an implementation, QUIC and DTLS-SRTP would need to
co-exist on the same port.



On Wed, May 17, 2017 at 2:52 PM, Mike Bishop <Michael.Bishop@microsoft.com>
wrote:

> 7983 is a bugfix to 5764 section 5.1.2, per a quick perusal.  I didn=E2=
=80=99t
> read either of those RFCs as purporting to constrain the first byte of al=
l
> future UDP-based protocols.  5764 says:
>
>
>
>    When DTLS-SRTP is used to protect an RTP session, the RTP receiver
>
>    needs to demultiplex packets that are arriving on the RTP port.
>
> ...
>
>    If other packet types are to be multiplexed as well, implementors
>
>    and/or designers SHOULD ensure that they can be demultiplexed from
>
>    these three *[or five, given 7983]* packet types.
>
>
>
> Is it a goal for QUIC and DTLS-SRTP to coexist on the same port?
>
>
>
> *From:* QUIC [mailto:quic-bounces@ietf.org] *On Behalf Of *Bernard Aboba
> *Sent:* Wednesday, May 17, 2017 2:37 PM
> *To:* quic@ietf.org
> *Cc:* marc@petit-huguenin.org; pthatcher@google.com
> *Subject:* Potential conflict between draft-ietf-quic-transport and RFC
> 7983
>
>
>
> There appears to be a potential conflict between draft-ietf-quic-transpor=
t
> and RFC 7983. Looking at draft-ietf-quic-transport, both the short and th=
e
> long headers appear to conflict with the de-multiplexing scheme defined i=
n
> RFC 7983, which is based on the value of the first byte of multi-plexed
> protocols:
>
>
>
>    The process for demultiplexing a packet is as follows.  The receiver
>
>    looks at the first byte of the packet.  If the value of this byte is
>
>    in between 0 and 3 (inclusive), then the packet is STUN.  If the
>
>    value is between 16 and 19 (inclusive), then the packet is ZRTP.  If
>
>    the value is between 20 and 63 (inclusive), then the packet is DTLS.
>
>    If the value is between 64 and 79 (inclusive), then the packet is
>
>    TURN Channel.  If the value is in between 128 and 191 (inclusive),
>
>    then the packet is RTP (or RTCP, if both RTCP and RTP are being
>
>    multiplexed over the same destination port).  If the value does not
>
>    match any known range, then the packet MUST be dropped and an alert
>
>    MAY be logged.  This process is summarized in Figure 3.
>
>
>
>                     +----------------+
>
>                     |        [0..3] -+--> forward to STUN
>
>                     |                |
>
>                     |      [16..19] -+--> forward to ZRTP
>
>                     |                |
>
>         packet -->  |      [20..63] -+--> forward to DTLS
>
>                     |                |
>
>                     |      [64..79] -+--> forward to TURN Channel
>
>                     |                |
>
>                     |    [128..191] -+--> forward to RTP/RTCP
>
>                     +----------------+
>
>
>
>      Figure 3: The DTLS-SRTP receiver's packet demultiplexing algorithm.
>
>
>
>
>
>

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<div dir=3D"ltr">Mike asked:=C2=A0<div><br></div><div>&quot;Is it a goal fo=
r QUIC and DTLS-SRTP to coexist on the same port&quot;?=C2=A0</div><div><br=
></div><div>[BA] RFC 7983 enables DTLS, SRTP/SRTCP, ZRTP, STUN and TURN to =
co-exist on the same port.=C2=A0 Within the scheme, DTLS is used for transp=
ort of WebRTC&#39;s DTLS/SCTP/UDP data channel and DTLS-SRTP is used for ke=
y management of SRTP/SRTCP. =C2=A0<br></div><div><br></div><div>While withi=
n WebRTC, DTLS currently provides both data channel transport as well as SR=
TP key management, it is not clear that initial QUIC implementations within=
 WebRTC will attempt to provide alternatives for both of these functions. =
=C2=A0</div><div><br></div><div>For example, experimental implementations o=
f a QUIC data channel may wish to focus solely on that objective without ha=
ving to provide an alternative to DTLS-SRTP.=C2=A0 In such an implementatio=
n, QUIC and DTLS-SRTP would need to co-exist on the same port.=C2=A0</div><=
div><br></div><div><br></div></div><div class=3D"gmail_extra"><br><div clas=
s=3D"gmail_quote">On Wed, May 17, 2017 at 2:52 PM, Mike Bishop <span dir=3D=
"ltr">&lt;<a href=3D"mailto:Michael.Bishop@microsoft.com" target=3D"_blank"=
>Michael.Bishop@microsoft.com</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">





<div lang=3D"EN-US" link=3D"#0563C1" vlink=3D"#954F72">
<div class=3D"m_5980825313082688534WordSection1">
<p class=3D"MsoNormal">7983 is a bugfix to 5764 section 5.1.2, per a quick =
perusal.=C2=A0 I didn=E2=80=99t read either of those RFCs as purporting to =
constrain the first byte of all future UDP-based protocols.=C2=A0 5764 says=
:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0 When DTLS-SRTP is used to protect=
 an RTP session, the RTP receiver<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0 needs to demultiplex packets that=
 are arriving on the RTP port.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">...<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0
<span style=3D"background:yellow">If other packet types are to be multiplex=
ed as well</span>, implementors<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0 and/or designers SHOULD ensure th=
at they can be demultiplexed from<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0 these three
</span><i><span style=3D"color:black">[or five, given 7983]</span></i><span=
 style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:black"=
> packet types.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal">Is it a goal for QUIC and DTLS-SRTP to coexist on th=
e same port?<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><b>From:</b> QUIC [mailto:<a href=3D"mailto:quic-bou=
nces@ietf.org" target=3D"_blank">quic-bounces@ietf.org</a>] <b>On Behalf Of
</b>Bernard Aboba<br>
<b>Sent:</b> Wednesday, May 17, 2017 2:37 PM<br>
<b>To:</b> <a href=3D"mailto:quic@ietf.org" target=3D"_blank">quic@ietf.org=
</a><br>
<b>Cc:</b> <a href=3D"mailto:marc@petit-huguenin.org" target=3D"_blank">mar=
c@petit-huguenin.org</a>; <a href=3D"mailto:pthatcher@google.com" target=3D=
"_blank">pthatcher@google.com</a><br>
<b>Subject:</b> Potential conflict between draft-ietf-quic-transport and RF=
C 7983<u></u><u></u></p><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">There appears to be a potential conflict between dra=
ft-ietf-quic-transport and=C2=A0RFC 7983. Looking at draft-ietf-quic-transp=
ort, both the short and the long headers appear to conflict with the de-mul=
tiplexing scheme defined in RFC 7983, which
 is based on the value of the first byte of multi-plexed protocols:=C2=A0<u=
></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<pre><span style=3D"color:black">=C2=A0=C2=A0 The process for demultiplexin=
g a packet is as follows.=C2=A0 The receiver<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 looks at the first byte of th=
e packet.=C2=A0 If the value of this byte is<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 in between 0 and 3 (inclusive=
), then the packet is STUN.=C2=A0 If the<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 value is between 16 and 19 (i=
nclusive), then the packet is ZRTP.=C2=A0 If<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 the value is between 20 and 6=
3 (inclusive), then the packet is DTLS.<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 If the value is between 64 an=
d 79 (inclusive), then the packet is<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 TURN Channel.=C2=A0 If the va=
lue is in between 128 and 191 (inclusive),<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 then the packet is RTP (or RT=
CP, if both RTCP and RTP are being<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 multiplexed over the same des=
tination port).=C2=A0 If the value does not<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 match any known range, then t=
he packet MUST be dropped and an alert<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 MAY be logged.=C2=A0 This pro=
cess is summarized in Figure 3.<u></u><u></u></span></pre>
<pre><span style=3D"color:black"><u></u>=C2=A0<u></u></span></pre>
<pre><span style=3D"color:black">=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 +-=
---------------+<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0 [0..3] -+--&gt; forward to STUN<=
u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 |<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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 [16..19] -+--&gt; forward to ZRTP<u></u><u><=
/u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 |<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 packet --&gt;=C2=A0 |=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 [20..63] -+--&gt; forw=
ard to DTLS<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 |<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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 [64..79] -+--&gt; forward to TURN Channel<u>=
</u><u></u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 |<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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 [128..191] -+--&gt; forward to RTP/RTCP<u></u><u></u></s=
pan></pre>
<pre><span style=3D"color:black">=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 +-=
---------------+<u></u><u></u></span></pre>
<pre><span style=3D"color:black"><u></u>=C2=A0<u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0=C2=A0=C2=A0 Figure 3: The DTL=
S-SRTP receiver&#39;s packet demultiplexing algorithm.<u></u><u></u></span>=
</pre>
<pre><span style=3D"color:black"><u></u>=C2=A0<u></u></span></pre>
<pre><span style=3D"color:black"><u></u>=C2=A0<u></u></span></pre>
</div>
</div>
</div>
</div></div></div>
</div>

</blockquote></div><br></div>

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From: Jana Iyengar <jri@google.com>
Date: Wed, 17 May 2017 18:02:43 -0700
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Subject: Re: Potential conflict between draft-ietf-quic-transport and RFC 7983
To: Bernard Aboba <bernard.aboba@gmail.com>
Cc: Mike Bishop <Michael.Bishop@microsoft.com>,  "marc@petit-huguenin.org" <marc@petit-huguenin.org>, "quic@ietf.org" <quic@ietf.org>,  "pthatcher@google.com" <pthatcher@google.com>
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This *may* be one of those things where keeping WebRTC in the back of our
minds might help QUIC not get stuck badly in the future.

So, as a thought exercise (not assuming that we need to solve this problem
here) I'd like to walk through what we *could* do if we actually supported
both WebRTC protocols and QUIC on the same port.

I think the simplest thing to do would be to pass everything through a QUIC
"dispatcher" that determines if this packet belongs to QUIC --- basically
confirms whether this packet should be consumed by QUIC or not --- and if
it isn't consumed by QUIC (including packets that QUIC decides MUST be
dropped because they are QUIC packets but violate some property) then it
can be consumed by non-QUIC applications on the same port.

The question of course is how bad is the likelihood of false
positives/negatives. With the encrypted packet types in QUIC, the QUIC
dispatcher will safely reject non-QUIC packets.

With the plaintext types, which are used during handshake, QUIC packets are
covered by a non-crypto FNV-1a hash. The cost that QUIC bears of a
mis-classification among plaintext packets as a QUIC packet is limited to
cases where the contents of the packet are not part of the final key
derivation. These packet types are limited to a few long-form packet types,
all of which have a number of version-independent fields upfront, including
version, which could be checked for sanity (since connection ID and packet
number can be arbitrary.) This still doesn't seem great, since 0x0001 is a
fine version number to have in QUIC and may be a completely reasonable
string in RTP.  So alternatively, we could renumber the long-form packet
types to start at 255 and count down, since 192 - 255 is left open by RFC
7983, and since we just need  6 cleartext packet types at the moment to be
distinct from the RTP packet types (it's smaller than that, since Stateless
Reject and Version Negotiation packets need to echo stuff from the client's
packets.)

TL;DR of this is that *if* we wanted to allow for a future coexistence of
WebRTC protocols with QUIC on the same port -- and this is a big "if",
since I wonder if they need to be on the same port --- we could renumber
just the long-form packets to start at 255 and grow downwards. We only need
<=3D 6 plaintext packet types to be out of the collision region with  RFC
7983.


On Wed, May 17, 2017 at 3:31 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> Mike asked:
>
> "Is it a goal for QUIC and DTLS-SRTP to coexist on the same port"?
>
> [BA] RFC 7983 enables DTLS, SRTP/SRTCP, ZRTP, STUN and TURN to co-exist o=
n
> the same port.  Within the scheme, DTLS is used for transport of WebRTC's
> DTLS/SCTP/UDP data channel and DTLS-SRTP is used for key management of
> SRTP/SRTCP.
>
> While within WebRTC, DTLS currently provides both data channel transport
> as well as SRTP key management, it is not clear that initial QUIC
> implementations within WebRTC will attempt to provide alternatives for bo=
th
> of these functions.
>
> For example, experimental implementations of a QUIC data channel may wish
> to focus solely on that objective without having to provide an alternativ=
e
> to DTLS-SRTP.  In such an implementation, QUIC and DTLS-SRTP would need t=
o
> co-exist on the same port.
>
>
>
> On Wed, May 17, 2017 at 2:52 PM, Mike Bishop <Michael.Bishop@microsoft.co=
m
> > wrote:
>
>> 7983 is a bugfix to 5764 section 5.1.2, per a quick perusal.  I didn=E2=
=80=99t
>> read either of those RFCs as purporting to constrain the first byte of a=
ll
>> future UDP-based protocols.  5764 says:
>>
>>
>>
>>    When DTLS-SRTP is used to protect an RTP session, the RTP receiver
>>
>>    needs to demultiplex packets that are arriving on the RTP port.
>>
>> ...
>>
>>    If other packet types are to be multiplexed as well, implementors
>>
>>    and/or designers SHOULD ensure that they can be demultiplexed from
>>
>>    these three *[or five, given 7983]* packet types.
>>
>>
>>
>> Is it a goal for QUIC and DTLS-SRTP to coexist on the same port?
>>
>>
>>
>> *From:* QUIC [mailto:quic-bounces@ietf.org] *On Behalf Of *Bernard Aboba
>> *Sent:* Wednesday, May 17, 2017 2:37 PM
>> *To:* quic@ietf.org
>> *Cc:* marc@petit-huguenin.org; pthatcher@google.com
>> *Subject:* Potential conflict between draft-ietf-quic-transport and RFC
>> 7983
>>
>>
>>
>> There appears to be a potential conflict between
>> draft-ietf-quic-transport and RFC 7983. Looking at
>> draft-ietf-quic-transport, both the short and the long headers appear to
>> conflict with the de-multiplexing scheme defined in RFC 7983, which is
>> based on the value of the first byte of multi-plexed protocols:
>>
>>
>>
>>    The process for demultiplexing a packet is as follows.  The receiver
>>
>>    looks at the first byte of the packet.  If the value of this byte is
>>
>>    in between 0 and 3 (inclusive), then the packet is STUN.  If the
>>
>>    value is between 16 and 19 (inclusive), then the packet is ZRTP.  If
>>
>>    the value is between 20 and 63 (inclusive), then the packet is DTLS.
>>
>>    If the value is between 64 and 79 (inclusive), then the packet is
>>
>>    TURN Channel.  If the value is in between 128 and 191 (inclusive),
>>
>>    then the packet is RTP (or RTCP, if both RTCP and RTP are being
>>
>>    multiplexed over the same destination port).  If the value does not
>>
>>    match any known range, then the packet MUST be dropped and an alert
>>
>>    MAY be logged.  This process is summarized in Figure 3.
>>
>>
>>
>>                     +----------------+
>>
>>                     |        [0..3] -+--> forward to STUN
>>
>>                     |                |
>>
>>                     |      [16..19] -+--> forward to ZRTP
>>
>>                     |                |
>>
>>         packet -->  |      [20..63] -+--> forward to DTLS
>>
>>                     |                |
>>
>>                     |      [64..79] -+--> forward to TURN Channel
>>
>>                     |                |
>>
>>                     |    [128..191] -+--> forward to RTP/RTCP
>>
>>                     +----------------+
>>
>>
>>
>>      Figure 3: The DTLS-SRTP receiver's packet demultiplexing algorithm.
>>
>>
>>
>>
>>
>>
>

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<div dir=3D"ltr"><div>This *may* be one of those things where keeping WebRT=
C in the back of our minds might help QUIC not get stuck badly in the futur=
e.</div><div><br></div><div>So, as a thought exercise (not assuming that we=
 need to solve this problem here) I&#39;d like to walk through what we *cou=
ld* do if we actually supported both WebRTC protocols and QUIC on the same =
port.</div><div><br></div><div>I think the simplest thing to do would be to=
 pass everything through a QUIC &quot;dispatcher&quot; that determines if t=
his packet belongs to QUIC --- basically confirms whether this packet shoul=
d be consumed by QUIC or not --- and if it isn&#39;t consumed by QUIC (incl=
uding packets that QUIC decides MUST be dropped because they are QUIC packe=
ts but violate some property) then it can be consumed by non-QUIC applicati=
ons on the same port.</div><div><br></div><div>The question of course is ho=
w bad is the likelihood of false positives/negatives. With the encrypted pa=
cket types in QUIC, the QUIC dispatcher will safely reject non-QUIC packets=
.=C2=A0</div><div><br></div><div>With the plaintext types, which are used d=
uring handshake, QUIC packets are covered by a non-crypto FNV-1a hash. The =
cost that QUIC bears of a mis-classification among plaintext packets as a Q=
UIC packet is limited to cases where the contents of the packet are not par=
t of the final key derivation. These packet types are limited to a few long=
-form packet types, all of which have a number of version-independent field=
s upfront, including version, which could be checked for sanity (since conn=
ection ID and packet number can be arbitrary.) This still doesn&#39;t seem =
great, since 0x0001 is a fine version number to have in QUIC and may be a c=
ompletely reasonable string in RTP.=C2=A0 So alternatively, we could renumb=
er the long-form packet types to start at 255 and count down, since 192 - 2=
55 is left open by RFC 7983, and since we just need =C2=A06 cleartext packe=
t types at the moment to be distinct from the RTP packet types (it&#39;s sm=
aller than that, since Stateless Reject and Version Negotiation packets nee=
d to echo stuff from the client&#39;s packets.)</div><div><br></div><div>TL=
;DR of this is that *if* we wanted to allow for a future coexistence of Web=
RTC protocols with QUIC on the same port -- and this is a big &quot;if&quot=
;, since I wonder if they need to be on the same port --- we could renumber=
 just the long-form packets to start at 255 and grow downwards. We only nee=
d &lt;=3D 6 plaintext packet types to be out of the collision region with =
=C2=A0RFC 7983.</div><div><br></div></div><div class=3D"gmail_extra"><br><d=
iv class=3D"gmail_quote">On Wed, May 17, 2017 at 3:31 PM, Bernard Aboba <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_bl=
ank">bernard.aboba@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex"><div dir=3D"ltr">Mike asked:=C2=A0<span class=3D""><div><br></div=
><div>&quot;Is it a goal for QUIC and DTLS-SRTP to coexist on the same port=
&quot;?=C2=A0</div><div><br></div></span><div>[BA] RFC 7983 enables DTLS, S=
RTP/SRTCP, ZRTP, STUN and TURN to co-exist on the same port.=C2=A0 Within t=
he scheme, DTLS is used for transport of WebRTC&#39;s DTLS/SCTP/UDP data ch=
annel and DTLS-SRTP is used for key management of SRTP/SRTCP. =C2=A0<br></d=
iv><div><br></div><div>While within WebRTC, DTLS currently provides both da=
ta channel transport as well as SRTP key management, it is not clear that i=
nitial QUIC implementations within WebRTC will attempt to provide alternati=
ves for both of these functions. =C2=A0</div><div><br></div><div>For exampl=
e, experimental implementations of a QUIC data channel may wish to focus so=
lely on that objective without having to provide an alternative to DTLS-SRT=
P.=C2=A0 In such an implementation, QUIC and DTLS-SRTP would need to co-exi=
st on the same port.=C2=A0</div><div><br></div><div><br></div></div><div cl=
ass=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Wed, May 17, 2017 at 2:52 PM, Mike Bishop <span dir=3D"=
ltr">&lt;<a href=3D"mailto:Michael.Bishop@microsoft.com" target=3D"_blank">=
Michael.Bishop@microsoft.com</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">





<div lang=3D"EN-US" link=3D"#0563C1" vlink=3D"#954F72">
<div class=3D"m_7572897580995963207m_5980825313082688534WordSection1">
<p class=3D"MsoNormal">7983 is a bugfix to 5764 section 5.1.2, per a quick =
perusal.=C2=A0 I didn=E2=80=99t read either of those RFCs as purporting to =
constrain the first byte of all future UDP-based protocols.=C2=A0 5764 says=
:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0 When DTLS-SRTP is used to protect=
 an RTP session, the RTP receiver<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0 needs to demultiplex packets that=
 are arriving on the RTP port.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">...<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0
<span style=3D"background:yellow">If other packet types are to be multiplex=
ed as well</span>, implementors<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0 and/or designers SHOULD ensure th=
at they can be demultiplexed from<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=C2=A0=C2=A0 these three
</span><i><span style=3D"color:black">[or five, given 7983]</span></i><span=
 style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:black"=
> packet types.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal">Is it a goal for QUIC and DTLS-SRTP to coexist on th=
e same port?<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><b>From:</b> QUIC [mailto:<a href=3D"mailto:quic-bou=
nces@ietf.org" target=3D"_blank">quic-bounces@ietf.org</a>] <b>On Behalf Of
</b>Bernard Aboba<br>
<b>Sent:</b> Wednesday, May 17, 2017 2:37 PM<br>
<b>To:</b> <a href=3D"mailto:quic@ietf.org" target=3D"_blank">quic@ietf.org=
</a><br>
<b>Cc:</b> <a href=3D"mailto:marc@petit-huguenin.org" target=3D"_blank">mar=
c@petit-huguenin.org</a>; <a href=3D"mailto:pthatcher@google.com" target=3D=
"_blank">pthatcher@google.com</a><br>
<b>Subject:</b> Potential conflict between draft-ietf-quic-transport and RF=
C 7983<u></u><u></u></p><div><div class=3D"m_7572897580995963207h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">There appears to be a potential conflict between dra=
ft-ietf-quic-transport and=C2=A0RFC 7983. Looking at draft-ietf-quic-transp=
ort, both the short and the long headers appear to conflict with the de-mul=
tiplexing scheme defined in RFC 7983, which
 is based on the value of the first byte of multi-plexed protocols:=C2=A0<u=
></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<pre><span style=3D"color:black">=C2=A0=C2=A0 The process for demultiplexin=
g a packet is as follows.=C2=A0 The receiver<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 looks at the first byte of th=
e packet.=C2=A0 If the value of this byte is<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 in between 0 and 3 (inclusive=
), then the packet is STUN.=C2=A0 If the<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 value is between 16 and 19 (i=
nclusive), then the packet is ZRTP.=C2=A0 If<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 the value is between 20 and 6=
3 (inclusive), then the packet is DTLS.<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 If the value is between 64 an=
d 79 (inclusive), then the packet is<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 TURN Channel.=C2=A0 If the va=
lue is in between 128 and 191 (inclusive),<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 then the packet is RTP (or RT=
CP, if both RTCP and RTP are being<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 multiplexed over the same des=
tination port).=C2=A0 If the value does not<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 match any known range, then t=
he packet MUST be dropped and an alert<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0 MAY be logged.=C2=A0 This pro=
cess is summarized in Figure 3.<u></u><u></u></span></pre>
<pre><span style=3D"color:black"><u></u>=C2=A0<u></u></span></pre>
<pre><span style=3D"color:black">=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 +-=
---------------+<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0 [0..3] -+--&gt; forward to STUN<=
u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 |<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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 [16..19] -+--&gt; forward to ZRTP<u></u><u><=
/u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 |<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
 packet --&gt;=C2=A0 |=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 [20..63] -+--&gt; forw=
ard to DTLS<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 |<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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 [64..79] -+--&gt; forward to TURN Channel<u>=
</u><u></u></span></pre>
<pre><span style=3D"color:black">=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=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 |<u></u><u></u></span></pre>
<pre><span style=3D"color:black">=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 [128..191] -+--&gt; forward to RTP/RTCP<u></u><u></u></s=
pan></pre>
<pre><span style=3D"color:black">=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 +-=
---------------+<u></u><u></u></span></pre>
<pre><span style=3D"color:black"><u></u>=C2=A0<u></u></span></pre>
<pre><span style=3D"color:black">=C2=A0=C2=A0=C2=A0=C2=A0 Figure 3: The DTL=
S-SRTP receiver&#39;s packet demultiplexing algorithm.<u></u><u></u></span>=
</pre>
<pre><span style=3D"color:black"><u></u>=C2=A0<u></u></span></pre>
<pre><span style=3D"color:black"><u></u>=C2=A0<u></u></span></pre>
</div>
</div>
</div>
</div></div></div>
</div>

</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

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From: "Eggert, Lars" <lars@netapp.com>
To: Bernard Aboba <bernard.aboba@gmail.com>
CC: IETF QUIC WG <quic@ietf.org>, "marc@petit-huguenin.org" <marc@petit-huguenin.org>, "pthatcher@google.com" <pthatcher@google.com>, Colin Perkins <csp@csperkins.org>
Subject: Re: Potential conflict between draft-ietf-quic-transport and RFC 7983
Thread-Topic: Potential conflict between draft-ietf-quic-transport and RFC 7983
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FYI, Colin raised this issue in the past and we have ticket #426 on it =
open: https://github.com/quicwg/base-drafts/issues/426

Lars

> On 2017-5-17, at 23:36, Bernard Aboba <bernard.aboba@gmail.com> wrote:
>=20
> There appears to be a potential conflict between =
draft-ietf-quic-transport and RFC 7983. Looking at =
draft-ietf-quic-transport, both the short and the long headers appear to =
conflict with the de-multiplexing scheme defined in RFC 7983, which is =
based on the value of the first byte of multi-plexed protocols:
>=20
>    The process for demultiplexing a packet is as follows.  The =
receiver
>    looks at the first byte of the packet.  If the value of this byte =
is
>    in between 0 and 3 (inclusive), then the packet is STUN.  If the
>    value is between 16 and 19 (inclusive), then the packet is ZRTP.  =
If
>    the value is between 20 and 63 (inclusive), then the packet is =
DTLS.
>    If the value is between 64 and 79 (inclusive), then the packet is
>    TURN Channel.  If the value is in between 128 and 191 (inclusive),
>    then the packet is RTP (or RTCP, if both RTCP and RTP are being
>    multiplexed over the same destination port).  If the value does not
>    match any known range, then the packet MUST be dropped and an alert
>    MAY be logged.  This process is summarized in Figure 3.
>=20
>                     +----------------+
>                     |        [0..3] -+--> forward to STUN
>                     |                |
>                     |      [16..19] -+--> forward to ZRTP
>                     |                |
>         packet -->  |      [20..63] -+--> forward to DTLS
>                     |                |
>                     |      [64..79] -+--> forward to TURN Channel
>                     |                |
>                     |    [128..191] -+--> forward to RTP/RTCP
>                     +----------------+
>=20
>      Figure 3: The DTLS-SRTP receiver's packet demultiplexing =
algorithm.
>=20
>=20
>=20


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From nobody Thu May 18 04:17:37 2017
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Subject: Re: Potential conflict between draft-ietf-quic-transport and RFC 7983
Date: Thu, 18 May 2017 12:12:43 +0100
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I sketched out one way of doing the demuxing of QUIC and the WebRTC =
protocols on the same port in =
https://github.com/quicwg/base-drafts/issues/426 =
<https://github.com/quicwg/base-drafts/issues/426> =E2=80=93 if we=E2=80=99=
re willing to lose a couple of bits from the QUIC type field, then =
separating the protocols seems straight-forward.

Colin




> On 18 May 2017, at 02:02, Jana Iyengar <jri@google.com> wrote:
>=20
> This *may* be one of those things where keeping WebRTC in the back of =
our minds might help QUIC not get stuck badly in the future.
>=20
> So, as a thought exercise (not assuming that we need to solve this =
problem here) I'd like to walk through what we *could* do if we actually =
supported both WebRTC protocols and QUIC on the same port.
>=20
> I think the simplest thing to do would be to pass everything through a =
QUIC "dispatcher" that determines if this packet belongs to QUIC --- =
basically confirms whether this packet should be consumed by QUIC or not =
--- and if it isn't consumed by QUIC (including packets that QUIC =
decides MUST be dropped because they are QUIC packets but violate some =
property) then it can be consumed by non-QUIC applications on the same =
port.
>=20
> The question of course is how bad is the likelihood of false =
positives/negatives. With the encrypted packet types in QUIC, the QUIC =
dispatcher will safely reject non-QUIC packets.=20
>=20
> With the plaintext types, which are used during handshake, QUIC =
packets are covered by a non-crypto FNV-1a hash. The cost that QUIC =
bears of a mis-classification among plaintext packets as a QUIC packet =
is limited to cases where the contents of the packet are not part of the =
final key derivation. These packet types are limited to a few long-form =
packet types, all of which have a number of version-independent fields =
upfront, including version, which could be checked for sanity (since =
connection ID and packet number can be arbitrary.) This still doesn't =
seem great, since 0x0001 is a fine version number to have in QUIC and =
may be a completely reasonable string in RTP.  So alternatively, we =
could renumber the long-form packet types to start at 255 and count =
down, since 192 - 255 is left open by RFC 7983, and since we just need  =
6 cleartext packet types at the moment to be distinct from the RTP =
packet types (it's smaller than that, since Stateless Reject and Version =
Negotiation packets need to echo stuff from the client's packets.)
>=20
> TL;DR of this is that *if* we wanted to allow for a future coexistence =
of WebRTC protocols with QUIC on the same port -- and this is a big =
"if", since I wonder if they need to be on the same port --- we could =
renumber just the long-form packets to start at 255 and grow downwards. =
We only need <=3D 6 plaintext packet types to be out of the collision =
region with  RFC 7983.
>=20
>=20
> On Wed, May 17, 2017 at 3:31 PM, Bernard Aboba =
<bernard.aboba@gmail.com <mailto:bernard.aboba@gmail.com>> wrote:
> Mike asked:=20
>=20
> "Is it a goal for QUIC and DTLS-SRTP to coexist on the same port"?=20
>=20
> [BA] RFC 7983 enables DTLS, SRTP/SRTCP, ZRTP, STUN and TURN to =
co-exist on the same port.  Within the scheme, DTLS is used for =
transport of WebRTC's DTLS/SCTP/UDP data channel and DTLS-SRTP is used =
for key management of SRTP/SRTCP. =20
>=20
> While within WebRTC, DTLS currently provides both data channel =
transport as well as SRTP key management, it is not clear that initial =
QUIC implementations within WebRTC will attempt to provide alternatives =
for both of these functions. =20
>=20
> For example, experimental implementations of a QUIC data channel may =
wish to focus solely on that objective without having to provide an =
alternative to DTLS-SRTP.  In such an implementation, QUIC and DTLS-SRTP =
would need to co-exist on the same port.=20
>=20
>=20
>=20
> On Wed, May 17, 2017 at 2:52 PM, Mike Bishop =
<Michael.Bishop@microsoft.com <mailto:Michael.Bishop@microsoft.com>> =
wrote:
> 7983 is a bugfix to 5764 section 5.1.2, per a quick perusal.  I =
didn=E2=80=99t read either of those RFCs as purporting to constrain the =
first byte of all future UDP-based protocols.  5764 says:
>=20
> =20
>=20
>    When DTLS-SRTP is used to protect an RTP session, the RTP receiver
>=20
>    needs to demultiplex packets that are arriving on the RTP port.
>=20
> ...
>=20
>    If other packet types are to be multiplexed as well, implementors
>=20
>    and/or designers SHOULD ensure that they can be demultiplexed from
>=20
>    these three [or five, given 7983] packet types.
>=20
> =20
>=20
> Is it a goal for QUIC and DTLS-SRTP to coexist on the same port?
>=20
> =20
>=20
> From: QUIC [mailto:quic-bounces@ietf.org =
<mailto:quic-bounces@ietf.org>] On Behalf Of Bernard Aboba
> Sent: Wednesday, May 17, 2017 2:37 PM
> To: quic@ietf.org <mailto:quic@ietf.org>
> Cc: marc@petit-huguenin.org <mailto:marc@petit-huguenin.org>; =
pthatcher@google.com <mailto:pthatcher@google.com>
> Subject: Potential conflict between draft-ietf-quic-transport and RFC =
7983
>=20
> =20
>=20
> There appears to be a potential conflict between =
draft-ietf-quic-transport and RFC 7983. Looking at =
draft-ietf-quic-transport, both the short and the long headers appear to =
conflict with the de-multiplexing scheme defined in RFC 7983, which is =
based on the value of the first byte of multi-plexed protocols:=20
>=20
> =20
>=20
>    The process for demultiplexing a packet is as follows.  The =
receiver
>    looks at the first byte of the packet.  If the value of this byte =
is
>    in between 0 and 3 (inclusive), then the packet is STUN.  If the
>    value is between 16 and 19 (inclusive), then the packet is ZRTP.  =
If
>    the value is between 20 and 63 (inclusive), then the packet is =
DTLS.
>    If the value is between 64 and 79 (inclusive), then the packet is
>    TURN Channel.  If the value is in between 128 and 191 (inclusive),
>    then the packet is RTP (or RTCP, if both RTCP and RTP are being
>    multiplexed over the same destination port).  If the value does not
>    match any known range, then the packet MUST be dropped and an alert
>    MAY be logged.  This process is summarized in Figure 3.
> =20
>                     +----------------+
>                     |        [0..3] -+--> forward to STUN
>                     |                |
>                     |      [16..19] -+--> forward to ZRTP
>                     |                |
>         packet -->  |      [20..63] -+--> forward to DTLS
>                     |                |
>                     |      [64..79] -+--> forward to TURN Channel
>                     |                |
>                     |    [128..191] -+--> forward to RTP/RTCP
>                     +----------------+
> =20
>      Figure 3: The DTLS-SRTP receiver's packet demultiplexing =
algorithm.
> =20
> =20
>=20
>=20



--=20
Colin Perkins
https://csperkins.org/





--Apple-Mail=_B1F34B37-3FC3-4DAE-A5F9-6DF8711A5A8D
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D"">I sketched out one way of doing the demuxing of QUIC and the =
WebRTC protocols on the same port in&nbsp;<a =
href=3D"https://github.com/quicwg/base-drafts/issues/426" =
class=3D"">https://github.com/quicwg/base-drafts/issues/426</a>&nbsp;=E2=80=
=93 if we=E2=80=99re willing to lose a couple of bits from the QUIC type =
field, then separating the protocols seems straight-forward.<div =
class=3D""><br class=3D""></div><div class=3D"">Colin</div><div =
class=3D""><br class=3D""></div><div class=3D""><br class=3D""><div =
class=3D""><br class=3D""></div><div class=3D""><br =
class=3D""><div><blockquote type=3D"cite" class=3D""><div class=3D"">On =
18 May 2017, at 02:02, Jana Iyengar &lt;<a href=3D"mailto:jri@google.com" =
class=3D"">jri@google.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D""><div class=3D"">This *may* be one of those things where =
keeping WebRTC in the back of our minds might help QUIC not get stuck =
badly in the future.</div><div class=3D""><br class=3D""></div><div =
class=3D"">So, as a thought exercise (not assuming that we need to solve =
this problem here) I'd like to walk through what we *could* do if we =
actually supported both WebRTC protocols and QUIC on the same =
port.</div><div class=3D""><br class=3D""></div><div class=3D"">I think =
the simplest thing to do would be to pass everything through a QUIC =
"dispatcher" that determines if this packet belongs to QUIC --- =
basically confirms whether this packet should be consumed by QUIC or not =
--- and if it isn't consumed by QUIC (including packets that QUIC =
decides MUST be dropped because they are QUIC packets but violate some =
property) then it can be consumed by non-QUIC applications on the same =
port.</div><div class=3D""><br class=3D""></div><div class=3D"">The =
question of course is how bad is the likelihood of false =
positives/negatives. With the encrypted packet types in QUIC, the QUIC =
dispatcher will safely reject non-QUIC packets.&nbsp;</div><div =
class=3D""><br class=3D""></div><div class=3D"">With the plaintext =
types, which are used during handshake, QUIC packets are covered by a =
non-crypto FNV-1a hash. The cost that QUIC bears of a mis-classification =
among plaintext packets as a QUIC packet is limited to cases where the =
contents of the packet are not part of the final key derivation. These =
packet types are limited to a few long-form packet types, all of which =
have a number of version-independent fields upfront, including version, =
which could be checked for sanity (since connection ID and packet number =
can be arbitrary.) This still doesn't seem great, since 0x0001 is a fine =
version number to have in QUIC and may be a completely reasonable string =
in RTP.&nbsp; So alternatively, we could renumber the long-form packet =
types to start at 255 and count down, since 192 - 255 is left open by =
RFC 7983, and since we just need &nbsp;6 cleartext packet types at the =
moment to be distinct from the RTP packet types (it's smaller than that, =
since Stateless Reject and Version Negotiation packets need to echo =
stuff from the client's packets.)</div><div class=3D""><br =
class=3D""></div><div class=3D"">TL;DR of this is that *if* we wanted to =
allow for a future coexistence of WebRTC protocols with QUIC on the same =
port -- and this is a big "if", since I wonder if they need to be on the =
same port --- we could renumber just the long-form packets to start at =
255 and grow downwards. We only need &lt;=3D 6 plaintext packet types to =
be out of the collision region with &nbsp;RFC 7983.</div><div =
class=3D""><br class=3D""></div></div><div class=3D"gmail_extra"><br =
class=3D""><div class=3D"gmail_quote">On Wed, May 17, 2017 at 3:31 PM, =
Bernard Aboba <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank" =
class=3D"">bernard.aboba@gmail.com</a>&gt;</span> wrote:<br =
class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr" =
class=3D"">Mike asked:&nbsp;<span class=3D""><div class=3D""><br =
class=3D""></div><div class=3D"">"Is it a goal for QUIC and DTLS-SRTP to =
coexist on the same port"?&nbsp;</div><div class=3D""><br =
class=3D""></div></span><div class=3D"">[BA] RFC 7983 enables DTLS, =
SRTP/SRTCP, ZRTP, STUN and TURN to co-exist on the same port.&nbsp; =
Within the scheme, DTLS is used for transport of WebRTC's DTLS/SCTP/UDP =
data channel and DTLS-SRTP is used for key management of SRTP/SRTCP. =
&nbsp;<br class=3D""></div><div class=3D""><br class=3D""></div><div =
class=3D"">While within WebRTC, DTLS currently provides both data =
channel transport as well as SRTP key management, it is not clear that =
initial QUIC implementations within WebRTC will attempt to provide =
alternatives for both of these functions. &nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D"">For example, experimental =
implementations of a QUIC data channel may wish to focus solely on that =
objective without having to provide an alternative to DTLS-SRTP.&nbsp; =
In such an implementation, QUIC and DTLS-SRTP would need to co-exist on =
the same port.&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D""><br class=3D""></div></div><div class=3D"HOEnZb"><div =
class=3D"h5"><div class=3D"gmail_extra"><br class=3D""><div =
class=3D"gmail_quote">On Wed, May 17, 2017 at 2:52 PM, Mike Bishop <span =
dir=3D"ltr" class=3D"">&lt;<a href=3D"mailto:Michael.Bishop@microsoft.com"=
 target=3D"_blank" class=3D"">Michael.Bishop@microsoft.com</a>&gt;</span> =
wrote:<br class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"#0563C1" vlink=3D"#954F72" class=3D"">
<div class=3D"m_7572897580995963207m_5980825313082688534WordSection1"><p =
class=3D"MsoNormal">7983 is a bugfix to 5764 section 5.1.2, per a quick =
perusal.&nbsp; I didn=E2=80=99t read either of those RFCs as purporting =
to constrain the first byte of all future UDP-based protocols.&nbsp; =
5764 says:<u class=3D""></u><u class=3D""></u></p><p =
class=3D"MsoNormal"><u class=3D""></u>&nbsp;<u class=3D""></u></p><p =
class=3D"MsoNormal"><span style=3D"font-size: 10pt; font-family: =
'Courier New';" class=3D"">&nbsp;&nbsp; When DTLS-SRTP is used to =
protect an RTP session, the RTP receiver<u class=3D""></u><u =
class=3D""></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:=
 10pt; font-family: 'Courier New';" class=3D"">&nbsp;&nbsp; needs to =
demultiplex packets that are arriving on the RTP port.<u class=3D""></u><u=
 class=3D""></u></span></p><p class=3D"MsoNormal"><span =
style=3D"font-size: 10pt; font-family: 'Courier New';" class=3D"">...<u =
class=3D""></u><u class=3D""></u></span></p><p class=3D"MsoNormal"><span =
style=3D"font-size: 10pt; font-family: 'Courier New';" =
class=3D"">&nbsp;&nbsp;
<span style=3D"background:yellow" class=3D"">If other packet types are =
to be multiplexed as well</span>, implementors<u class=3D""></u><u =
class=3D""></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:=
 10pt; font-family: 'Courier New';" class=3D"">&nbsp;&nbsp; and/or =
designers SHOULD ensure that they can be demultiplexed from<u =
class=3D""></u><u class=3D""></u></span></p><p class=3D"MsoNormal"><span =
style=3D"font-size: 10pt; font-family: 'Courier New';" =
class=3D"">&nbsp;&nbsp; these three
</span><i class=3D""><span style=3D"" class=3D"">[or five, given =
7983]</span></i><span style=3D"font-size: 10pt; font-family: 'Courier =
New';" class=3D""> packet types.<u class=3D""></u><u =
class=3D""></u></span></p><p class=3D"MsoNormal"><span style=3D"font-size:=
 10pt; font-family: 'Courier New';" class=3D""><u class=3D""></u>&nbsp;<u =
class=3D""></u></span></p><p class=3D"MsoNormal">Is it a goal for QUIC =
and DTLS-SRTP to coexist on the same port?<u class=3D""></u><u =
class=3D""></u></p><p class=3D"MsoNormal"><u class=3D""></u>&nbsp;<u =
class=3D""></u></p><p class=3D"MsoNormal"><b class=3D"">From:</b> QUIC =
[mailto:<a href=3D"mailto:quic-bounces@ietf.org" target=3D"_blank" =
class=3D"">quic-bounces@ietf.org</a>] <b class=3D"">On Behalf Of
</b>Bernard Aboba<br class=3D"">
<b class=3D"">Sent:</b> Wednesday, May 17, 2017 2:37 PM<br class=3D"">
<b class=3D"">To:</b> <a href=3D"mailto:quic@ietf.org" target=3D"_blank" =
class=3D"">quic@ietf.org</a><br class=3D"">
<b class=3D"">Cc:</b> <a href=3D"mailto:marc@petit-huguenin.org" =
target=3D"_blank" class=3D"">marc@petit-huguenin.org</a>; <a =
href=3D"mailto:pthatcher@google.com" target=3D"_blank" =
class=3D"">pthatcher@google.com</a><br class=3D"">
<b class=3D"">Subject:</b> Potential conflict between =
draft-ietf-quic-transport and RFC 7983<u class=3D""></u><u =
class=3D""></u></p><div class=3D""><div =
class=3D"m_7572897580995963207h5"><p class=3D"MsoNormal"><u =
class=3D""></u>&nbsp;<u class=3D""></u></p>
<div class=3D""><p class=3D"MsoNormal">There appears to be a potential =
conflict between draft-ietf-quic-transport and&nbsp;RFC 7983. Looking at =
draft-ietf-quic-transport, both the short and the long headers appear to =
conflict with the de-multiplexing scheme defined in RFC 7983, which
 is based on the value of the first byte of multi-plexed =
protocols:&nbsp;<u class=3D""></u><u class=3D""></u></p>
<div class=3D"">
<div class=3D""><p class=3D"MsoNormal"><u class=3D""></u>&nbsp;<u =
class=3D""></u></p>
</div>
<div class=3D"">
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; The process for =
demultiplexing a packet is as follows.&nbsp; The receiver<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; looks at the =
first byte of the packet.&nbsp; If the value of this byte is<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; in between 0 =
and 3 (inclusive), then the packet is STUN.&nbsp; If the<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; value is =
between 16 and 19 (inclusive), then the packet is ZRTP.&nbsp; If<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; the value is =
between 20 and 63 (inclusive), then the packet is DTLS.<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; If the value is =
between 64 and 79 (inclusive), then the packet is<u class=3D""></u><u =
class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; TURN =
Channel.&nbsp; If the value is in between 128 and 191 (inclusive),<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; then the packet =
is RTP (or RTCP, if both RTCP and RTP are being<u class=3D""></u><u =
class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; multiplexed =
over the same destination port).&nbsp; If the value does not<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; match any known =
range, then the packet MUST be dropped and an alert<u class=3D""></u><u =
class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp; MAY be =
logged.&nbsp; This process is summarized in Figure 3.<u class=3D""></u><u =
class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D""><u class=3D""></u>&nbsp;<u =
class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; +----------------+<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&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; [0..3] -+--&gt; forward to =
STUN<u class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&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; |<u class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&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; [16..19] -+--&gt; forward to ZRTP<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&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; |<u class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; packet =
--&gt;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [20..63] -+--&gt; forward =
to DTLS<u class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&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; |<u class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&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; [64..79] -+--&gt; forward to TURN =
Channel<u class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&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; |<u class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
[128..191] -+--&gt; forward to RTP/RTCP<u class=3D""></u><u =
class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" =
class=3D"">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; +----------------+<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D""><u class=3D""></u>&nbsp;<u =
class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D"">&nbsp;&nbsp;&nbsp;&nbsp; =
Figure 3: The DTLS-SRTP receiver's packet demultiplexing algorithm.<u =
class=3D""></u><u class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D""><u class=3D""></u>&nbsp;<u =
class=3D""></u></span></pre>
<pre class=3D""><span style=3D"" class=3D""><u class=3D""></u>&nbsp;<u =
class=3D""></u></span></pre>
</div>
</div>
</div>
</div></div></div>
</div>

</blockquote></div><br class=3D""></div>
</div></div></blockquote></div><br class=3D""></div>
</div></blockquote></div><br class=3D""><div class=3D"">
<br class=3D""><br class=3D"">--&nbsp;<br class=3D"">Colin Perkins<br =
class=3D""><a href=3D"https://csperkins.org/" =
class=3D"">https://csperkins.org/</a><br class=3D""><br class=3D""><br =
class=3D""><br class=3D"">

</div>
<br class=3D""></div></div></body></html>=

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Subject: Re: Potential conflict between draft-ietf-quic-transport and RFC 7983
From: Colin Perkins <csp@csperkins.org>
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Date: Thu, 18 May 2017 12:15:08 +0100
Cc: Mike Bishop <Michael.Bishop@microsoft.com>, "marc@petit-huguenin.org" <marc@petit-huguenin.org>, "quic@ietf.org" <quic@ietf.org>, "pthatcher@google.com" <pthatcher@google.com>
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> On 17 May 2017, at 23:31, Bernard Aboba <bernard.aboba@gmail.com> =
wrote:
>=20
> Mike asked:=20
>=20
> "Is it a goal for QUIC and DTLS-SRTP to coexist on the same port"?=20
>=20
> [BA] RFC 7983 enables DTLS, SRTP/SRTCP, ZRTP, STUN and TURN to =
co-exist on the same port.  Within the scheme, DTLS is used for =
transport of WebRTC's DTLS/SCTP/UDP data channel and DTLS-SRTP is used =
for key management of SRTP/SRTCP. =20
>=20
> While within WebRTC, DTLS currently provides both data channel =
transport as well as SRTP key management, it is not clear that initial =
QUIC implementations within WebRTC will attempt to provide alternatives =
for both of these functions. =20
>=20
> For example, experimental implementations of a QUIC data channel may =
wish to focus solely on that objective without having to provide an =
alternative to DTLS-SRTP.  In such an implementation, QUIC and DTLS-SRTP =
would need to co-exist on the same port.=20

Being able to co-exist with STUN and TURN on the same port is likely =
important if we ever want QUIC to work peer-to-peer through NATs, too.

Colin


--=20
Colin Perkins
https://csperkins.org/





From nobody Sun May 21 21:54:24 2017
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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 QUIC of the IETF.

        Title           : QUIC: A UDP-Based Multiplexed and Secure Transport
        Authors         : Jana Iyengar
                          Martin Thomson
	Filename        : draft-ietf-quic-transport-03.txt
	Pages           : 76
	Date            : 2017-05-21

Abstract:
   This document defines the core of the QUIC transport protocol.  This
   document describes connection establishment, packet format,
   multiplexing and reliability.  Accompanying documents describe the
   cryptographic handshake and loss detection.


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

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From nobody Sun May 21 22:21:51 2017
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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 QUIC of the IETF.

        Title           : Hypertext Transfer Protocol (HTTP) over QUIC
        Author          : Mike Bishop
	Filename        : draft-ietf-quic-http-03.txt
	Pages           : 28
	Date            : 2017-05-21

Abstract:
   The QUIC transport protocol has several features that are desirable
   in a transport for HTTP, such as stream multiplexing, per-stream flow
   control, and low-latency connection establishment.  This document
   describes a mapping of HTTP semantics over QUIC.  This document also
   identifies HTTP/2 features that are subsumed by QUIC, and describes
   how HTTP/2 extensions can be ported to QUIC.


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https://datatracker.ietf.org/doc/draft-ietf-quic-http/

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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 QUIC of the IETF.

        Title           : Using Transport Layer Security (TLS) to Secure QUIC
        Authors         : Martin Thomson
                          Sean Turner
	Filename        : draft-ietf-quic-tls-03.txt
	Pages           : 37
	Date            : 2017-05-21

Abstract:
   This document describes how Transport Layer Security (TLS) is used to
   secure QUIC.

Note to Readers

   Discussion of this draft takes place on the QUIC working group
   mailing list (quic@ietf.org), which is archived at
   https://mailarchive.ietf.org/arch/search/?email_list=quic .

   Working Group information can be found at https://github.com/quicwg ;
   source code and issues list for this draft can be found at
   https://github.com/quicwg/base-drafts/labels/tls .


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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 QUIC of the IETF.

        Title           : QUIC Loss Detection and Congestion Control
        Authors         : Jana Iyengar
                          Ian Swett
	Filename        : draft-ietf-quic-recovery-03.txt
	Pages           : 18
	Date            : 2017-05-21

Abstract:
   This document describes loss detection and congestion control
   mechanisms for QUIC.

Note to Readers

   Discussion of this draft takes place on the QUIC working group
   mailing list (quic@ietf.org), which is archived at
   https://mailarchive.ietf.org/arch/search/?email_list=quic .

   Working Group information can be found at https://github.com/quicwg ;
   source code and issues list for this draft can be found at
   https://github.com/quicwg/base-drafts/labels/recovery .


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Everyone,

The -02 drafts incorporate the proposed resolutions to another group of =
issues, as listed below.=20

Please have a look through them. If there are any resolutions that you =
feel need more discussion, please bring it up, either here on the =
mailing list or by commenting on the issue itself. In either case, =
please explicitly ask for the issue to be re-opened if you want it to be =
reconsidered.

If folks do this before the Paris interim, we can hopefully have any =
necessary discussions before or during the meeting, allowing us to =
publish the First Implementation Draft afterwards, incorporating any =
changes needed.

See =
<https://github.com/quicwg/base-drafts/blob/master/CONTRIBUTING.md#resolvi=
ng-issues> for a reminder about the process we're using here. Even when =
we have consensus, we can reopen an issue if new information emerges =
(and that can take a variety of forms).

This list is also available at =
<https://github.com/quicwg/base-drafts/issues?page=3D2&q=3Dis%3Aissue+is%3=
Aclosed+-label%3Aduplicate+-label%3Aeditorial+-label%3Ahas-consensus&utf8=3D=
=E2=9C=93>.

* #543: Reduce the number of different offset sizes for STREAM   =
(transport)
* #542: Don't include timestamps until after the handshake completes   =
(transport)
* #513: Create more certainty about 0-RTT transport parameters   =
(transport)
* #481: FNV-1a 64   (transport)
* #451: No rules limit the sending of BLOCKED   (transport)
* #443: Split WINDOW_UPDATE   (transport)
* #442: Invert the connection ID logic during the handshake   =
(transport)
* #439: Unclear stream close semantics   (transport)
* #434: PADDING Frame performance   (transport)
* #432: Specify what concurrent stream limit protects   (transport)
* #425: Do not require clients to reset initial streams   (transport)
* #419: Get rid of the concurrent stream limit by advertising a maximum =
stream ID   (transport)
* #405: Transport parameter that limits 0-RTT data   (transport tls)
* #388: Timestamps in acknowledgment of PING?   (transport)
* #370: Out-of-order Flow Control   (transport)
* #344: Justify retransmitting handshake packets so slowly   (recovery)
* #294: Ignore version negotiation if the client version is present   =
(transport)
* #284: Ignore Version Negotiation if it was already done   (transport)
* #267: Server enforcement of 1280 octet packet size   (transport)
* #265: Define source address validation in the transport   (transport)
* #264: Define how source address validation interactions with TLS work  =
 (transport tls)
* #263: Counting closed streams against the concurrent stream limit   =
(transport)
* #248: Exemption from congestion control   (transport recovery)
* #241: What do we do if special packets are lost   (transport)
* #237: Create a firm recommendation on when to send WINDOW_UPDATE   =
(transport)
* #232: New connection ID message   (transport)
* #227: Encrypt the initial cleartext packets with a deterministic key   =
(transport)
* #217: Packet reordering / loss distorts open stream limits   =
(transport)
* #200: Race condition between stream creation and MSPC   (transport)
* #199: Why are reason phrases potentially 65k long?   (transport)
* #194: Frames should have a length field   (transport)
* #181: Remove SETTINGS[_ACK]   (transport http)
* #167: Hash for unencrypted packets   (transport tls)
* #158: Padding between frames   (transport)
* #143: Repeating Version Negotiation   (transport)
* #134: Certificate compression   (transport tls)
* #123: ACK frame timestamp format   (transport)
* #89: Version negotiation gaps   (transport)
* #60: Stateless Reject mechanism needs description   (transport tls)
* #55: What can change in a different version   (transport)
* #45: Handshake protocol selection   (transport)



--
Mark Nottingham   https://www.mnot.net/


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Subject: Re: Consensus Call on issues closed by the -03 drafts
From: Mark Nottingham <mnot@mnot.net>
In-Reply-To: <DC978CD6-248E-4A1C-9464-2F432EC5AB62@mnot.net>
Date: Tue, 23 May 2017 09:56:26 +1000
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To: IETF QUIC WG <quic@ietf.org>
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> On 22 May 2017, at 5:56 pm, Mark Nottingham <mnot@mnot.net> wrote:
>=20
> Everyone,
>=20
> The -02 drafts incorporate the proposed resolutions to another group =
of issues, as listed below.=20

That's -03, of course.

>=20
> Please have a look through them. If there are any resolutions that you =
feel need more discussion, please bring it up, either here on the =
mailing list or by commenting on the issue itself. In either case, =
please explicitly ask for the issue to be re-opened if you want it to be =
reconsidered.
>=20
> If folks do this before the Paris interim, we can hopefully have any =
necessary discussions before or during the meeting, allowing us to =
publish the First Implementation Draft afterwards, incorporating any =
changes needed.
>=20
> See =
<https://github.com/quicwg/base-drafts/blob/master/CONTRIBUTING.md#resolvi=
ng-issues> for a reminder about the process we're using here. Even when =
we have consensus, we can reopen an issue if new information emerges =
(and that can take a variety of forms).
>=20
> This list is also available at =
<https://github.com/quicwg/base-drafts/issues?page=3D2&q=3Dis%3Aissue+is%3=
Aclosed+-label%3Aduplicate+-label%3Aeditorial+-label%3Ahas-consensus&utf8=3D=
=E2=9C=93>.
>=20
> * #543: Reduce the number of different offset sizes for STREAM   =
(transport)
> * #542: Don't include timestamps until after the handshake completes   =
(transport)
> * #513: Create more certainty about 0-RTT transport parameters   =
(transport)
> * #481: FNV-1a 64   (transport)
> * #451: No rules limit the sending of BLOCKED   (transport)
> * #443: Split WINDOW_UPDATE   (transport)
> * #442: Invert the connection ID logic during the handshake   =
(transport)
> * #439: Unclear stream close semantics   (transport)
> * #434: PADDING Frame performance   (transport)
> * #432: Specify what concurrent stream limit protects   (transport)
> * #425: Do not require clients to reset initial streams   (transport)
> * #419: Get rid of the concurrent stream limit by advertising a =
maximum stream ID   (transport)
> * #405: Transport parameter that limits 0-RTT data   (transport tls)
> * #388: Timestamps in acknowledgment of PING?   (transport)
> * #370: Out-of-order Flow Control   (transport)
> * #344: Justify retransmitting handshake packets so slowly   =
(recovery)
> * #294: Ignore version negotiation if the client version is present   =
(transport)
> * #284: Ignore Version Negotiation if it was already done   =
(transport)
> * #267: Server enforcement of 1280 octet packet size   (transport)
> * #265: Define source address validation in the transport   =
(transport)
> * #264: Define how source address validation interactions with TLS =
work   (transport tls)
> * #263: Counting closed streams against the concurrent stream limit   =
(transport)
> * #248: Exemption from congestion control   (transport recovery)
> * #241: What do we do if special packets are lost   (transport)
> * #237: Create a firm recommendation on when to send WINDOW_UPDATE   =
(transport)
> * #232: New connection ID message   (transport)
> * #227: Encrypt the initial cleartext packets with a deterministic key =
  (transport)
> * #217: Packet reordering / loss distorts open stream limits   =
(transport)
> * #200: Race condition between stream creation and MSPC   (transport)
> * #199: Why are reason phrases potentially 65k long?   (transport)
> * #194: Frames should have a length field   (transport)
> * #181: Remove SETTINGS[_ACK]   (transport http)
> * #167: Hash for unencrypted packets   (transport tls)
> * #158: Padding between frames   (transport)
> * #143: Repeating Version Negotiation   (transport)
> * #134: Certificate compression   (transport tls)
> * #123: ACK frame timestamp format   (transport)
> * #89: Version negotiation gaps   (transport)
> * #60: Stateless Reject mechanism needs description   (transport tls)
> * #55: What can change in a different version   (transport)
> * #45: Handshake protocol selection   (transport)
>=20
>=20
>=20
> --
> Mark Nottingham   https://www.mnot.net/
>=20

--
Mark Nottingham   https://www.mnot.net/


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From: Mark Nottingham <mnot@mnot.net>
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Subject: Paris Agenda
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Date: Tue, 23 May 2017 15:36:52 +1000
Cc: Lars Eggert <lars@netapp.com>
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Everyone,

We've updated the agenda for Paris:
  https://datatracker.ietf.org/meeting/interim-2017-quic-02/agenda/quic/

As you can see, the focus of the meeting is getting the First =
Implementation Draft ready.  In particular, we'll be focusing on issues =
that block that as a matter of priority, so if you want to discuss any =
of the proposed resolutions in -03 (see previous message), or feel that =
any other issue blocks the First Implementation Draft, please bring it =
up as soon as you can.

Regards,

--
Mark Nottingham   https://www.mnot.net/


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "Jana Iyengar (jri@google.com)" <jri@google.com>, Ian Swett <ianswett@google.com>
CC: "quic@ietf.org" <quic@ietf.org>
Subject: On bytes per packet in draft-ietf-quic-recovery-03 
Thread-Topic: On bytes per packet in draft-ietf-quic-recovery-03 
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--_000_DB4PR07MB3487F71991FCFAB81938136C2F00DB4PR07MB348eurprd_
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Hi

I am reading myself though the QUIC drafts, in particular draft-ietf-quic-r=
ecovery, trying to get an idea of the meaning of bytes in a packet.
What does a QUIC packet constitute ?
QUIC
QUIC+UDP
QUIC+UDP+IP

I would believe that it is only the QUIC packet alone that counts in the dr=
afts or ?, on the other hand the MTU is 1460 bytes so I am not sure ?

BTW, a minor nit in draft-ietf-quic-recovery-03
  acket_packet -> acked_packet



=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D
Ingemar Johansson  M.Sc.
Master Researcher

Ericsson AB
Wireless Access Networks
Labratoriegr=E4nd 11
971 28, Lule=E5, Sweden
Phone +46-1071 43042
SMS/MMS +46-73 078 3289
ingemar.s.johansson@ericsson.com<mailto:ingemar.s.johansson@ericsson.com>
www.ericsson.com

A mistake is to commit a misunderstanding
                     Bob Dylan
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I am reading myself though the QUIC drafts, in parti=
cular draft-ietf-quic-recovery, trying to get an idea of the meaning of byt=
es in a packet.
<o:p></o:p></p>
<p class=3D"MsoNormal">What does a QUIC packet constitute ?<o:p></o:p></p>
<p class=3D"MsoNormal">QUIC<o:p></o:p></p>
<p class=3D"MsoNormal">QUIC&#43;UDP<o:p></o:p></p>
<p class=3D"MsoNormal">QUIC&#43;UDP&#43;IP<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I would believe that it is only the QUIC packet alon=
e that counts in the drafts or ?, on the other hand the MTU is 1460 bytes s=
o I am not sure ?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">BTW, a minor nit in draft-ietf-quic-recovery-03 <o:p=
></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;acket_packet -&gt; acked_packet<o:p></o:=
p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,sans-serif"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,sans-serif"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Ingemar Johansson&nbsp;=
 M.Sc.
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Master Researcher<o:p><=
/o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif"><o:p>&nbsp;</o:p></span=
></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Ericsson AB<o:p></o:p><=
/span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Wireless Access Network=
s<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Labratoriegr=E4nd 11<o:=
p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">971 28, Lule=E5, Sweden=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">Phone &#43;46-1071 4304=
2<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif">SMS/MMS &#43;46-73 078 =
3289<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><a href=3D"mailto:inge=
mar.s.johansson@ericsson.com"><span style=3D"font-size:10.0pt;font-family:&=
quot;Arial&quot;,sans-serif;color:blue">ingemar.s.johansson@ericsson.com</s=
pan></a><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,sans-=
serif"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><a href=3D"www.ericsso=
n.com"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,sans-s=
erif;color:blue">www.ericsson.com</span></a><span style=3D"font-size:10.0pt=
;font-family:&quot;Arial&quot;,sans-serif"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif;background:#CCCCDD"><o:p=
>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Arial&quot;,sans-serif;color:#333333;background=
:white">A mistake is to commit a misunderstanding</span><span style=3D"font=
-size:10.0pt;font-family:&quot;Arial&quot;,sans-serif;color:#333333"><br>
<span style=3D"background:white">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; Bob Dylan<br>
</span>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<span style=3D"background:white"><o:p><=
/o:p></span></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
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From: "Brian Trammell (IETF)" <ietf@trammell.ch>
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Subject: Should QUIC have a path-verifiable proof of source address?
Date: Wed, 31 May 2017 13:53:21 +0200
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Greetings, all,

We have a few contentious points in the design of QUIC, and how that =
design interacts with devices deployed in the Internet. I'd like to =
start the discussion on two such points, to see if we can come to some =
kind of consensus as to what we want QUIC to do without regard to how it =
does it, keeping in mind our charter requirement that QUIC have =
"security and privacy properties that are at least as good as a stack =
composed of TLS 1.3 using TCP (or MPTCP when using multipath)."

I'll start with the easier one:

"Shoud QUIC allow devices on path to distinguish QUIC traffic with valid =
source addresses from traffic with spoofed source addresses?"

The utility of this property is obvious: it would allow in-network =
reflection/amplification, backscatter, and spoofed-DoS protection =
equivalent to that available with TCP. This is already provided by TCP =
state tracking devices for TCP endpoints more or less as a side effect =
of their operation.

One could argue that this is pointless, since botnets-of-things have =
made spoofing irrelevant, and careful design (of new UDP protocols), =
aggressive patching (e.g. of ntpd), and lagging DNSSEC deployment are =
taking care of the most severe amplification vectors. The other side of =
that argument is essentially that every little bit helps.

The risks and efficiency tradeoffs are a matter of the exact design =
used, of course. However, it seems to me that establishing whether this =
is a desirable feature, and at what cost, seems like a good way to =
decide whether we should design for it, as opposed to letting discussion =
of mechanisms drive discussion of requirements.

Cheers,

Brian

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Subject: Re: Should QUIC have a path-verifiable proof of source address?
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On 31 May 2017 at 20:53, Brian Trammell (IETF) <ietf@trammell.ch> wrote:

> Greetings, all,
>
> We have a few contentious points in the design of QUIC, and how that
> design interacts with devices deployed in the Internet. I'd like to start
> the discussion on two such points, to see if we can come to some kind of
> consensus as to what we want QUIC to do without regard to how it does it,
> keeping in mind our charter requirement that QUIC have "security and
> privacy properties that are at least as good as a stack composed of TLS 1.3
> using TCP (or MPTCP when using multipath)."
>
> I'll start with the easier one:
>
> "Shoud QUIC allow devices on path to distinguish QUIC traffic with valid
> source addresses from traffic with spoofed source addresses?"
>
> The utility of this property is obvious: it would allow in-network
> reflection/amplification, backscatter, and spoofed-DoS protection
> equivalent to that available with TCP. This is already provided by TCP
> state tracking devices for TCP endpoints more or less as a side effect of
> their operation.
>
> One could argue that this is pointless, since botnets-of-things have made
> spoofing irrelevant, and careful design (of new UDP protocols), aggressive
> patching (e.g. of ntpd), and lagging DNSSEC deployment are taking care of
> the most severe amplification vectors. The other side of that argument is
> essentially that every little bit helps.
>
> The risks and efficiency tradeoffs are a matter of the exact design used,
> of course. However, it seems to me that establishing whether this is a
> desirable feature, and at what cost, seems like a good way to decide
> whether we should design for it, as opposed to letting discussion of
> mechanisms drive discussion of requirements.
>
> Cheers,
>
> Brian
>

My main concern here would be that preserve the ability to migrate
connections.

Allowing on-path but non-endpoint devices to distinguish traffic as you
describe opens up the opportunity for them to do it incorrectly/badly
thereby making some implementations of connection migration unreliable (at
best).

There may still be ways to support connection migration and support this
kind of traffic analysis, though.

--001a114920c694b7ee0550d0b8ba
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On 3=
1 May 2017 at 20:53, Brian Trammell (IETF) <span dir=3D"ltr">&lt;<a href=3D=
"mailto:ietf@trammell.ch" target=3D"_blank">ietf@trammell.ch</a>&gt;</span>=
 wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex">Greetings, all,<br>
<br>
We have a few contentious points in the design of QUIC, and how that design=
 interacts with devices deployed in the Internet. I&#39;d like to start the=
 discussion on two such points, to see if we can come to some kind of conse=
nsus as to what we want QUIC to do without regard to how it does it, keepin=
g in mind our charter requirement that QUIC have &quot;security and privacy=
 properties that are at least as good as a stack composed of TLS 1.3 using =
TCP (or MPTCP when using multipath).&quot;<br>
<br>
I&#39;ll start with the easier one:<br>
<br>
&quot;Shoud QUIC allow devices on path to distinguish QUIC traffic with val=
id source addresses from traffic with spoofed source addresses?&quot;<br>
<br>
The utility of this property is obvious: it would allow in-network reflecti=
on/amplification, backscatter, and spoofed-DoS protection equivalent to tha=
t available with TCP. This is already provided by TCP state tracking device=
s for TCP endpoints more or less as a side effect of their operation.<br>
<br>
One could argue that this is pointless, since botnets-of-things have made s=
poofing irrelevant, and careful design (of new UDP protocols), aggressive p=
atching (e.g. of ntpd), and lagging DNSSEC deployment are taking care of th=
e most severe amplification vectors. The other side of that argument is ess=
entially that every little bit helps.<br>
<br>
The risks and efficiency tradeoffs are a matter of the exact design used, o=
f course. However, it seems to me that establishing whether this is a desir=
able feature, and at what cost, seems like a good way to decide whether we =
should design for it, as opposed to letting discussion of mechanisms drive =
discussion of requirements.<br>
<br>
Cheers,<br>
<br>
Brian<br>
</blockquote></div><br></div><div class=3D"gmail_extra">My main concern her=
e would be that preserve the ability to migrate connections.</div><div clas=
s=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Allowing on-path but=
 non-endpoint devices to distinguish traffic as you describe opens up the o=
pportunity for them to do it incorrectly/badly thereby making some implemen=
tations of connection migration unreliable (at best).</div><div class=3D"gm=
ail_extra"><br></div><div class=3D"gmail_extra">There may still be ways to =
support connection migration and support this kind of traffic analysis, tho=
ugh.</div></div>

--001a114920c694b7ee0550d0b8ba--

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Subject: Re: Should QUIC have a path-verifiable proof of source address?
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Hi, Erik,


> On 31 May 2017, at 14:04, Erik Kline <ek@google.com> wrote:
>=20
>> On 31 May 2017 at 20:53, Brian Trammell (IETF) <ietf@trammell.ch> =
wrote:
>> Greetings, all,

<snip>

>> "Shoud QUIC allow devices on path to distinguish QUIC traffic with =
valid source addresses from traffic with spoofed source addresses?"

<snip>

> My main concern here would be that preserve the ability to migrate =
connections.
>=20
> Allowing on-path but non-endpoint devices to distinguish traffic as =
you describe opens up the opportunity for them to do it =
incorrectly/badly thereby making some implementations of connection =
migration unreliable (at best).
>=20
> There may still be ways to support connection migration and support =
this kind of traffic analysis, though.

Okay... so I'll mark that down "this seems useful, but not at the cost =
of any flexibility in connection migration."

Actually, your answer suggests an alternate and potentially more useful =
framing for this question: "which design properties are you not willing =
to sacrifice to get the ability for devices on path to distinguish QUIC =
traffic with valid source addresses from traffic with spoofed source =
addresses?", which captures the risk/utility tradeoff nicely.

Thanks, cheers,

Brian

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From: Erik Kline <ek@google.com>
Date: Wed, 31 May 2017 22:43:13 +0900
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Subject: Re: Should QUIC have a path-verifiable proof of source address?
To: "Brian Trammell (IETF)" <ietf@trammell.ch>
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On 31 May 2017 at 22:22, Brian Trammell (IETF) <ietf@trammell.ch> wrote:

> Hi, Erik,
>
>
> > On 31 May 2017, at 14:04, Erik Kline <ek@google.com> wrote:
> >
> >> On 31 May 2017 at 20:53, Brian Trammell (IETF) <ietf@trammell.ch>
> wrote:
> >> Greetings, all,
>
> <snip>
>
> >> "Shoud QUIC allow devices on path to distinguish QUIC traffic with
> valid source addresses from traffic with spoofed source addresses?"
>
> <snip>
>
> > My main concern here would be that preserve the ability to migrate
> connections.
> >
> > Allowing on-path but non-endpoint devices to distinguish traffic as you
> describe opens up the opportunity for them to do it incorrectly/badly
> thereby making some implementations of connection migration unreliable (at
> best).
> >
> > There may still be ways to support connection migration and support this
> kind of traffic analysis, though.
>
> Okay... so I'll mark that down "this seems useful, but not at the cost of
> any flexibility in connection migration."
>
> Actually, your answer suggests an alternate and potentially more useful
> framing for this question: "which design properties are you not willing to
> sacrifice to get the ability for devices on path to distinguish QUIC
> traffic with valid source addresses from traffic with spoofed source
> addresses?", which captures the risk/utility tradeoff nicely.
>
> Thanks, cheers,
>
> Brian
>

A comment and request to make a distinction (though I know it's
complicated).

[ comment ]
Anything on-path boxes can do analytically can become a point of policy
control in a vendor's marketing slide deck.  There is only embrittlement
down this road, I fear.

[ (complicated) distinction ]
I would like to argue that there is a distinction between:

    [1] on-path elements that are within the administrative domains of
either end-point ("administratively related to an endpoint") and

    [2] on-path elements in administrative domains not associated (except
by peering/transit) with either endpoint.

It might be possible to permit analysis (and therefore policy enforcement)
by the former and not the latter, if we assume there can be some
cooperation with endpoints and on-path elements in their administrative
domain.  If possible, I think this might give the best of both worlds: the
benefits of analysis and policy enforcement but only afforded to the
endpoints and their administrative domains (no interference from unrelated
parties).

--001a114baca83d791a0550d21b5d
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 31 May 2017 at 22:22, Brian Trammell (IETF) <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:ietf@trammell.ch" target=3D"_blank">ietf@trammell.ch</a>&gt=
;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hi, Er=
ik,<br>
<span class=3D"gmail-"><br>
<br>
&gt; On 31 May 2017, at 14:04, Erik Kline &lt;<a href=3D"mailto:ek@google.c=
om">ek@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; On 31 May 2017 at 20:53, Brian Trammell (IETF) &lt;<a href=3D"mail=
to:ietf@trammell.ch">ietf@trammell.ch</a>&gt; wrote:<br>
&gt;&gt; Greetings, all,<br>
<br>
</span>&lt;snip&gt;<br>
<span class=3D"gmail-"><br>
&gt;&gt; &quot;Shoud QUIC allow devices on path to distinguish QUIC traffic=
 with valid source addresses from traffic with spoofed source addresses?&qu=
ot;<br>
<br>
</span>&lt;snip&gt;<br>
<span class=3D"gmail-"><br>
&gt; My main concern here would be that preserve the ability to migrate con=
nections.<br>
&gt;<br>
&gt; Allowing on-path but non-endpoint devices to distinguish traffic as yo=
u describe opens up the opportunity for them to do it incorrectly/badly the=
reby making some implementations of connection migration unreliable (at bes=
t).<br>
&gt;<br>
&gt; There may still be ways to support connection migration and support th=
is kind of traffic analysis, though.<br>
<br>
</span>Okay... so I&#39;ll mark that down &quot;this seems useful, but not =
at the cost of any flexibility in connection migration.&quot;<br>
<br>
Actually, your answer suggests an alternate and potentially more useful fra=
ming for this question: &quot;which design properties are you not willing t=
o sacrifice to get the ability for devices on path to distinguish QUIC traf=
fic with valid source addresses from traffic with spoofed source addresses?=
&quot;, which captures the risk/utility tradeoff nicely.<br>
<br>
Thanks, cheers,<br>
<br>
Brian<br>
</blockquote></div><br></div><div class=3D"gmail_extra">A comment and reque=
st to make a distinction (though I know it&#39;s complicated).</div><div cl=
ass=3D"gmail_extra"><br></div><div class=3D"gmail_extra">[ comment ]</div><=
div class=3D"gmail_extra">Anything on-path boxes can do analytically can be=
come a point of policy control in a vendor&#39;s marketing slide deck.=C2=
=A0 There is only embrittlement down this road, I fear.</div><div class=3D"=
gmail_extra"><br></div><div class=3D"gmail_extra">[ (complicated) distincti=
on ]</div><div class=3D"gmail_extra">I would like to argue that there is a =
distinction between:</div><div class=3D"gmail_extra"><br></div><div class=
=3D"gmail_extra">=C2=A0 =C2=A0 [1] on-path elements that are within the adm=
inistrative domains of either end-point (&quot;administratively related to =
an endpoint&quot;) and</div><div class=3D"gmail_extra"><br></div><div class=
=3D"gmail_extra">=C2=A0 =C2=A0 [2] on-path elements in administrative domai=
ns not associated (except by peering/transit) with either endpoint.</div><d=
iv class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">It might be p=
ossible to permit analysis (and therefore policy enforcement) by the former=
 and not the latter, if we assume there can be some cooperation with endpoi=
nts and on-path elements in their administrative domain.=C2=A0 If possible,=
 I think this might give the best of both worlds: the benefits of analysis =
and policy enforcement but only afforded to the endpoints and their adminis=
trative domains (no interference from unrelated parties).</div></div>

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From: Dmitri Tikhonov <dtikhonov@litespeedtech.com>
Subject: Why mandate stream creation order?
Date: Wed, 31 May 2017 11:23:48 -0400
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Hello,

Section 10.1 of draft-ietf-quic-transport-03 states that =E2=80=9CStreams M=
UST be created in sequential order.=E2=80=9D  My question is: why mandate s=
tream creation order, since the other side may receive out-of-order packets=
?  Let the implementation do what it wants with Stream IDs =E2=80=93 as lon=
g as it does not go over the limit, all should be well.

This phrase is a holdover from the previous version; this requirement seems=
 to be unnecessary since issue 435 has been resolved.

- Dmitri.

P.S.  I have read CONTRIBUTING.md and decided that mailing list is the way =
to ask this question, rather than opening a GitHub issue.  If this is incor=
rect, please let me know.


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ss=3DWordSection1><p class=3DMsoNormal><span style=3D'font-family:"Times Ne=
w Roman",serif'>Hello,<o:p></o:p></span></p><p class=3DMsoNormal><span styl=
e=3D'font-family:"Times New Roman",serif'><o:p>&nbsp;</o:p></span></p><p cl=
ass=3DMsoNormal><span style=3D'font-family:"Times New Roman",serif'>Section=
 10.1 of draft-ietf-quic-transport-03 states that =E2=80=9C</span><span sty=
le=3D'font-family:"Courier New"'>Streams MUST be created in sequential orde=
r</span><span style=3D'font-family:"Times New Roman",serif'>.=E2=80=9D=C2=
=A0 My question is: why mandate stream creation order, since the other side=
 may receive out-of-order packets?=C2=A0 Let the implementation do what it =
wants with Stream IDs =E2=80=93 as long as it does not go over the limit, a=
ll should be well.<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D=
'font-family:"Times New Roman",serif'><o:p>&nbsp;</o:p></span></p><p class=
=3DMsoNormal><span style=3D'font-family:"Times New Roman",serif'>This phras=
e is a holdover from the previous version; this requirement seems to be unn=
ecessary since <a href=3D"https://github.com/quicwg/base-drafts/issues/435"=
>issue 435</a> has been resolved.<o:p></o:p></span></p><p class=3DMsoNormal=
><span style=3D'font-family:"Times New Roman",serif'><o:p>&nbsp;</o:p></spa=
n></p><ul style=3D'margin-top:0in' type=3Ddisc><li class=3DMsoListParagraph=
 style=3D'margin-left:-12.75pt;mso-list:l0 level1 lfo1'><span style=3D'font=
-family:"Times New Roman",serif'>Dmitri.<o:p></o:p></span></li></ul><p clas=
s=3DMsoNormal><span style=3D'font-family:"Times New Roman",serif'><o:p>&nbs=
p;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-family:"Times N=
ew Roman",serif'>P.S.=C2=A0 I have read <a href=3D"https://github.com/quicw=
g/base-drafts/blob/master/CONTRIBUTING.md">CONTRIBUTING.md</a> and decided =
that mailing list is the way to ask this question, rather than opening a Gi=
tHub issue.=C2=A0 If this is incorrect, please let me know.<o:p></o:p></spa=
n></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>=

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> On 31 May 2017, at 15:43, Erik Kline <ek@google.com> wrote:
>=20
> A comment and request to make a distinction (though I know it's =
complicated).
>=20
> [ comment ]
> Anything on-path boxes can do analytically can become a point of =
policy control in a vendor's marketing slide deck.  There is only =
embrittlement down this road, I fear.

Of course.

The higher-level argument I'm making behind this whole line of =
questioning is that there is a principle of conservation of (perceived) =
control in the deployed Internet, and that this will cause embrittlement =
to happen anyway. By being explicit and conservative about our aims for =
what can be done with the QUIC wire image by devices on path, we as =
protocol designers gain some measure of control over where this =
embrittlement occurs.

> [ (complicated) distinction ]
> I would like to argue that there is a distinction between:
>=20
>     [1] on-path elements that are within the administrative domains of =
either end-point ("administratively related to an endpoint") and
>=20
>     [2] on-path elements in administrative domains not associated =
(except by peering/transit) with either endpoint.
>=20
> It might be possible to permit analysis (and therefore policy =
enforcement) by the former and not the latter, if we assume there can be =
some cooperation with endpoints and on-path elements in their =
administrative domain.  If possible, I think this might give the best of =
both worlds: the benefits of analysis and policy enforcement but only =
afforded to the endpoints and their administrative domains (no =
interference from unrelated parties).

In general, I agree with you that there's a useful distinction here. And =
I agree completely that this function, spoofed traffic recognition and =
rejection, only really makes sense within the receiver's administrative =
domain.

(...and now, ignoring my own entreaty to focus on requirements as =
opposed to mechanisms in this aside...)

Making this distinction leveragable at runtime, however, is tricky: you =
need some way to establish trust with those devices; and either =
something like out-of-band like PCP, relying on on the "local to the =
path" endpoint to use it; or a much more elaborate inline signaling =
mechanism than IMO it makes sense to wedge into QUIC.

With respect to this *specific* question, though, I don't see how this =
would work with respect to spoofed traffic detection/rejection, since =
any design pattern leveraging this distinction necessarily involves the =
endpoints, and the whole point of spoof rejection as it's currently =
deployed is to reduce load on the endpoints.

Cheers,

Brian





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From nobody Wed May 31 10:14:15 2017
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From: Dave Dolson <ddolson@sandvine.com>
To: "Brian Trammell (IETF)" <ietf@trammell.ch>, Erik Kline <ek@google.com>
CC: QUIC WG <quic@ietf.org>
Subject: RE: Should QUIC have a path-verifiable proof of source address?
Thread-Topic: Should QUIC have a path-verifiable proof of source address?
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Date: Wed, 31 May 2017 17:14:01 +0000
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References: <179F2CCB-89DB-4E6E-9175-F850F89B4E5F@trammell.ch> <CAAedzxr2VyXpvvz8iyiyUWUZA4TZeWfCY58-tDgNfhRHqg5Oqg@mail.gmail.com> <C46523AE-6145-4CF1-9ED3-791F87DE9E3E@trammell.ch> <CAAedzxp5bvFDruo4hesa7nXDPiLSet-V8i+yD5JwjEQUg-PUGQ@mail.gmail.com> <54DB143C-3E91-4AB0-9132-8194A9AB0010@trammell.ch>
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> ... the whole point of spoof rejection as it's currently deployed is to r=
educe load on the endpoints

I would add, to also reduce load on the network.=20
In many cases the network (or one link thereof) presents more of a bottlene=
ck than the capacity of the endpoint.
In such cases, mitigation at the receiver is too late.

-Dave


-----Original Message-----
From: QUIC [mailto:quic-bounces@ietf.org] On Behalf Of Brian Trammell (IETF=
)
Sent: Wednesday, May 31, 2017 11:31 AM
To: Erik Kline
Cc: QUIC WG
Subject: Re: Should QUIC have a path-verifiable proof of source address?


> On 31 May 2017, at 15:43, Erik Kline <ek@google.com> wrote:
>=20
> A comment and request to make a distinction (though I know it's complicat=
ed).
>=20
> [ comment ]
> Anything on-path boxes can do analytically can become a point of policy c=
ontrol in a vendor's marketing slide deck.  There is only embrittlement dow=
n this road, I fear.

Of course.

The higher-level argument I'm making behind this whole line of questioning =
is that there is a principle of conservation of (perceived) control in the =
deployed Internet, and that this will cause embrittlement to happen anyway.=
 By being explicit and conservative about our aims for what can be done wit=
h the QUIC wire image by devices on path, we as protocol designers gain som=
e measure of control over where this embrittlement occurs.

> [ (complicated) distinction ]
> I would like to argue that there is a distinction between:
>=20
>     [1] on-path elements that are within the administrative domains of ei=
ther end-point ("administratively related to an endpoint") and
>=20
>     [2] on-path elements in administrative domains not associated (except=
 by peering/transit) with either endpoint.
>=20
> It might be possible to permit analysis (and therefore policy enforcement=
) by the former and not the latter, if we assume there can be some cooperat=
ion with endpoints and on-path elements in their administrative domain.  If=
 possible, I think this might give the best of both worlds: the benefits of=
 analysis and policy enforcement but only afforded to the endpoints and the=
ir administrative domains (no interference from unrelated parties).

In general, I agree with you that there's a useful distinction here. And I =
agree completely that this function, spoofed traffic recognition and reject=
ion, only really makes sense within the receiver's administrative domain.

(...and now, ignoring my own entreaty to focus on requirements as opposed t=
o mechanisms in this aside...)

Making this distinction leveragable at runtime, however, is tricky: you nee=
d some way to establish trust with those devices; and either something like=
 out-of-band like PCP, relying on on the "local to the path" endpoint to us=
e it; or a much more elaborate inline signaling mechanism than IMO it makes=
 sense to wedge into QUIC.

With respect to this *specific* question, though, I don't see how this woul=
d work with respect to spoofed traffic detection/rejection, since any desig=
n pattern leveraging this distinction necessarily involves the endpoints, a=
nd the whole point of spoof rejection as it's currently deployed is to redu=
ce load on the endpoints.

Cheers,

Brian





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Subject: enumerate packets not to ack?
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--001a114cad4c87300e0550d617e9
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Quic-Friends,

It seems clear that Version Negotiation and Server Stateless Retry packets
should not be acknowledged (afterall they cannot be retransmitted, and they
use packet numbers not generated by the sender).. but I can't find any text
that says this. Am I missing it, should we add it, other?

I guess I'm particularly concerned about interop running into "The sender
MUST close the connection if an unsent packet number is acknowledged" if
acks are sent here..

-P

--001a114cad4c87300e0550d617e9
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<div dir=3D"ltr"><div>Quic-Friends,</div><div><br></div><div>It seems clear=
 that Version Negotiation and Server Stateless Retry packets should not be =
acknowledged (afterall they cannot be retransmitted, and they use packet nu=
mbers not generated by the sender).. but I can&#39;t find any text that say=
s this. Am I missing it, should we add it, other?</div><div><br></div><div>=
I guess I&#39;m particularly concerned about interop running into &quot;The=
 sender MUST close the connection if an unsent packet number is acknowledge=
d&quot; if acks are sent here..</div><div><br></div><div>-P</div><div><br><=
/div></div>

--001a114cad4c87300e0550d617e9--


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From: Jana Iyengar <jri@google.com>
Date: Wed, 31 May 2017 11:57:55 -0700
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Subject: Re: enumerate packets not to ack?
To: Patrick McManus <pmcmanus@mozilla.com>
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Section 9 says acks must not generate acks, but this is a gap. We should
enumerate the packet types (and perhaps frame types too) that must not
generate acks. I've filed #563
<https://github.com/quicwg/base-drafts/issues/563>.


On Wed, May 31, 2017 at 11:28 AM, Patrick McManus <pmcmanus@mozilla.com>
wrote:

> Quic-Friends,
>
> It seems clear that Version Negotiation and Server Stateless Retry packets
> should not be acknowledged (afterall they cannot be retransmitted, and they
> use packet numbers not generated by the sender).. but I can't find any text
> that says this. Am I missing it, should we add it, other?
>
> I guess I'm particularly concerned about interop running into "The sender
> MUST close the connection if an unsent packet number is acknowledged" if
> acks are sent here..
>
> -P
>
>

--001a113900be7d9f700550d67f6d
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Section 9 says acks must not generate acks, but this is a =
gap. We should enumerate the packet types (and perhaps frame types too) tha=
t must not generate acks. I&#39;ve filed <a href=3D"https://github.com/quic=
wg/base-drafts/issues/563">#563</a>.<div><div><br></div></div></div><div cl=
ass=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, May 31, 2017 at =
11:28 AM, Patrick McManus <span dir=3D"ltr">&lt;<a href=3D"mailto:pmcmanus@=
mozilla.com" target=3D"_blank">pmcmanus@mozilla.com</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"><div dir=3D"ltr"><div>Quic-Friends,</div><=
div><br></div><div>It seems clear that Version Negotiation and Server State=
less Retry packets should not be acknowledged (afterall they cannot be retr=
ansmitted, and they use packet numbers not generated by the sender).. but I=
 can&#39;t find any text that says this. Am I missing it, should we add it,=
 other?</div><div><br></div><div>I guess I&#39;m particularly concerned abo=
ut interop running into &quot;The sender MUST close the connection if an un=
sent packet number is acknowledged&quot; if acks are sent here..</div><span=
 class=3D"HOEnZb"><font color=3D"#888888"><div><br></div><div>-P</div><div>=
<br></div></font></span></div>
</blockquote></div><br></div>

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From: Dmitri Tikhonov <dtikhonov@litespeedtech.com>
Subject: All stream frames are valid in IDLE state
Date: Wed, 31 May 2017 15:16:00 -0400
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Hello,

Section 10.2.1 of draft-ietf-quic-transport-03 says that =E2=80=9Cany frame=
 other than STREAM, MAX_STREAM_DATA, STREAM_BLOCKED, or RST_STREAM on a str=
eam in this [that is, idle] state MUST be treated as a connection error.=E2=
=80=9D

This list is exhaustive: no other frame types apply to a stream.  This para=
graph can be removed.

- Dmitri.


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<html xmlns:o=3D"urn:schemas-microsoft-com:office:office" xmlns:w=3D"urn:sc=
hemas-microsoft-com:office:word" xmlns:m=3D"http://schemas.microsoft.com/of=
fice/2004/12/omml" xmlns=3D"http://www.w3.org/TR/REC-html40"><head><meta ht=
tp-equiv=3DContent-Type content=3D"text/html; charset=3Dutf-8"><meta name=
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soNormal><span style=3D'font-family:"Times New Roman",serif'>Hello,<o:p></o=
:p></span></p><p class=3DMsoNormal><span style=3D'font-family:"Times New Ro=
man",serif'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D=
'font-family:"Times New Roman",serif'>Section 10.2.1 of draft-ietf-quic-tra=
nsport-03 says that =E2=80=9Cany frame other than STREAM, MAX_STREAM_DATA, =
STREAM_BLOCKED, or RST_STREAM on a stream in this [that is, idle] state MUS=
T be treated as a connection error.=E2=80=9D<o:p></o:p></span></p><p class=
=3DMsoNormal><span style=3D'font-family:"Times New Roman",serif'><o:p>&nbsp=
;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-family:"Times Ne=
w Roman",serif'>This list is exhaustive: no other frame types apply to a st=
ream.=C2=A0 This paragraph can be removed.<o:p></o:p></span></p><p class=3D=
MsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><span style=3D'font-fam=
ily:"Times New Roman",serif'>- Dmitri.<o:p></o:p></span></p><p class=3DMsoN=
ormal><o:p>&nbsp;</o:p></p></div></body></html>=

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From: Christian Huitema <huitema@huitema.net>
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To: quic@ietf.org
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Subject: Re: Should QUIC have a path-verifiable proof of source address?
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On 5/31/2017 4:53 AM, Brian Trammell (IETF) wrote:
> "Shoud QUIC allow devices on path to distinguish QUIC traffic with vali=
d source addresses from traffic with spoofed source addresses?"

Brian, I am not sure I understand your requirement. Do you mean
something like the reachability verification implicit in the three ways
handshake of TCP? Or do you mean some kind of cryptographic proof that
the device is allowed to use the source address?

-- Christian Huitema



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From: Martin Thomson <martin.thomson@gmail.com>
Date: Thu, 1 Jun 2017 11:21:52 +1000
Message-ID: <CABkgnnXvYGQJ=RFtkZ_wdN2WQGhOBu0_AvZsBQcQcgcDJWgJ+w@mail.gmail.com>
Subject: Re: enumerate packets not to ack?
To: Jana Iyengar <jri@google.com>
Cc: Patrick McManus <pmcmanus@mozilla.com>, IETF QUIC WG <quic@ietf.org>
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On 1 June 2017 at 04:57, Jana Iyengar <jri@google.com> wrote:
> Section 9 says acks must not generate acks, but this is a gap. We should
> enumerate the packet types (and perhaps frame types too) that must not
> generate acks. I've filed #563.

What's the taxonomy?  Can we split this based on frame type?

PADDING and ACK alone don't generate the need for an ACK.  However, a
packet with any other frame type does.  Except when they are carried
in Server Stateless Retry packets, which never generates
acknowledgments.

Version Negotiation packets don't contain frames so they don't cause
acknowledgments to be needed.   Here's a thought: maybe they should.


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Subject: Re: Should QUIC have a path-verifiable proof of source address?
To: Christian Huitema <huitema@huitema.net>
Cc: QUIC WG <quic@ietf.org>
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On 1 June 2017 at 06:09, Christian Huitema <huitema@huitema.net> wrote:
> Brian, I am not sure I understand your requirement. Do you mean
> something like the reachability verification implicit in the three ways
> handshake of TCP? Or do you mean some kind of cryptographic proof that
> the device is allowed to use the source address?

At risk of speaking for Brian, I think that this is intended to cover
the simple reachability verification implicit in TCP.  That is, proof
that the endpoint was able to see a packet that was sent to their
claimed address.

The verification provided by ICE also works.  There, it's not a
three-way handshake, but two independent request/response validations.

(I think that I see a companion to Brian's "what is an endpoint?" doc:
"what is a path?")

