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To: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>, Alexey Melnikov <aamelnikov@fastmail.fm>
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From: Benoit Claise <bclaise@cisco.com>
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Date: Mon, 3 Oct 2016 09:05:23 +0200
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Cc: quic@ietf.org, Alissa Cooper <alissa@cooperw.in>, The IESG <iesg@ietf.org>
Subject: Re: [QUIC] Alissa Cooper's Yes on charter-ietf-quic-00-08: (with COMMENT)
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On 9/29/2016 2:59 PM, Spencer Dawkins at IETF wrote:
> Dear IESG,
>
> On Wed, Sep 28, 2016 at 11:21 PM, Alexey Melnikov 
> <aamelnikov@fastmail.fm <mailto:aamelnikov@fastmail.fm>> wrote:
>
>
>     On 29 Sep 2016, at 03:54, Spencer Dawkins at IETF
>     <spencerdawkins.ietf@gmail.com
>     <mailto:spencerdawkins.ietf@gmail.com>> wrote:
>
>>     Hi, Alissa,
>>
>>     On Wed, Sep 28, 2016 at 8:46 PM, Alissa Cooper <alissa@cooperw.in
>>     <mailto:alissa@cooperw.in>> wrote:
>>
>>         Alissa Cooper has entered the following ballot position for
>>         charter-ietf-quic-00-08: Yes
>>
>>         When responding, please keep the subject line intact and
>>         reply to all
>>         email addresses included in the To and CC lines. (Feel free
>>         to cut this
>>         introductory paragraph, however.)
>>
>>
>>
>>         The document, along with other ballot positions, can be found
>>         here:
>>         https://datatracker.ietf.org/doc/charter-ietf-quic/
>>         <https://datatracker.ietf.org/doc/charter-ietf-quic/>
>>
>>
>>
>>         ----------------------------------------------------------------------
>>         COMMENT:
>>         ----------------------------------------------------------------------
>>
>>         "The QUIC
>>         protocol need not be defined to enable each of these
>>         abilities, or
>>         enable them in the same way as they are enabled by TCP when used
>>         with TLS 1.3, but the working group must consider the impact
>>         of the
>>         protocol on network management practices, in line with RFC 7258."
>>
>>         The way this is phrased makes it sound like RFC 7258 is about
>>         protocol
>>         impact on network management, which isn't the case. I would
>>         suggest
>>         dropping "in line with RFC 7258." Alternatively, you could
>>         say something
>>         like:
>>
>>         "The QUIC
>>         protocol need not be defined to enable each of these
>>         abilities, or
>>         enable them in the same way as they are enabled by TCP when used
>>         with TLS 1.3, but the working group must consider the impact
>>         of the
>>         protocol on network management practices, reflecting the tensions
>>         described in RFC 7258."
>>
>>
>>     I like your second alternative better.
>
> Having slept on this, I like your second alternative even better this 
> morning. I made the change, resulting in version 00-09.
>
> Thanks for the text suggestion, Alissa. *I* knew what I meant, but the 
> new text makes that clearer.
Fine with me, from the "network management practices" point of view.

Regards, Benoit
>
> Spencer
>
>     +1.
>>     I'll see if anyone else has thoughts about this, before we
>>     approve and announce.
>>
>>     Thank you for the suggestion.
>>
>>     Spencer
>
>


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    <div class="moz-cite-prefix">On 9/29/2016 2:59 PM, Spencer Dawkins
      at IETF wrote:<br>
    </div>
    <blockquote
cite="mid:CAKKJt-fYUv_P8GW+F730pXH8FNBB2TTi3N4ifcRvnSDX--m=+g@mail.gmail.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html; charset=utf-8">
      <div dir="ltr">Dear IESG,
        <div class="gmail_extra"><br>
          <div class="gmail_quote">On Wed, Sep 28, 2016 at 11:21 PM,
            Alexey Melnikov <span dir="ltr">&lt;<a
                moz-do-not-send="true"
                href="mailto:aamelnikov@fastmail.fm" target="_blank">aamelnikov@fastmail.fm</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div dir="auto">
                <div>
                  <div class="h5">
                    <div><br>
                    </div>
                    <div>On 29 Sep 2016, at 03:54, Spencer Dawkins at
                      IETF &lt;<a moz-do-not-send="true"
                        href="mailto:spencerdawkins.ietf@gmail.com"
                        target="_blank">spencerdawkins.ietf@gmail.com</a><wbr>&gt;
                      wrote:<br>
                      <br>
                    </div>
                    <blockquote type="cite">
                      <div>
                        <div dir="ltr">Hi, Alissa,
                          <div class="gmail_extra"><br>
                            <div class="gmail_quote">On Wed, Sep 28,
                              2016 at 8:46 PM, Alissa Cooper <span
                                dir="ltr">&lt;<a moz-do-not-send="true"
                                  href="mailto:alissa@cooperw.in"
                                  target="_blank">alissa@cooperw.in</a>&gt;</span>
                              wrote:<br>
                              <blockquote class="gmail_quote"
                                style="margin:0 0 0 .8ex;border-left:1px
                                #ccc solid;padding-left:1ex">Alissa
                                Cooper has entered the following ballot
                                position for<br>
                                charter-ietf-quic-00-08: Yes<br>
                                <br>
                                When responding, please keep the subject
                                line intact and reply to all<br>
                                email addresses included in the To and
                                CC lines. (Feel free to cut this<br>
                                introductory paragraph, however.)<br>
                                <br>
                                <br>
                                <br>
                                The document, along with other ballot
                                positions, can be found here:<br>
                                <a moz-do-not-send="true"
                                  href="https://datatracker.ietf.org/doc/charter-ietf-quic/"
                                  rel="noreferrer" target="_blank">https://datatracker.ietf.org/d<wbr>oc/charter-ietf-quic/</a><br>
                                <br>
                                <br>
                                <br>
                                ------------------------------<wbr>------------------------------<wbr>----------<br>
                                COMMENT:<br>
                                ------------------------------<wbr>------------------------------<wbr>----------<br>
                                <br>
                                "The QUIC<br>
                                protocol need not be defined to enable
                                each of these abilities, or<br>
                                enable them in the same way as they are
                                enabled by TCP when used<br>
                                with TLS 1.3, but the working group must
                                consider the impact of the<br>
                                protocol on network management
                                practices, in line with RFC 7258."<br>
                                <br>
                                The way this is phrased makes it sound
                                like RFC 7258 is about protocol<br>
                                impact on network management, which
                                isn't the case. I would suggest<br>
                                dropping "in line with RFC 7258."
                                Alternatively, you could say something<br>
                                like:<br>
                                <br>
                                "The QUIC<br>
                                protocol need not be defined to enable
                                each of these abilities, or<br>
                                enable them in the same way as they are
                                enabled by TCP when used<br>
                                with TLS 1.3, but the working group must
                                consider the impact of the<br>
                                protocol on network management
                                practices, reflecting the tensions<br>
                                described in RFC 7258."<br>
                              </blockquote>
                              <div><br>
                              </div>
                              <div>I like your second alternative
                                better.</div>
                            </div>
                          </div>
                        </div>
                      </div>
                    </blockquote>
                  </div>
                </div>
              </div>
            </blockquote>
            <div>Having slept on this, I like your second alternative
              even better this morning. I made the change, resulting in
              version 00-09.</div>
            <div><br>
            </div>
            <div>Thanks for the text suggestion, Alissa. *I* knew what I
              meant, but the new text makes that clearer.</div>
          </div>
        </div>
      </div>
    </blockquote>
    Fine with me, from the "network management practices" point of view.<br>
    <br>
    Regards, Benoit<br>
    <blockquote
cite="mid:CAKKJt-fYUv_P8GW+F730pXH8FNBB2TTi3N4ifcRvnSDX--m=+g@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div><br>
            </div>
            <div>Spencer</div>
            <div> </div>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div dir="auto">+1.<span class=""><br>
                  <blockquote type="cite">
                    <div>
                      <div dir="ltr">
                        <div class="gmail_extra">
                          <div class="gmail_quote">
                            <div> I'll see if anyone else has thoughts
                              about this, before we approve and
                              announce.</div>
                            <div><br>
                            </div>
                            <div>Thank you for the suggestion.</div>
                            <div><br>
                            </div>
                            <div>Spencer </div>
                          </div>
                        </div>
                      </div>
                    </div>
                  </blockquote>
                </span></div>
            </blockquote>
          </div>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

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From nobody Tue Oct  4 09:39:06 2016
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Subject: [QUIC] WG Action: Formed QUIC (quic)
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A new IETF WG has been formed in the Transport Area. For additional
information, please contact the Area Directors or the WG Chairs.

QUIC (quic)
-----------------------------------------------------------------------
Current status: Proposed WG

Chairs:
  Lars Eggert <lars@netapp.com>
  Mark Nottingham <mnot@mnot.net>

Assigned Area Director:
  Spencer Dawkins <spencerdawkins.ietf@gmail.com>

Transport Area Directors:
  Spencer Dawkins <spencerdawkins.ietf@gmail.com>
  Mirja Kühlewind <ietf@kuehlewind.net>
 
Mailing list:
  Address: quic@ietf.org
  To subscribe: https://www.ietf.org/mailman/listinfo/quic
  Archive: https://mailarchive.ietf.org/arch/browse/quic/

Charter: https://datatracker.ietf.org/doc/charter-ietf-quic/

The QUIC working group will provide a standards-track specification for 
a UDP-based, stream-multiplexing, encrypted transport protocol, based 
on pre-standardization implementation and deployment experience, and
generalizing the design described in 
draft-hamilton-quic-transport-protocol, 
draft-iyengar-quic-loss-recovery, 
draft-shade-quic-http2-mapping, and 
draft-thomson-quic-tls.

Key goals for QUIC are:

- Minimizing connection establishment and overall transport latency 
  for applications, starting with HTTP/2;
- Providing multiplexing without head-of-line blocking;
- Requiring only changes to path endpoints to enable deployment;
- Enabling multipath and forward error correction extensions; and
- Providing always-secure transport, using TLS 1.3 by default.

The work of the group will have five main focus areas, corresponding
to five core deliverables.

The first of these is the core transport work, which will describe 
the wire format, along with the mechanisms for connection 
establishment, stream multiplexing, data reliability, loss detection 
and recovery, congestion control, version negotiation, and options 
negotiation. Work on congestion control will describe use of a 
standardized congestion controller as a default scheme for
QUIC. Defining new congestion control schemes is explicitly out of 
scope for this group. QUIC is expected to support rapid, 
distributed development and testing of features.

The second of these focus areas is security. This work will describe 
how the protocol uses TLS 1.3 for key negotiation and will also 
describe how those keys are used to provide confidentiality and 
integrity protection of both application data and QUIC headers. 
This work will ensure that QUIC has 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).

The third focus area will describe mappings between specific 
application protocols and the transport facilities of QUIC. The first 
mapping will be a description of HTTP/2 semantics using QUIC, 
specifically with the goal of minimizing web latency using QUIC. This 
mapping will accommodate the extension mechanisms defined in the HTTP/2
specification. Upon completion of that mapping, additional protocols 
may be added by updating this charter to include them.

The fourth focus area will extend core protocol facilities to enable 
multipath capabilities for connection migration between paths and load 
sharing across multiple paths.

The fifth focus area will provide an Applicability and Manageability 
Statement, describing how, and under what circumstances, QUIC may be 
safely used, and describing deployment and manageability implications 
of the protocol. 

Current practices for network management of transport protocols 
include the ability to apply access control lists (ACLs), hashing of 
flows for equal-cost multipath routing (ECMP), directional signaling 
of flows, signaling of flow setup and teardown, and the ability to 
export information about flows for accounting purposes. The QUIC 
protocol need not be defined to enable each of these abilities, or 
enable them in the same way as they are enabled by TCP when used 
with TLS 1.3, but the working group must consider the impact of the
protocol on network management practices, reflecting the tensions
described in RFC 7258.

Extensions that will support partial reliability, and negotiation 
and use of Forward Error Correction schemes, are out of scope in 
this version of the working group charter.

Note that consensus is required both for changes to the current 
protocol mechanisms and retention of current mechanisms. In particular, 
because something is in the initial document set does not imply that 
there is consensus around the feature or around how it is specified.

The QUIC working group will work closely with the HTTPbis working 
group, especially on the QUIC mapping for HTTP/2.

In order to achieve the milestones set out below, the group expects 
to make extensive use of interim meetings, especially in its first year.

Milestones:
  Feb 2017 - Working group adoption of Core Protocol document
  Feb 2017 - Working group adoption of Loss detection and Congestion
Control document
  Feb 2017 - Working group adoption of TLS 1.3 mapping document
  Feb 2017 - Working group adoption of HTTP/2 mapping document
  Feb 2017 - Working group adoption of QUIC Applicability and
Manageability Statement
  Nov 2017 - Working group adoption of Multipath extension document
  Mar 2018 - Core Protocol document to IESG
  Mar 2018 - Loss detection and Congestion Control document to IESG
  Mar 2018 - TLS 1.3 Mapping document to IESG
  Nov 2018 - HTTP/2 mapping document to IESG
  Nov 2018 - QUIC Applicability and Manageability Statement to IESG
  May 2019 - Multipath extension document to IESG



From nobody Wed Oct  5 06:34:51 2016
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Subject: [QUIC] quic - New Meeting Session Request for IETF 97
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A new meeting session request has just been submitted by Cindy Morgan, on behalf of the quic working group.


---------------------------------------------------------
Working Group Name: QUIC
Area Name: Transport Area
Session Requester: Cindy Morgan

Number of Sessions: 1
Length of Session(s):  2.5 Hours
Number of Attendees: 250
Conflicts to Avoid: 
 First Priority: TSVAREA, SAAG, TLS, HTTPBIS, WEBPUSH, RTCWEB, ACME, DISPATCH, MAPRG, TSVWG, ICCRG, TCPM, TCPINC, MPTCP, TAPS, T2TRG, IANAPLAN, ARTAREA




Special Requests:
  If the schedule is tight, this could go in a 2-hour slot.
---------------------------------------------------------


From nobody Thu Oct  6 02:37:30 2016
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From: marcelo bagnulo braun <marcelo@it.uc3m.es>
To: quic@ietf.org, draft-hamilton-quic-transport-protocol@ietf.org
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Subject: [QUIC] about draft-hamilton-quic-transport-protocol-00
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Hi,


Thanks for the document. I have been reading 
draft-hamilton-quic-transport-protocol-00 and I find it not easy to 
read, probably because it is an early version and mostly because of my 
own limitations.

I attach a few comments and questions regarding the draft. Hope you find 
them useful.


Section 3.5 on Stream and Connection Flow control.

Is the offset announced for the connection the sum of the offsets 
announced for each of the stream of that connection?

If so, it seems redundant to announce both of them, no?

If it is not, I am uncertain how this would work. If the connection one 
is larger than the sum of the streams one, I am not sure how can the 
sender use the remaining buffer, since it cannot allocate it any 
existing stream. Is this meant to the be used for new streams? If the 
sum of the streams one is higher than the connection buffer, then i 
guess the sender needs to to a double check, first for available rcv 
window in the stream and then available rcv window in the connection. 
Not sure what is the gain of having both. Maybe it is worthwhile making 
explicit in this section why this is useful?


Section 3.7 on Resilience to NAT Rebinding.

I assume that the same mechanisms that allow the connection to survive a 
nat rebinding would be also useful to support some simple forms of mobility?

I mean, probably these mechanisms are enough to support the client 
changing IP address (for instance when doing a handover from cellular to 
wifi)?

I guess they are not enough to support more complex mobility scenarios, 
such as simultaneous movement of both endpoints, but in any case, it may 
be worthwhile mentioning this mobility support in this section?


Section 4.1 on Version Negotiation

I understand that during version negotiation, the following information 
is exchanged: version, version flag, connection ID, and in some cases 
the server's supported versions. I understand that all this information 
is exchanged before the crypto handshake, so it is exchange in clear 
text and without authentication. Since in section 3.6 it is claimed that 
packets are always authenticated, I guess we have a (mild) contradiction 
here (i understand you verify this later on once the crypto has been 
exchanged). In any case, maybe it is worthwhile to explicitly state in 
this section that this information is exchanged in plain text and non 
authenticated. Also, the last paragraph of this section states that 
there are other transport parameters exchanged in the connection 
establishment. I assume that those are exchanged after the crypto, so 
they are encrypted and authenticated? Is there any other transport 
parameter exchanged in clear text? Probably it would be useful to make 
this explicit?


In section 4.2.1.2 on Required Transport Parameters.

One thing I am missing here (probably because suffering a TCP bias) is 
the initial packet sequence number to be used in the connection.

Do you always use 0 as initial PSN? If so, this is because you assume 
that the crypto mechanisms established during the crypto handshake will 
prevent packet form previous connections to re-emerge later on and be 
confused for packets in the new connection? If this is the case, maybe 
you should state it explicitly in section 4.2.3?

However, from reading section 4.2.2 it seems that multiple connections 
between two endpoints can reuse the same crypto material, is that the case?

In this case, does each connection uses 0 as initial PSN? How can you 
guarantee in this case that old packets with same PSN will no re surface 
and appear as packet of a new connection, since they can be using the 
same PSN?


Section 4.2.2 on Proof of Source Address Ownership.

I understand that the STK provides less protection than a reachability 
check at the beginning of each connection, correct?

I mean, the STK proves that the client was once in that location and it 
can later leave the location and because it still has the STK, it can, 
by spoofing the source address, pretend to still be there, since only 
the STK is enough to pass this test.

This means that for instance, an attacker can download a large file by 
simply sending paced ACKs to any location it once was (given that the 
PSN initial number is always 0, it doesnt even needs to guess the PSNs), 
correct?

Maybe this is acceptable, but maybe it is worthwhile stating it in the 
security considerations section?


Section 4.2.3 on Crypto Handshake Protocol features

In the bullet about Connection establishment in 0-RTT, it requires 0-RTT 
connection establishment for subsequent connections between the same two 
endpoints, in the subsequent connections, the client and server roles 
must be preserved or they can be changed?

I mean, suppose that host A initiates a quic connection to host b. Does 
this condition requires that a future connection (with the same ports) 
initiated from host b to host A should be established in 0-RTT?

I think this should be made explicitly because mixed with the STK, I 
believe it changes the threat model significantly.

Nit: the paragraph on the bullet on Server Config Update is truncated.


Section 5.1 on Life of a stream.

It states that quic accommodates the possibility of out-of-order 
delivery. Do you mean out of order delivery within the same stream or 
across streams? clarify?

I understand that the creation of a stream is very lightweight and a new 
stream is established by simply data with a new stream id, correct?

I also understand that the initial stream sequence number is always 0, 
correct? (I mean, the offset of the current data to the beginning of the 
streams is included, and the initial packet has offset 0).

So, similar to the question i posed about the initial PSN, how does quic 
prevent old packets to reappear and be confused as being part of the new 
stream while they were part of an older stream? It cannot be done using 
the offset, since for all connection it starts in 0. I guess it could be 
done using the PSN, but then the receiver needs to check that all PSN of 
packets for a new stream are higher than the PSN of the packet that 
terminated the older stream, is this the proposed approach?

Probably it is worthwhile clarifying what are the means that quic deals 
with old packets re emerging after the connection/stream has ended, 
given that it does not relies on the sequence number to do so, afaict.

In particular, in section 5.1.1 it states that a stream is open by 
sending or receiving a data packet with that particular stream ID. I 
assume that the data packet MUST contain a 0 offset to create the new 
stream? is this correct?


Section 5.2 on Stream Identifiers.

I guess you should reserve a range of identifiers for future use, in 
case other "special" streams are needed in the future?


Section 5.4 on Sending and Receiving data.

Is the initial offset for the data packet sent on a stream set to zero? 
If this is the case, would be good to state it.


Section 6. on Packetization and reliability. What is the initial PSN in 
a given connection? State it explicitly.

It is stated that a sender MAY wait for a short period of time to bundle 
multiple frames. I guess it would be useful to provide guidance on what 
you consider a short period of time. Is that 500ms as in the delayed 
acks or are you thinking much less?

Later on, it states that only ACK, STOP_WAITING and PADDING frames MUST 
not be retransmitted, but all other frames must be retransmitted. This 
implies than that WINDOW_UPDATE and BLOCKED frames must be 
retransmitted, but i am not sure if this makes sense. I mean, maybe 
there were other WINDOW_UPDATE frames than were sent after this one so 
retransmitting this one probably doesnt make sense, since the 
information it contains may be stalled, no?

I still dont understand why having both connection and stream flow 
control is useful. I would assume that only stream flow control would be 
useful. Or having one or the other in a given connection, but i fail to 
understand why is useful to have both of them simultaneously in the same 
connection.


Section 6 states

    Regular QUIC Packets are "containers" of frames; a packet is never
    retransmitted whole, but frames in a lost packet may be rebundled and
    transmitted in a subsequent packet as necessary.

So, let me see if i understand this.

This means that if the sender sends packet with PSN 5, and if the packet 
is lost (i.e. there is a timeout for that packet), the packet with PSN 5 
NOT retransmitted. the frames in the packet with PSN 5 are then 
retransmitted in other packets with other PSN (other than 5). If this 
correct?

This means that there will be holes in the sequence of packets received 
by the receiver i.e. any packet that get's lost, is simply lost and will 
never be received by the receiver. Am i understanding this correctly?

    A receiver acknowledges receipt of a received packet by sending one
    or more ACK frames containing the PSN of the received packet.  To
    avoid perpetual acking between endpoints, a receiver MUST NOT
    generate an ack in response to every packet containing only ACK
    frames.

Does this means that ACKs transmission is unreliable, correct?

Since packets are never retransmitted with the same PSN, this basically 
means that if an ACK is lost, then the sender gives up on this packet 
and eventually will retransmit the data in other packets with a 
different PSN, correct?


Section 7.1.1 on Mid-stream RST_STREAM

After a RST, when the stream identifier can be reused? probably some 
guidance about this would be useful.


Section 7.1.4.

Is the BLOCKED frame reliably transmitted? I assume it is, but, does it 
make sense?

I mean, if a sender sends a blocked frame and it get's lost, but after 
that the receiver open the window, then retransmitting the BLOCKED frame 
is not useful (doesnt harm neither i guess)

OTOH, what if the BLOCKED frame get's lost? Do we end up with the silly 
window syndrome?

I guess we could do what TCP does i.e. periodically transmit the BLOCKED 
msg when there is no window (which is different from reliably 
transmitting the BLOCKED message, since it is periodically sent only 
when the window is 0)


Section 8 on connection termination

After a connection close, does the same stream identifier can be reused 
for a new connection? shouldnt also be a hold time to avoid delayed 
packets of the old connection? How much time would be safe?

Section 10.4

I understand ACK frames are unreliably transmitted.

Also, I understand that packets are never retransmitted with the same 
PSN. So, if a packet is lost, there will be a hole in the receiver for 
that PSN and the data will be sent in other packets with othew PSN. 
Hence, you cannot have cummulative acks.

So, this means that it is possible for a packet to be received by the 
receiver but that all the ACKs for that packet to be lost on the way 
back and then the sender will timeout and retransmit, right? (I guess 
that is ok, since data will be retransmitted in other packets)

Let me verify that I understand when a stop waiting frames should be 
sent by the sender. I guess that a sender should send a given PSN X in a 
stop waiting frame if all the packet with PSN lower than X have been 
received or have timed out, correct?

Regards, marcelo


From nobody Thu Oct  6 10:58:16 2016
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Subject: [QUIC] quic - Update to a Meeting Session Request for IETF 97
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An update to a meeting session request has just been submitted by Cindy Morgan, on behalf of the quic working group.


---------------------------------------------------------
Working Group Name: QUIC
Area Name: Transport Area
Session Requester: Cindy Morgan

Number of Sessions: 1
Length of Session(s):  2.5 Hours
Number of Attendees: 250
Conflicts to Avoid: 
 First Priority: tsvarea saag tls httpbis  dispatch maprg tsvwg iccrg tcpm tcpinc mptcp taps t2trg ianaplan artarea irtfopen
 Second Priority: webpush rtcweb acme



Special Requests:
  If the schedule is tight, this could go in a 2-hour slot.
---------------------------------------------------------


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Thanks much, Marcelo! There's definitely work to be done on the draft, so
comments are very appreciated!
I'll be revising the draft soon, so will respond to your detailed comments
soon.

On Thu, Oct 6, 2016 at 2:37 AM, marcelo bagnulo braun <marcelo@it.uc3m.es>
wrote:

> Hi,
>
>
> Thanks for the document. I have been reading draft-hamilton-quic-transport-protocol-00
> and I find it not easy to read, probably because it is an early version and
> mostly because of my own limitations.
>
> I attach a few comments and questions regarding the draft. Hope you find
> them useful.
>
>
> Section 3.5 on Stream and Connection Flow control.
>
> Is the offset announced for the connection the sum of the offsets
> announced for each of the stream of that connection?
>
> If so, it seems redundant to announce both of them, no?
>
> If it is not, I am uncertain how this would work. If the connection one is
> larger than the sum of the streams one, I am not sure how can the sender
> use the remaining buffer, since it cannot allocate it any existing stream.
> Is this meant to the be used for new streams? If the sum of the streams one
> is higher than the connection buffer, then i guess the sender needs to to a
> double check, first for available rcv window in the stream and then
> available rcv window in the connection. Not sure what is the gain of having
> both. Maybe it is worthwhile making explicit in this section why this is
> useful?
>
>
> Section 3.7 on Resilience to NAT Rebinding.
>
> I assume that the same mechanisms that allow the connection to survive a
> nat rebinding would be also useful to support some simple forms of mobility?
>
> I mean, probably these mechanisms are enough to support the client
> changing IP address (for instance when doing a handover from cellular to
> wifi)?
>
> I guess they are not enough to support more complex mobility scenarios,
> such as simultaneous movement of both endpoints, but in any case, it may be
> worthwhile mentioning this mobility support in this section?
>
>
> Section 4.1 on Version Negotiation
>
> I understand that during version negotiation, the following information is
> exchanged: version, version flag, connection ID, and in some cases the
> server's supported versions. I understand that all this information is
> exchanged before the crypto handshake, so it is exchange in clear text and
> without authentication. Since in section 3.6 it is claimed that packets are
> always authenticated, I guess we have a (mild) contradiction here (i
> understand you verify this later on once the crypto has been exchanged). In
> any case, maybe it is worthwhile to explicitly state in this section that
> this information is exchanged in plain text and non authenticated. Also,
> the last paragraph of this section states that there are other transport
> parameters exchanged in the connection establishment. I assume that those
> are exchanged after the crypto, so they are encrypted and authenticated? Is
> there any other transport parameter exchanged in clear text? Probably it
> would be useful to make this explicit?
>
>
> In section 4.2.1.2 on Required Transport Parameters.
>
> One thing I am missing here (probably because suffering a TCP bias) is the
> initial packet sequence number to be used in the connection.
>
> Do you always use 0 as initial PSN? If so, this is because you assume that
> the crypto mechanisms established during the crypto handshake will prevent
> packet form previous connections to re-emerge later on and be confused for
> packets in the new connection? If this is the case, maybe you should state
> it explicitly in section 4.2.3?
>
> However, from reading section 4.2.2 it seems that multiple connections
> between two endpoints can reuse the same crypto material, is that the case?
>
> In this case, does each connection uses 0 as initial PSN? How can you
> guarantee in this case that old packets with same PSN will no re surface
> and appear as packet of a new connection, since they can be using the same
> PSN?
>
>
> Section 4.2.2 on Proof of Source Address Ownership.
>
> I understand that the STK provides less protection than a reachability
> check at the beginning of each connection, correct?
>
> I mean, the STK proves that the client was once in that location and it
> can later leave the location and because it still has the STK, it can, by
> spoofing the source address, pretend to still be there, since only the STK
> is enough to pass this test.
>
> This means that for instance, an attacker can download a large file by
> simply sending paced ACKs to any location it once was (given that the PSN
> initial number is always 0, it doesnt even needs to guess the PSNs),
> correct?
>
> Maybe this is acceptable, but maybe it is worthwhile stating it in the
> security considerations section?
>
>
> Section 4.2.3 on Crypto Handshake Protocol features
>
> In the bullet about Connection establishment in 0-RTT, it requires 0-RTT
> connection establishment for subsequent connections between the same two
> endpoints, in the subsequent connections, the client and server roles must
> be preserved or they can be changed?
>
> I mean, suppose that host A initiates a quic connection to host b. Does
> this condition requires that a future connection (with the same ports)
> initiated from host b to host A should be established in 0-RTT?
>
> I think this should be made explicitly because mixed with the STK, I
> believe it changes the threat model significantly.
>
> Nit: the paragraph on the bullet on Server Config Update is truncated.
>
>
> Section 5.1 on Life of a stream.
>
> It states that quic accommodates the possibility of out-of-order delivery.
> Do you mean out of order delivery within the same stream or across streams?
> clarify?
>
> I understand that the creation of a stream is very lightweight and a new
> stream is established by simply data with a new stream id, correct?
>
> I also understand that the initial stream sequence number is always 0,
> correct? (I mean, the offset of the current data to the beginning of the
> streams is included, and the initial packet has offset 0).
>
> So, similar to the question i posed about the initial PSN, how does quic
> prevent old packets to reappear and be confused as being part of the new
> stream while they were part of an older stream? It cannot be done using the
> offset, since for all connection it starts in 0. I guess it could be done
> using the PSN, but then the receiver needs to check that all PSN of packets
> for a new stream are higher than the PSN of the packet that terminated the
> older stream, is this the proposed approach?
>
> Probably it is worthwhile clarifying what are the means that quic deals
> with old packets re emerging after the connection/stream has ended, given
> that it does not relies on the sequence number to do so, afaict.
>
> In particular, in section 5.1.1 it states that a stream is open by sending
> or receiving a data packet with that particular stream ID. I assume that
> the data packet MUST contain a 0 offset to create the new stream? is this
> correct?
>
>
> Section 5.2 on Stream Identifiers.
>
> I guess you should reserve a range of identifiers for future use, in case
> other "special" streams are needed in the future?
>
>
> Section 5.4 on Sending and Receiving data.
>
> Is the initial offset for the data packet sent on a stream set to zero? If
> this is the case, would be good to state it.
>
>
> Section 6. on Packetization and reliability. What is the initial PSN in a
> given connection? State it explicitly.
>
> It is stated that a sender MAY wait for a short period of time to bundle
> multiple frames. I guess it would be useful to provide guidance on what you
> consider a short period of time. Is that 500ms as in the delayed acks or
> are you thinking much less?
>
> Later on, it states that only ACK, STOP_WAITING and PADDING frames MUST
> not be retransmitted, but all other frames must be retransmitted. This
> implies than that WINDOW_UPDATE and BLOCKED frames must be retransmitted,
> but i am not sure if this makes sense. I mean, maybe there were other
> WINDOW_UPDATE frames than were sent after this one so retransmitting this
> one probably doesnt make sense, since the information it contains may be
> stalled, no?
>
> I still dont understand why having both connection and stream flow control
> is useful. I would assume that only stream flow control would be useful. Or
> having one or the other in a given connection, but i fail to understand why
> is useful to have both of them simultaneously in the same connection.
>
>
> Section 6 states
>
>    Regular QUIC Packets are "containers" of frames; a packet is never
>    retransmitted whole, but frames in a lost packet may be rebundled and
>    transmitted in a subsequent packet as necessary.
>
> So, let me see if i understand this.
>
> This means that if the sender sends packet with PSN 5, and if the packet
> is lost (i.e. there is a timeout for that packet), the packet with PSN 5
> NOT retransmitted. the frames in the packet with PSN 5 are then
> retransmitted in other packets with other PSN (other than 5). If this
> correct?
>
> This means that there will be holes in the sequence of packets received by
> the receiver i.e. any packet that get's lost, is simply lost and will never
> be received by the receiver. Am i understanding this correctly?
>
>    A receiver acknowledges receipt of a received packet by sending one
>    or more ACK frames containing the PSN of the received packet.  To
>    avoid perpetual acking between endpoints, a receiver MUST NOT
>    generate an ack in response to every packet containing only ACK
>    frames.
>
> Does this means that ACKs transmission is unreliable, correct?
>
> Since packets are never retransmitted with the same PSN, this basically
> means that if an ACK is lost, then the sender gives up on this packet and
> eventually will retransmit the data in other packets with a different PSN,
> correct?
>
>
> Section 7.1.1 on Mid-stream RST_STREAM
>
> After a RST, when the stream identifier can be reused? probably some
> guidance about this would be useful.
>
>
> Section 7.1.4.
>
> Is the BLOCKED frame reliably transmitted? I assume it is, but, does it
> make sense?
>
> I mean, if a sender sends a blocked frame and it get's lost, but after
> that the receiver open the window, then retransmitting the BLOCKED frame is
> not useful (doesnt harm neither i guess)
>
> OTOH, what if the BLOCKED frame get's lost? Do we end up with the silly
> window syndrome?
>
> I guess we could do what TCP does i.e. periodically transmit the BLOCKED
> msg when there is no window (which is different from reliably transmitting
> the BLOCKED message, since it is periodically sent only when the window is
> 0)
>
>
> Section 8 on connection termination
>
> After a connection close, does the same stream identifier can be reused
> for a new connection? shouldnt also be a hold time to avoid delayed packets
> of the old connection? How much time would be safe?
>
> Section 10.4
>
> I understand ACK frames are unreliably transmitted.
>
> Also, I understand that packets are never retransmitted with the same PSN.
> So, if a packet is lost, there will be a hole in the receiver for that PSN
> and the data will be sent in other packets with othew PSN. Hence, you
> cannot have cummulative acks.
>
> So, this means that it is possible for a packet to be received by the
> receiver but that all the ACKs for that packet to be lost on the way back
> and then the sender will timeout and retransmit, right? (I guess that is
> ok, since data will be retransmitted in other packets)
>
> Let me verify that I understand when a stop waiting frames should be sent
> by the sender. I guess that a sender should send a given PSN X in a stop
> waiting frame if all the packet with PSN lower than X have been received or
> have timed out, correct?
>
> Regards, marcelo
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

--001a114ba8a0be2f7a053e377cc6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks much, Marcelo! There&#39;s definitely work to be do=
ne on the draft, so comments are very appreciated!<div>I&#39;ll be revising=
 the draft soon, so will respond to your detailed comments soon.</div></div=
><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Oct 6, 2=
016 at 2:37 AM, marcelo bagnulo braun <span dir=3D"ltr">&lt;<a href=3D"mail=
to:marcelo@it.uc3m.es" target=3D"_blank">marcelo@it.uc3m.es</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">Hi,<br>
<br>
<br>
Thanks for the document. I have been reading draft-hamilton-quic-transport-=
<wbr>protocol-00 and I find it not easy to read, probably because it is an =
early version and mostly because of my own limitations.<br>
<br>
I attach a few comments and questions regarding the draft. Hope you find th=
em useful.<br>
<br>
<br>
Section 3.5 on Stream and Connection Flow control.<br>
<br>
Is the offset announced for the connection the sum of the offsets announced=
 for each of the stream of that connection?<br>
<br>
If so, it seems redundant to announce both of them, no?<br>
<br>
If it is not, I am uncertain how this would work. If the connection one is =
larger than the sum of the streams one, I am not sure how can the sender us=
e the remaining buffer, since it cannot allocate it any existing stream. Is=
 this meant to the be used for new streams? If the sum of the streams one i=
s higher than the connection buffer, then i guess the sender needs to to a =
double check, first for available rcv window in the stream and then availab=
le rcv window in the connection. Not sure what is the gain of having both. =
Maybe it is worthwhile making explicit in this section why this is useful?<=
br>
<br>
<br>
Section 3.7 on Resilience to NAT Rebinding.<br>
<br>
I assume that the same mechanisms that allow the connection to survive a na=
t rebinding would be also useful to support some simple forms of mobility?<=
br>
<br>
I mean, probably these mechanisms are enough to support the client changing=
 IP address (for instance when doing a handover from cellular to wifi)?<br>
<br>
I guess they are not enough to support more complex mobility scenarios, suc=
h as simultaneous movement of both endpoints, but in any case, it may be wo=
rthwhile mentioning this mobility support in this section?<br>
<br>
<br>
Section 4.1 on Version Negotiation<br>
<br>
I understand that during version negotiation, the following information is =
exchanged: version, version flag, connection ID, and in some cases the serv=
er&#39;s supported versions. I understand that all this information is exch=
anged before the crypto handshake, so it is exchange in clear text and with=
out authentication. Since in section 3.6 it is claimed that packets are alw=
ays authenticated, I guess we have a (mild) contradiction here (i understan=
d you verify this later on once the crypto has been exchanged). In any case=
, maybe it is worthwhile to explicitly state in this section that this info=
rmation is exchanged in plain text and non authenticated. Also, the last pa=
ragraph of this section states that there are other transport parameters ex=
changed in the connection establishment. I assume that those are exchanged =
after the crypto, so they are encrypted and authenticated? Is there any oth=
er transport parameter exchanged in clear text? Probably it would be useful=
 to make this explicit?<br>
<br>
<br>
In section 4.2.1.2 on Required Transport Parameters.<br>
<br>
One thing I am missing here (probably because suffering a TCP bias) is the =
initial packet sequence number to be used in the connection.<br>
<br>
Do you always use 0 as initial PSN? If so, this is because you assume that =
the crypto mechanisms established during the crypto handshake will prevent =
packet form previous connections to re-emerge later on and be confused for =
packets in the new connection? If this is the case, maybe you should state =
it explicitly in section 4.2.3?<br>
<br>
However, from reading section 4.2.2 it seems that multiple connections betw=
een two endpoints can reuse the same crypto material, is that the case?<br>
<br>
In this case, does each connection uses 0 as initial PSN? How can you guara=
ntee in this case that old packets with same PSN will no re surface and app=
ear as packet of a new connection, since they can be using the same PSN?<br=
>
<br>
<br>
Section 4.2.2 on Proof of Source Address Ownership.<br>
<br>
I understand that the STK provides less protection than a reachability chec=
k at the beginning of each connection, correct?<br>
<br>
I mean, the STK proves that the client was once in that location and it can=
 later leave the location and because it still has the STK, it can, by spoo=
fing the source address, pretend to still be there, since only the STK is e=
nough to pass this test.<br>
<br>
This means that for instance, an attacker can download a large file by simp=
ly sending paced ACKs to any location it once was (given that the PSN initi=
al number is always 0, it doesnt even needs to guess the PSNs), correct?<br=
>
<br>
Maybe this is acceptable, but maybe it is worthwhile stating it in the secu=
rity considerations section?<br>
<br>
<br>
Section 4.2.3 on Crypto Handshake Protocol features<br>
<br>
In the bullet about Connection establishment in 0-RTT, it requires 0-RTT co=
nnection establishment for subsequent connections between the same two endp=
oints, in the subsequent connections, the client and server roles must be p=
reserved or they can be changed?<br>
<br>
I mean, suppose that host A initiates a quic connection to host b. Does thi=
s condition requires that a future connection (with the same ports) initiat=
ed from host b to host A should be established in 0-RTT?<br>
<br>
I think this should be made explicitly because mixed with the STK, I believ=
e it changes the threat model significantly.<br>
<br>
Nit: the paragraph on the bullet on Server Config Update is truncated.<br>
<br>
<br>
Section 5.1 on Life of a stream.<br>
<br>
It states that quic accommodates the possibility of out-of-order delivery. =
Do you mean out of order delivery within the same stream or across streams?=
 clarify?<br>
<br>
I understand that the creation of a stream is very lightweight and a new st=
ream is established by simply data with a new stream id, correct?<br>
<br>
I also understand that the initial stream sequence number is always 0, corr=
ect? (I mean, the offset of the current data to the beginning of the stream=
s is included, and the initial packet has offset 0).<br>
<br>
So, similar to the question i posed about the initial PSN, how does quic pr=
event old packets to reappear and be confused as being part of the new stre=
am while they were part of an older stream? It cannot be done using the off=
set, since for all connection it starts in 0. I guess it could be done usin=
g the PSN, but then the receiver needs to check that all PSN of packets for=
 a new stream are higher than the PSN of the packet that terminated the old=
er stream, is this the proposed approach?<br>
<br>
Probably it is worthwhile clarifying what are the means that quic deals wit=
h old packets re emerging after the connection/stream has ended, given that=
 it does not relies on the sequence number to do so, afaict.<br>
<br>
In particular, in section 5.1.1 it states that a stream is open by sending =
or receiving a data packet with that particular stream ID. I assume that th=
e data packet MUST contain a 0 offset to create the new stream? is this cor=
rect?<br>
<br>
<br>
Section 5.2 on Stream Identifiers.<br>
<br>
I guess you should reserve a range of identifiers for future use, in case o=
ther &quot;special&quot; streams are needed in the future?<br>
<br>
<br>
Section 5.4 on Sending and Receiving data.<br>
<br>
Is the initial offset for the data packet sent on a stream set to zero? If =
this is the case, would be good to state it.<br>
<br>
<br>
Section 6. on Packetization and reliability. What is the initial PSN in a g=
iven connection? State it explicitly.<br>
<br>
It is stated that a sender MAY wait for a short period of time to bundle mu=
ltiple frames. I guess it would be useful to provide guidance on what you c=
onsider a short period of time. Is that 500ms as in the delayed acks or are=
 you thinking much less?<br>
<br>
Later on, it states that only ACK, STOP_WAITING and PADDING frames MUST not=
 be retransmitted, but all other frames must be retransmitted. This implies=
 than that WINDOW_UPDATE and BLOCKED frames must be retransmitted, but i am=
 not sure if this makes sense. I mean, maybe there were other WINDOW_UPDATE=
 frames than were sent after this one so retransmitting this one probably d=
oesnt make sense, since the information it contains may be stalled, no?<br>
<br>
I still dont understand why having both connection and stream flow control =
is useful. I would assume that only stream flow control would be useful. Or=
 having one or the other in a given connection, but i fail to understand wh=
y is useful to have both of them simultaneously in the same connection.<br>
<br>
<br>
Section 6 states<br>
<br>
=C2=A0 =C2=A0Regular QUIC Packets are &quot;containers&quot; of frames; a p=
acket is never<br>
=C2=A0 =C2=A0retransmitted whole, but frames in a lost packet may be rebund=
led and<br>
=C2=A0 =C2=A0transmitted in a subsequent packet as necessary.<br>
<br>
So, let me see if i understand this.<br>
<br>
This means that if the sender sends packet with PSN 5, and if the packet is=
 lost (i.e. there is a timeout for that packet), the packet with PSN 5 NOT =
retransmitted. the frames in the packet with PSN 5 are then retransmitted i=
n other packets with other PSN (other than 5). If this correct?<br>
<br>
This means that there will be holes in the sequence of packets received by =
the receiver i.e. any packet that get&#39;s lost, is simply lost and will n=
ever be received by the receiver. Am i understanding this correctly?<br>
<br>
=C2=A0 =C2=A0A receiver acknowledges receipt of a received packet by sendin=
g one<br>
=C2=A0 =C2=A0or more ACK frames containing the PSN of the received packet.=
=C2=A0 To<br>
=C2=A0 =C2=A0avoid perpetual acking between endpoints, a receiver MUST NOT<=
br>
=C2=A0 =C2=A0generate an ack in response to every packet containing only AC=
K<br>
=C2=A0 =C2=A0frames.<br>
<br>
Does this means that ACKs transmission is unreliable, correct?<br>
<br>
Since packets are never retransmitted with the same PSN, this basically mea=
ns that if an ACK is lost, then the sender gives up on this packet and even=
tually will retransmit the data in other packets with a different PSN, corr=
ect?<br>
<br>
<br>
Section 7.1.1 on Mid-stream RST_STREAM<br>
<br>
After a RST, when the stream identifier can be reused? probably some guidan=
ce about this would be useful.<br>
<br>
<br>
Section 7.1.4.<br>
<br>
Is the BLOCKED frame reliably transmitted? I assume it is, but, does it mak=
e sense?<br>
<br>
I mean, if a sender sends a blocked frame and it get&#39;s lost, but after =
that the receiver open the window, then retransmitting the BLOCKED frame is=
 not useful (doesnt harm neither i guess)<br>
<br>
OTOH, what if the BLOCKED frame get&#39;s lost? Do we end up with the silly=
 window syndrome?<br>
<br>
I guess we could do what TCP does i.e. periodically transmit the BLOCKED ms=
g when there is no window (which is different from reliably transmitting th=
e BLOCKED message, since it is periodically sent only when the window is 0)=
<br>
<br>
<br>
Section 8 on connection termination<br>
<br>
After a connection close, does the same stream identifier can be reused for=
 a new connection? shouldnt also be a hold time to avoid delayed packets of=
 the old connection? How much time would be safe?<br>
<br>
Section 10.4<br>
<br>
I understand ACK frames are unreliably transmitted.<br>
<br>
Also, I understand that packets are never retransmitted with the same PSN. =
So, if a packet is lost, there will be a hole in the receiver for that PSN =
and the data will be sent in other packets with othew PSN. Hence, you canno=
t have cummulative acks.<br>
<br>
So, this means that it is possible for a packet to be received by the recei=
ver but that all the ACKs for that packet to be lost on the way back and th=
en the sender will timeout and retransmit, right? (I guess that is ok, sinc=
e data will be retransmitted in other packets)<br>
<br>
Let me verify that I understand when a stop waiting frames should be sent b=
y the sender. I guess that a sender should send a given PSN X in a stop wai=
ting frame if all the packet with PSN lower than X have been received or ha=
ve timed out, correct?<br>
<br>
Regards, marcelo<br>
<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org" target=3D"_blank">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/quic</a><br>
</blockquote></div><br></div>

--001a114ba8a0be2f7a053e377cc6--


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From: marcelo bagnulo braun <marcelo@it.uc3m.es>
To: quic@ietf.org, draft-iyengar-quic-loss-recovery@ietf.org
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Subject: [QUIC] About loss detection and recovery in draft-iyengar-quic-loss-recovery-00
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Hi,

So, i have been trying to understand the packet loss detection and 
retransmission schemes for quic, described in 
draft-iyengar-quic-loss-recovery-00.

I had a few difficulties to understand how this works, based on the 
description in the draft. I include below what i think i understood, let 
me know if i got this right.

Section 2 and 3 mention the retransmission of lost packets, but if i 
understood it correctly, i think the draft would benefit from dropping 
the concept of retransmitted packet. I mean, I dont think there is such 
a thing as a retransmitted packet. What quic retransmits is the data 
within packets that were detected as lost, but the packet is not 
retransmitted. For starters, the packet containing the retransmitted 
data has a different PSN. But also, the content of the packet may be 
different as some frames are not to be retransmitted.

Moreover, if eventually some form of priority is implemented, i would 
guess that data with higher priority would be retransmitted earlier than 
data with lower priority. I think such flexibility would be useful.

Let me see if I understand the overall framework for loss detection and 
retransmission for quic. I think that quic detect packet loss, but 
retransmits data (that was sent in packets) so it probably possible to 
decouple the loss detection from the retransmission.

I understand that all data (control and application data) that is 
reliably transmitted is transmitted within a stream. Other data that is 
not transmitted within a given stream (like ACKs) is not reliably 
transmitted. Is this correct? If so, it may be worth stating it as a 
design principle, so people know that if they want to include control 
information such as other options to be reliably transmitted, they 
should do it within a stream.

So, if a sender has M ongoing streams, then it has M+1 sliding windows, 
i.e. one for each stream plus one for the packets at the connection 
level. Let's call the packet one, the packet window and the other the 
stream n window. The packet window is in packets, while the stream 
windows are in bytes.

The packet window has the following types of packets: packets acked, 
non-acked packets and lost packets. The left side of the window moves to 
packet number n when all packets with packet number lower than n are 
marked as acked or as lost. In other words, the left side of the window 
is the non-acked packet with lowest packet number. This corresponds to 
the "least unacked" packet mentioned in the draft.

Each stream window has the following types of bytes. bytes acked, bytes 
non-acked, bytes awaiting for transmission. Bytes non-acked are bytes 
that are contained in packets on flight. Bytes awaiting for transmission 
are bytes that either have not been sent yet or bytes that have been 
sent in packets that have been determined as lost.

Whenever a packet is acked, the bytes in each stream window 
corresponding to the data of each stream included in the acked packet 
are marked as acked.

Whenever a packet is determined as lost, the bytes in each stream window 
corresponding to the data of that stream included in the packet are 
marked as bytes awaiting for transmission.

Whenever a new packet is transmitted, the bytes in each stream window 
included in the packet are marked as non-acked.

Packet loss detection only operates using the packet window. So, each 
non-acked packet has an associated timer that when it times out, the 
packet is marked as lost. You can define any other mechanisms to detect 
packet loss only operating in this window, i believe.

Packet transmission otoh, needs to get data from the different stream 
windows, probably using the bytes awaiting for transmission with the 
lowest offset. This would include both the transmission of new data as 
well as retransmission of old data.

Do you think this properly describes the framework of operation for loss 
detection and retransmission of quic?

Regards, marcelo



From nobody Fri Oct  7 12:14:05 2016
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From: Ian Swett <ianswett@google.com>
Date: Fri, 7 Oct 2016 15:13:39 -0400
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To: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Cc: draft-iyengar-quic-loss-recovery@ietf.org, quic@ietf.org
Subject: Re: [QUIC] About loss detection and recovery in draft-iyengar-quic-loss-recovery-00
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Thanks for taking such a close look Marcelo, comments inline.

On Fri, Oct 7, 2016 at 2:05 AM, marcelo bagnulo braun <marcelo@it.uc3m.es>
 wrote:

> Hi,
>
> So, i have been trying to understand the packet loss detection and
> retransmission schemes for quic, described in draft-iyengar-quic-loss-recove
> ry-00.
>
> I had a few difficulties to understand how this works, based on the
> description in the draft. I include below what i think i understood, let me
> know if i got this right.
>
> Section 2 and 3 mention the retransmission of lost packets, but if i
> understood it correctly, i think the draft would benefit from dropping the
> concept of retransmitted packet. I mean, I dont think there is such a thing
> as a retransmitted packet. What quic retransmits is the data within packets
> that were detected as lost, but the packet is not retransmitted. For
> starters, the packet containing the retransmitted data has a different PSN.
> But also, the content of the packet may be different as some frames are not
> to be retransmitted.
>
>
Completely true that QUIC never retransmits packets, only data.  It would
be better to drop that wording.


> Moreover, if eventually some form of priority is implemented, i would
> guess that data with higher priority would be retransmitted earlier than
> data with lower priority. I think such flexibility would be useful.
>
>
Completely agreed, though I'm not sure that needs to be specified in this
doc or left up to implementations?


> Let me see if I understand the overall framework for loss detection and
> retransmission for quic. I think that quic detect packet loss, but
> retransmits data (that was sent in packets) so it probably possible to
> decouple the loss detection from the retransmission.
>
>
Yes, very much so.


> I understand that all data (control and application data) that is reliably
> transmitted is transmitted within a stream. Other data that is not
> transmitted within a given stream (like ACKs) is not reliably transmitted.
> Is this correct? If so, it may be worth stating it as a design principle,
> so people know that if they want to include control information such as
> other options to be reliably transmitted, they should do it within a stream.
>
>
It's correct that acks and stop waiting frames are not reliable frames and
hence are not retransmitted, but other frame types are retransmittable and
not be part of a stream(ie: PING frames).

So, if a sender has M ongoing streams, then it has M+1 sliding windows,
> i.e. one for each stream plus one for the packets at the connection level.
> Let's call the packet one, the packet window and the other the stream n
> window. The packet window is in packets, while the stream windows are in
> bytes.
>
>
Yes.


> The packet window has the following types of packets: packets acked,
> non-acked packets and lost packets. The left side of the window moves to
> packet number n when all packets with packet number lower than n are marked
> as acked or as lost. In other words, the left side of the window is the
> non-acked packet with lowest packet number. This corresponds to the "least
> unacked" packet mentioned in the draft.
>
> Each stream window has the following types of bytes. bytes acked, bytes
> non-acked, bytes awaiting for transmission. Bytes non-acked are bytes that
> are contained in packets on flight. Bytes awaiting for transmission are
> bytes that either have not been sent yet or bytes that have been sent in
> packets that have been determined as lost.
>
> Whenever a packet is acked, the bytes in each stream window corresponding
> to the data of each stream included in the acked packet are marked as acked.
>
> Whenever a packet is determined as lost, the bytes in each stream window
> corresponding to the data of that stream included in the packet are marked
> as bytes awaiting for transmission.
>
> Whenever a new packet is transmitted, the bytes in each stream window
> included in the packet are marked as non-acked.
>
> Packet loss detection only operates using the packet window. So, each
> non-acked packet has an associated timer that when it times out, the packet
> is marked as lost. You can define any other mechanisms to detect packet
> loss only operating in this window, i believe.
>

Yes, the default mechanism actually uses FACK style packet number gaps, but
time is an option and the important thing is loss detection operates on
this connection-level packet window.


>
> Packet transmission otoh, needs to get data from the different stream
> windows, probably using the bytes awaiting for transmission with the lowest
> offset. This would include both the transmission of new data as well as
> retransmission of old data.
>
> Do you think this properly describes the framework of operation for loss
> detection and retransmission of quic?
>
>
I think that captures the core ideas, yes.


> Regards, marcelo

--001a114dacbc6d25ba053e4b36d4
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_quote" style=3D"font-size:12.8px">Than=
ks for taking such a close look Marcelo, comments inline.</div><div class=
=3D"gmail_quote" style=3D"font-size:12.8px"><br></div><div class=3D"gmail_q=
uote" style=3D"font-size:12.8px"><span class=3D"gmail-im">On Fri, Oct 7, 20=
16 at 2:05 AM, marcelo bagnulo braun=C2=A0<span dir=3D"ltr">&lt;<a href=3D"=
mailto:marcelo@it.uc3m.es" target=3D"_blank">marcelo@it.uc3m.es</a>&gt;</sp=
an>=C2=A0wrote:<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,<br=
><br>So, i have been trying to understand the packet loss detection and ret=
ransmission schemes for quic, described in draft-iyengar-quic-loss-recove<w=
br>ry-00.<br><br>I had a few difficulties to understand how this works, bas=
ed on the description in the draft. I include below what i think i understo=
od, let me know if i got this right.<br><br>Section 2 and 3 mention the ret=
ransmission of lost packets, but if i understood it correctly, i think the =
draft would benefit from dropping the concept of retransmitted packet. I me=
an, I dont think there is such a thing as a retransmitted packet. What quic=
 retransmits is the data within packets that were detected as lost, but the=
 packet is not retransmitted. For starters, the packet containing the retra=
nsmitted data has a different PSN. But also, the content of the packet may =
be different as some frames are not to be retransmitted.<br><br></blockquot=
e><div><br></div></span><div>Completely true that QUIC never retransmits pa=
ckets, only data.=C2=A0 It would be better to drop that wording.</div><span=
 class=3D"gmail-im"><div>=C2=A0</div><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);paddin=
g-left:1ex">Moreover, if eventually some form of priority is implemented, i=
 would guess that data with higher priority would be retransmitted earlier =
than data with lower priority. I think such flexibility would be useful.<br=
><br></blockquote><div><br></div></span><div>Completely agreed, though I&#3=
9;m not sure that needs to be specified in this doc or left up to implement=
ations?</div><span class=3D"gmail-im"><div>=C2=A0</div><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(2=
04,204,204);padding-left:1ex">Let me see if I understand the overall framew=
ork for loss detection and retransmission for quic. I think that quic detec=
t packet loss, but retransmits data (that was sent in packets) so it probab=
ly possible to decouple the loss detection from the retransmission.<br><br>=
</blockquote></span><div><br>Yes, very much so.</div><span class=3D"gmail-i=
m"><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">I und=
erstand that all data (control and application data) that is reliably trans=
mitted is transmitted within a stream. Other data that is not transmitted w=
ithin a given stream (like ACKs) is not reliably transmitted. Is this corre=
ct? If so, it may be worth stating it as a design principle, so people know=
 that if they want to include control information such as other options to =
be reliably transmitted, they should do it within a stream.<br><br></blockq=
uote><div><br></div></span><div>It&#39;s correct that acks and stop waiting=
 frames are not reliable frames and hence are not retransmitted, but other =
frame types are retransmittable and not be part of a stream(ie: PING frames=
). =C2=A0</div><span class=3D"gmail-im"><div><br></div><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(2=
04,204,204);padding-left:1ex">So, if a sender has M ongoing streams, then i=
t has M+1 sliding windows, i.e. one for each stream plus one for the packet=
s at the connection level. Let&#39;s call the packet one, the packet window=
 and the other the stream n window. The packet window is in packets, while =
the stream windows are in bytes.<br><br></blockquote><div><br></div></span>=
<div>Yes.</div><span class=3D"gmail-im"><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">The packet window has the following types =
of packets: packets acked, non-acked packets and lost packets. The left sid=
e of the window moves to packet number n when all packets with packet numbe=
r lower than n are marked as acked or as lost. In other words, the left sid=
e of the window is the non-acked packet with lowest packet number. This cor=
responds to the &quot;least unacked&quot; packet mentioned in the draft.<br=
><br>Each stream window has the following types of bytes. bytes acked, byte=
s non-acked, bytes awaiting for transmission. Bytes non-acked are bytes tha=
t are contained in packets on flight. Bytes awaiting for transmission are b=
ytes that either have not been sent yet or bytes that have been sent in pac=
kets that have been determined as lost.<br><br>Whenever a packet is acked, =
the bytes in each stream window corresponding to the data of each stream in=
cluded in the acked packet are marked as acked.<br><br>Whenever a packet is=
 determined as lost, the bytes in each stream window corresponding to the d=
ata of that stream included in the packet are marked as bytes awaiting for =
transmission.<br><br>Whenever a new packet is transmitted, the bytes in eac=
h stream window included in the packet are marked as non-acked.<br><br>Pack=
et loss detection only operates using the packet window. So, each non-acked=
 packet has an associated timer that when it times out, the packet is marke=
d as lost. You can define any other mechanisms to detect packet loss only o=
perating in this window, i believe.<br></blockquote><div><br></div></span><=
div>Yes, the default mechanism actually uses FACK style packet number gaps,=
 but time is an option and the important thing is loss detection operates o=
n this connection-level packet window.</div><span class=3D"gmail-im"><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"><br>Packet tr=
ansmission otoh, needs to get data from the different stream windows, proba=
bly using the bytes awaiting for transmission with the lowest offset. This =
would include both the transmission of new data as well as retransmission o=
f old data.<br><br>Do you think this properly describes the framework of op=
eration for loss detection and retransmission of quic?<br><br></blockquote>=
<div><br></div></span><div>I think that captures the core ideas, yes.=C2=A0=
</div><div>=C2=A0<br></div><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=
">Regards, marcelo</blockquote></div></div>

--001a114dacbc6d25ba053e4b36d4--


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To: Ian Swett <ianswett@google.com>
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From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Date: Fri, 7 Oct 2016 21:27:14 +0200
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Cc: draft-iyengar-quic-loss-recovery@ietf.org, quic@ietf.org
Subject: Re: [QUIC] About loss detection and recovery in draft-iyengar-quic-loss-recovery-00
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thanks for the reply
a few follow ups below...

El 07/10/16 a las 21:13, Ian Swett escribió:
>
>     Moreover, if eventually some form of priority is implemented, i
>     would guess that data with higher priority would be retransmitted
>     earlier than data with lower priority. I think such flexibility
>     would be useful.
>
>
>
>
> Completely agreed, though I'm not sure that needs to be specified in 
> this doc or left up to implementations?

mmm, probably you should allow the applications to express somehow the 
priority of the data so that quic can distribute them into the stream 
with the right priority , so i guess at least you would need to define 
this in the API, no?


>
>     I understand that all data (control and application data) that is
>     reliably transmitted is transmitted within a stream. Other data
>     that is not transmitted within a given stream (like ACKs) is not
>     reliably transmitted. Is this correct? If so, it may be worth
>     stating it as a design principle, so people know that if they want
>     to include control information such as other options to be
>     reliably transmitted, they should do it within a stream.
>
>
> It's correct that acks and stop waiting frames are not reliable frames 
> and hence are not retransmitted, but other frame types are 
> retransmittable and not be part of a stream(ie: PING frames).
>

I fail to see the point of reliably transmitting pings. I mean, a ping 
is a test packet, if it fails, i guess this is the test output, no?
you may want to send anohter ping, is basically you want to do anohter 
test. There is no point in retransmitting the old one (conceptually, i mean)

In any case, I understand you are saying that there is some information 
that is not transmitted in streams that needs to be reliably 
transmitted, is that correct?
I wonder if this doenst creates a bit of a mess and wouldnt be cleaner 
to simply state that reliable transmission is only achieved in the 
streams and other frames transmitted outside a stream are non reliably 
transmitted. I dont really understand the implications of this design 
choice, but seems appealing because seems like a cleanner design choice.

>     So, if a sender has M ongoing streams, then it has M+1 sliding
>     windows, i.e. one for each stream plus one for the packets at the
>     connection level. Let's call the packet one, the packet window and
>     the other the stream n window. The packet window is in packets,
>     while the stream windows are in bytes.
>
>
> Yes.
>
>     The packet window has the following types of packets: packets
>     acked, non-acked packets and lost packets. The left side of the
>     window moves to packet number n when all packets with packet
>     number lower than n are marked as acked or as lost. In other
>     words, the left side of the window is the non-acked packet with
>     lowest packet number. This corresponds to the "least unacked"
>     packet mentioned in the draft.
>
>     Each stream window has the following types of bytes. bytes acked,
>     bytes non-acked, bytes awaiting for transmission. Bytes non-acked
>     are bytes that are contained in packets on flight. Bytes awaiting
>     for transmission are bytes that either have not been sent yet or
>     bytes that have been sent in packets that have been determined as
>     lost.
>
>     Whenever a packet is acked, the bytes in each stream window
>     corresponding to the data of each stream included in the acked
>     packet are marked as acked.
>
>     Whenever a packet is determined as lost, the bytes in each stream
>     window corresponding to the data of that stream included in the
>     packet are marked as bytes awaiting for transmission.
>
>     Whenever a new packet is transmitted, the bytes in each stream
>     window included in the packet are marked as non-acked.
>
>     Packet loss detection only operates using the packet window. So,
>     each non-acked packet has an associated timer that when it times
>     out, the packet is marked as lost. You can define any other
>     mechanisms to detect packet loss only operating in this window, i
>     believe.
>
>
> Yes, the default mechanism actually uses FACK style packet number 
> gaps, but time is an option and the important thing is loss detection 
> operates on this connection-level packet window.

Talking about default mechanisms, have you consider using RACK as the 
default mechanism for quic?
It seems to me a very clean fit, since you already have per packet 
timers, no?


Regards, marcelo

>
>     Packet transmission otoh, needs to get data from the different
>     stream windows, probably using the bytes awaiting for transmission
>     with the lowest offset. This would include both the transmission
>     of new data as well as retransmission of old data.
>
>     Do you think this properly describes the framework of operation
>     for loss detection and retransmission of quic?
>
>
> I think that captures the core ideas, yes.
>
>     Regards, marcelo
>


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From: "Christian Huitema" <huitema@huitema.net>
To: "'marcelo bagnulo braun'" <marcelo@it.uc3m.es>, "'Ian Swett'" <ianswett@google.com>
References: <73a98c3a-aaad-e2d2-3932-1f1eed60ae16@it.uc3m.es> <CAKcm_gP9pKeHQ-B9KVYWJ22-xTpg+_VR2DWPbDuYZfmsHQXu5A@mail.gmail.com> <047a0e75-95d0-3ca6-0c9d-6fc52de7f0aa@it.uc3m.es>
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Cc: draft-iyengar-quic-loss-recovery@ietf.org, quic@ietf.org
Subject: Re: [QUIC] About loss detection and recovery in draft-iyengar-quic-loss-recovery-00
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On Friday, October 7, 2016 12:27 PM, Marcelo Bagnulo Braun wrote:
>
> I fail to see the point of reliably transmitting pings. I mean, a ping =

> is a test packet, if it fails, i guess this is the test output, no?
> you may want to send anohter ping, is basically you want to do anohter =

> test. There is no point in retransmitting the old one (conceptually, i =
mean)

The way I see it, the ACK mechanism of QUIC performs loss detection. =
Conceptually, upon detecting a loss, the protocol engine examines the =
content of the lost packet. (There are multiple ways to do that in =
practice.) Then, based on what was lost, the protocol engine decides =
further actions. For example, stream data will be retransmitted most of =
the time, but not always -- e.g., not retransmitted if the stream was =
aborted. If a control packet is missing, the equivalent control may be =
transmitted again, but this time with updated value. An example would be =
a stream flow control packet, for which it is preferable to send the =
up-to-date information rather than resend the old value.

The case of the ping is particular. The engine sends a ping to check =
that the other end is alive. Arguably, it knows that if it receives an =
acknowledgement for the packet containing the ping frame, or for any =
later packet. Pings will thus only need to be retransmitted in some kind =
of "tail loss probe" scenario resulting in a timeout. The ping frame is =
actually a great way to implement the tail loss probe algorithm.

By the way, we are bound to see implementations that don't bother with =
these subtleties and just retransmit the entire contents of the lost =
packets. We should verify that this suboptimal behavior does not cause a =
protocol error, e.g. if two consecutive flow control frames slightly =
differ...

-- Christian Huitema






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As some of you have no doubt seen, QUIC is now chartered as am IETF Working=
 Group:

https://datatracker.ietf.org/wg/quic/charter/

Welcome!

As a reminder, as an IETF list, activity here is covered by the NOTE WELL t=
erms; for more information, see:

https://www.ietf.org/about/note-well.html

This WG *will* be meeting at the Seoul IETF; we'll have a draft agenda out =
for comments soon, but we expect to go over the charter and the input docum=
ents there. If you have not already, please familiarise yourself with all o=
f them beforehand.

Our milestones are on an aggressive schedule, so we will be having a number=
 of Interim meetings (as highlighted in the charter). Details will be forth=
coming soon, but at this point we anticipate having at least three standalo=
ne Interims in 2017, roughly half-way between the IETF meetings (with the f=
irst in the mid-January to mid-Feburary timeframe), with the potential for =
Interims held adjacent to the three IETF meetings as well (in addition to m=
eeting during the Plenary weeks, of course).

As many other WGs have, we plan on using Github <https://github.com/> as th=
e repository for our documents and issues list(s). Best practices for using=
 Github in IETF work are still evolving, but we will take appropriate steps=
 to assure that a) no information is "locked" into Github, and b) it is pos=
sible to participate without disadvantage if you choose not to create a Git=
hub account. If this causes you concern, please contact the chairs.

We'll be in touch soon with more information, but in the meantime, please c=
ontinue to discuss the documents and implementation of them on the list. We=
 look forward to working with all of you.


Regards,

Lars Eggert and Mark Nottingham, QUIC Working Group Chairs=


From nobody Fri Oct 14 11:44:01 2016
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To: "Eggert, Lars" <lars@netapp.com>
Date: Fri, 14 Oct 2016 14:43:54 -0400
Thread-Topic: RE: [QUIC] Welcome to QUIC
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I have a question on the clarity of text around network management practice=
s as stated in the charter:

Text from the QUIC charter:

Current practices for network management of transport protocols
include the ability to apply access control lists (ACLs), hashing of
flows for equal-cost multipath routing (ECMP), directional signaling
of flows, signaling of flow setup and teardown, and the ability to
export information about flows for accounting purposes. The QUIC
protocol need not be defined to enable each of these abilities, or
enable them in the same way as they are enabled by TCP when used
with TLS 1.3, but the working group must consider the impact of the
protocol on network management practices, reflecting the tensions
described in RFC 7258.

The efficacy of WG's consideration on  striving to achieve a balance consis=
tent with tension described in RFC 7258 may find that its hands are tied gi=
ven the charter text "The QUIC protocol need not be defined to enable each =
of these abilities...". I may be missing, but what other mechanisms in the =
charter help ensure working group can guide to an outcome consistent RFC 72=
58? Specifically, the statement from RFC 7258 that says "Making  networks u=
nmanageable to mitigate PM is not an acceptable outcome,  but ignoring PM w=
ould go against the consensus documented here."

Thanks
Sanjay

-----Original Message-----
From: QUIC [mailto:quic-bounces@ietf.org] On Behalf Of Eggert, Lars
Sent: Monday, October 10, 2016 4:46 AM
To: quic@ietf.org<mailto:quic@ietf.org>
Subject: [E] [QUIC] Welcome to QUIC

As some of you have no doubt seen, QUIC is now chartered as am IETF Working=
 Group:

https://datatracker.ietf.org/wg/quic/charter/

Welcome!

As a reminder, as an IETF list, activity here is covered by the NOTE WELL t=
erms; for more information, see:

https://www.ietf.org/about/note-well.html

This WG *will* be meeting at the Seoul IETF; we'll have a draft agenda out =
for comments soon, but we expect to go over the charter and the input docum=
ents there. If you have not already, please familiarise yourself with all o=
f them beforehand.

Our milestones are on an aggressive schedule, so we will be having a number=
 of Interim meetings (as highlighted in the charter). Details will be forth=
coming soon, but at this point we anticipate having at least three standalo=
ne Interims in 2017, roughly half-way between the IETF meetings (with the f=
irst in the mid-January to mid-Feburary timeframe), with the potential for =
Interims held adjacent to the three IETF meetings as well (in addition to m=
eeting during the Plenary weeks, of course).

As many other WGs have, we plan on using Github <https://github.com/> as th=
e repository for our documents and issues list(s). Best practices for using=
 Github in IETF work are still evolving, but we will take appropriate steps=
 to assure that a) no information is "locked" into Github, and b) it is pos=
sible to participate without disadvantage if you choose not to create a Git=
hub account. If this causes you concern, please contact the chairs.

We'll be in touch soon with more information, but in the meantime, please c=
ontinue to discuss the documents and implementation of them on the list. We=
 look forward to working with all of you.


Regards,

Lars Eggert and Mark Nottingham, QUIC Working Group Chairs
_______________________________________________
QUIC mailing list
QUIC@ietf.org<mailto:QUIC@ietf.org>
https://www.ietf.org/mailman/listinfo/quic


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>I have a questio=
n on the clarity of text around network management practices as stated in t=
he charter:<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p clas=
s=3DMsoNormal><span style=3D'mso-fareast-language:ZH-TW'>Text from the QUIC=
 charter:<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'mso-fare=
ast-language:ZH-TW'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal style=
=3D'margin-left:.5in'><span lang=3DEN style=3D'font-size:11.5pt;font-family=
:"PT Serif";color:#222222;mso-fareast-language:ZH-TW'>Current practices for=
 network management of transport protocols <br>include the ability to apply=
 access control lists (ACLs), hashing of <br>flows for equal-cost multipath=
 routing (ECMP), directional signaling <br>of flows, signaling of flow setu=
p and teardown, and the ability to <br>export information about flows for a=
ccounting purposes. The QUIC <br>protocol <span style=3D'background:yellow;=
mso-highlight:yellow'>need not be defined to enable</span> each of these ab=
ilities, or <br>enable them in the same way as they are enabled by TCP when=
 used <br>with TLS 1.3, <span style=3D'background:yellow;mso-highlight:yell=
ow'>but the working group must consider the impact of the<br>protocol on ne=
twork management practices</span>, reflecting the tensions<br>described in =
RFC 7258.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN><o:p>&n=
bsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN>The efficacy of W=
G&#8217;s consideration on &nbsp;striving to achieve a balance consistent w=
ith tension described in RFC 7258 may find that its hands are tied given th=
e charter text &#8220;The QUIC protocol need not be defined to enable each =
of these abilities&#8230;&#8221;. I may be missing, but what other mechanis=
ms in the charter help ensure working group can guide to an outcome consist=
ent RFC 7258? Specifically, the statement from RFC 7258 that says &#8220;Ma=
king&nbsp; networks unmanageable to mitigate PM is not an acceptable outcom=
e,&nbsp; but ignoring PM would go against the consensus documented here.&#8=
221;<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN><o:p>&nbsp;<=
/o:p></span></p><p class=3DMsoNormal>Thanks<o:p></o:p></p><p class=3DMsoNor=
mal>Sanjay<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=
=3DMsoNormal>-----Original Message-----<br>From: QUIC [<a href=3D"mailto:qu=
ic-bounces@ietf.org">mailto:quic-bounces@ietf.org</a>] On Behalf Of Eggert,=
 Lars<br>Sent: Monday, October 10, 2016 4:46 AM<br>To: <a href=3D"mailto:qu=
ic@ietf.org">quic@ietf.org</a><br>Subject: [E] [QUIC] Welcome to QUIC<o:p><=
/o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>As =
some of you have no doubt seen, QUIC is now chartered as am IETF Working Gr=
oup:<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMso=
Normal><a href=3D"https://datatracker.ietf.org/wg/quic/charter/">https://da=
tatracker.ietf.org/wg/quic/charter/</a><o:p></o:p></p><p class=3DMsoNormal>=
<o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Welcome!<o:p></o:p></p><p class=
=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>As a reminder, as an=
 IETF list, activity here is covered by the NOTE WELL terms; for more infor=
mation, see:<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p cla=
ss=3DMsoNormal><a href=3D"https://www.ietf.org/about/note-well.html">https:=
//www.ietf.org/about/note-well.html</a><o:p></o:p></p><p class=3DMsoNormal>=
<o:p>&nbsp;</o:p></p><p class=3DMsoNormal>This WG *will* be meeting at the =
Seoul IETF; we'll have a draft agenda out for comments soon, but we expect =
to go over the charter and the input documents there. If you have not alrea=
dy, please familiarise yourself with all of them beforehand.<o:p></o:p></p>=
<p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Our mileston=
es are on an aggressive schedule, so we will be having a number of Interim =
meetings (as highlighted in the charter). Details will be forthcoming soon,=
 but at this point we anticipate having at least three standalone Interims =
in 2017, roughly half-way between the IETF meetings (with the first in the =
mid-January to mid-Feburary timeframe), with the potential for Interims hel=
d adjacent to the three IETF meetings as well (in addition to meeting durin=
g the Plenary weeks, of course).<o:p></o:p></p><p class=3DMsoNormal><o:p>&n=
bsp;</o:p></p><p class=3DMsoNormal>As many other WGs have, we plan on using=
 Github &lt;<a href=3D"https://github.com/">https://github.com/</a>&gt; as =
the repository for our documents and issues list(s). Best practices for usi=
ng Github in IETF work are still evolving, but we will take appropriate ste=
ps to assure that a) no information is &quot;locked&quot; into Github, and =
b) it is possible to participate without disadvantage if you choose not to =
create a Github account. If this causes you concern, please contact the cha=
irs.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMso=
Normal>We'll be in touch soon with more information, but in the meantime, p=
lease continue to discuss the documents and implementation of them on the l=
ist. We look forward to working with all of you.<o:p></o:p></p><p class=3DM=
soNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Regards,<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</=
o:p></p><p class=3DMsoNormal>Lars Eggert and Mark Nottingham, QUIC Working =
Group Chairs <o:p></o:p></p><p class=3DMsoNormal>__________________________=
_____________________<o:p></o:p></p><p class=3DMsoNormal>QUIC mailing list<=
o:p></o:p></p><p class=3DMsoNormal><a href=3D"mailto:QUIC@ietf.org">QUIC@ie=
tf.org</a><o:p></o:p></p><p class=3DMsoNormal><a href=3D"https://www.ietf.o=
rg/mailman/listinfo/quic">https://www.ietf.org/mailman/listinfo/quic</a><o:=
p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>=

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hi Sanjay,

> On 14 Oct 2016, at 20:43, Mishra, Sanjay <sanjay.mishra@verizon.com> =
wrote:
>=20
> I have a question on the clarity of text around network management =
practices as stated in the charter:
>=20
> Text from the QUIC charter:
>=20
> Current practices for network management of transport protocols
> include the ability to apply access control lists (ACLs), hashing of
> flows for equal-cost multipath routing (ECMP), directional signaling
> of flows, signaling of flow setup and teardown, and the ability to
> export information about flows for accounting purposes. The QUIC
> protocol need not be defined to enable each of these abilities, or
> enable them in the same way as they are enabled by TCP when used
> with TLS 1.3, but the working group must consider the impact of the
> protocol on network management practices, reflecting the tensions
> described in RFC 7258.
>=20
> The efficacy of WG=E2=80=99s consideration on  striving to achieve a =
balance consistent with tension described in RFC 7258 may find that its =
hands are tied given the charter text =E2=80=9CThe QUIC protocol need =
not be defined to enable each of these abilities=E2=80=A6=E2=80=9D. I =
may be missing, but what other mechanisms in the charter help ensure =
working group can guide to an outcome consistent RFC 7258? Specifically, =
the statement from RFC 7258 that says =E2=80=9CMaking  networks =
unmanageable to mitigate PM is not an acceptable outcome,  but ignoring =
PM would go against the consensus documented here."

Since many of the words you seek to clarify come from language helped =
write to be added to the charter, let me try to clarify the intent, and =
explain what I see as the way to ensure an outcome consistent with =
7258's recognition that realization of the extent of large-scale =
pervasive surveillance does not change the fact that manageability is a =
first-order requirement of network protocols.

Since much of the controversy on this point at the BoF was that "network =
management and operations" was too vague, we wanted to be concrete about =
what kinds of network management practices should be considered in =
designing a new transport protocol. The five chosen are widely deployed, =
well understood, and relatively uncontroversial.

A requirement to support every one of these operations exactly as they =
are presently supported by TCP would constrain the QUIC design space too =
tightly: it would have to look and behave exactly like vanilla TCP on =
the wire. We already have a transport protocol that does that. It's =
called TCP.

QUIC has one advantage that TCP doesn't: it can fall back to TLS over =
TCP. Indeed, it is defined in such a way that applications that cannot =
handle a fall back to TLS over TCP cannot run over it. We have to do =
this to handle UDP blocking (relatively common, about 5%) and other =
performance impairment (much rarer, still present). I personally think =
that this should be seen as the starting point of an indefinite =
transition, as opposed to a permanent convenience; the latter risks =
stalling the deployment of the protocol. (Fallbacks are great for =
reliability but terrible for ensuring deployment. Case in point, I'm a =
huge fan of Happy Eyeballs, but last time I looked about 10% of the =
sites in the Alexa top million that bothered to publish AAAA records had =
them point to nonsensical places, or weren't listening where they =
thought they were. Happy Eyeballs means nobody notices, so these =
problems never get fixed.)

But IETF-standard QUIC doesn't exist yet, in-network devices that =
support these management functions in probably-somewhat-TCP-like but =
not-TCP-wire-compatible don't either. They'll have to be developed in =
tandem with the protocol. And if the working group finds that there is a =
tradeoff -- for example, the working group may find that either flow =
directionality or flow setup and teardown signaling needs to be impaired =
or removed to meet some other more important requirement. Indeed, either =
directionality signaling or end signaling is probably enough to do most =
of what most firewalls need at layer 3/4. This text gives the working =
group the flexibility to do that without rechartering.

So I would interpret the text to mean that the working group must =
consider operations and management functions in the design of the =
protocol, especially how the protocol interacts with devices other than =
the endpoints for these five management functions, and needs good =
reasons to trade off against supporting them in the final protocol =
design. I would not interpret this text to mean "QUIC may be defined to =
enable *none* of these abilities".

Hope this helps, cheers,

Brian

> Thanks
> Sanjay
>=20
> -----Original Message-----
> From: QUIC [mailto:quic-bounces@ietf.org] On Behalf Of Eggert, Lars
> Sent: Monday, October 10, 2016 4:46 AM
> To: quic@ietf.org
> Subject: [E] [QUIC] Welcome to QUIC
>=20
> As some of you have no doubt seen, QUIC is now chartered as am IETF =
Working Group:
>=20
> https://datatracker.ietf.org/wg/quic/charter/
>=20
> Welcome!
>=20
> As a reminder, as an IETF list, activity here is covered by the NOTE =
WELL terms; for more information, see:
>=20
> https://www.ietf.org/about/note-well.html
>=20
> This WG *will* be meeting at the Seoul IETF; we'll have a draft agenda =
out for comments soon, but we expect to go over the charter and the =
input documents there. If you have not already, please familiarise =
yourself with all of them beforehand.
>=20
> Our milestones are on an aggressive schedule, so we will be having a =
number of Interim meetings (as highlighted in the charter). Details will =
be forthcoming soon, but at this point we anticipate having at least =
three standalone Interims in 2017, roughly half-way between the IETF =
meetings (with the first in the mid-January to mid-Feburary timeframe), =
with the potential for Interims held adjacent to the three IETF meetings =
as well (in addition to meeting during the Plenary weeks, of course).
>=20
> As many other WGs have, we plan on using Github <https://github.com/> =
as the repository for our documents and issues list(s). Best practices =
for using Github in IETF work are still evolving, but we will take =
appropriate steps to assure that a) no information is "locked" into =
Github, and b) it is possible to participate without disadvantage if you =
choose not to create a Github account. If this causes you concern, =
please contact the chairs.
>=20
> We'll be in touch soon with more information, but in the meantime, =
please continue to discuss the documents and implementation of them on =
the list. We look forward to working with all of you.
>=20
>=20
> Regards,
>=20
> Lars Eggert and Mark Nottingham, QUIC Working Group Chairs
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>=20
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic


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From nobody Wed Oct 19 15:26:24 2016
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Hi, all,

A couple of days ago, in a thread on the network management paragraph in =
the charter, I wrote:

> Indeed, either directionality signaling or end signaling is probably =
enough to do most of what most firewalls need at layer 3/4.

We've just posted a short draft on the signaling required to drive a =
generic, transport-independent state machine for flow-state-keeping =
network devices, as part of the PLUS effort:

https://tools.ietf.org/html/draft-trammell-plus-statefulness

Aside from the ability to associate packets to flows (generally via =
their five-tuple), this approach requires only an association signal (to =
indicate a receiving endpoint is interested in keeping a connection =
open) and a stop signal (to indicate either endpoint is shutting the =
connection down, and that state should be freed). Depending on how the =
association signal is encoded, it might also provide information about =
the directionality of a flow.

We note that QUIC in its current state would need only very small =
changes to support this state machine: the Connection ID already works =
as an association signal (albeit without directionality), and the Public =
Reset facility could be repurposed as a stop signal, if exposed on every =
association shutdown, as opposed to just abnormal ones.

This would be a first step toward the fulfillment of the charter goal to =
ensure that QUIC is manageable; especially as "flow setup and teardown" =
signaling is one of the example functions mentioned directly in the =
charter.

Cheers,

Brian

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From nobody Wed Oct 19 19:36:58 2016
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From: Martin Thomson <martin.thomson@gmail.com>
Date: Thu, 20 Oct 2016 13:36:54 +1100
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Subject: Re: [QUIC] draft-trammell-plus-statefulness
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You say:

   QUIC's cryptographic protocol, to be based on TLS
   [I-D.thomson-quic-tls], will provide the necessary integrity
   protection to drive the state machine.

This isn't currently true.  Any signal that you create will
necessarily be outside of the confidentiality protection envelope.  We
would have to change the packet protection structure so that was added
to the AAD.  We currently do not use the AAD, so it's a kinda big
change.

I like the formulation of the association signal idea.  This means
that you only need to include a connection ID in packet 0.  This
covers the current mechanism (connection ID in initial packets and the
combination of shorter or absent connection IDs and the 5-tuple in
later packets).  It also suggests the possibility that you might do
other things that make things easier or harder depending on where you
sit.

One possibility is that we could make it difficult to trace backwards
along a flow.  We could even make it impossible to trace a flow
without packet 0..n.  Of course, I can't imagine a use case for
either.  It does suggest that there's a trade-off here though.

One thing I'd like to see made a lot clearer is that you are talking
about a single flow.  There is no requirement for the network to be
able to identify packets across disparate flows as belonging to the
same connection (for multipath or even just for mobility).  I can
imagine no good reason to have on-path adversaries being able to
correlate flows on different paths (yep, that's a challenge folks, let
the counterarguments fly).  The main reason being that correlation
across flows would defeat some of the traffic analysis defenses we are
considering.


On 20 October 2016 at 09:26, Brian Trammell <ietf@trammell.ch> wrote:
> Hi, all,
>
> A couple of days ago, in a thread on the network management paragraph in =
the charter, I wrote:
>
>> Indeed, either directionality signaling or end signaling is probably eno=
ugh to do most of what most firewalls need at layer 3/4.
>
> We've just posted a short draft on the signaling required to drive a gene=
ric, transport-independent state machine for flow-state-keeping network dev=
ices, as part of the PLUS effort:
>
> https://tools.ietf.org/html/draft-trammell-plus-statefulness
>
> Aside from the ability to associate packets to flows (generally via their=
 five-tuple), this approach requires only an association signal (to indicat=
e a receiving endpoint is interested in keeping a connection open) and a st=
op signal (to indicate either endpoint is shutting the connection down, and=
 that state should be freed). Depending on how the association signal is en=
coded, it might also provide information about the directionality of a flow=
.
>
> We note that QUIC in its current state would need only very small changes=
 to support this state machine: the Connection ID already works as an assoc=
iation signal (albeit without directionality), and the Public Reset facilit=
y could be repurposed as a stop signal, if exposed on every association shu=
tdown, as opposed to just abnormal ones.
>
> This would be a first step toward the fulfillment of the charter goal to =
ensure that QUIC is manageable; especially as "flow setup and teardown" sig=
naling is one of the example functions mentioned directly in the charter.
>
> Cheers,
>
> Brian
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>


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From: Jana Iyengar <jri@google.com>
Date: Wed, 19 Oct 2016 22:49:16 -0700
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--001a11452b167e937a053f457c4a
Content-Type: text/plain; charset=UTF-8

Hi Marcelo,

Thanks much again for your detailed read of the document. Reponses inline:

On Thu, Oct 6, 2016 at 2:37 AM, marcelo bagnulo braun <marcelo@it.uc3m.es>
wrote:

> Hi,
>
>
> Thanks for the document. I have been reading draft-hamilton-quic-transport-protocol-00
> and I find it not easy to read, probably because it is an early version and
> mostly because of my own limitations.
>
> I attach a few comments and questions regarding the draft. Hope you find
> them useful.
>
>
> Section 3.5 on Stream and Connection Flow control.
>
> Is the offset announced for the connection the sum of the offsets
> announced for each of the stream of that connection?
>
> If so, it seems redundant to announce both of them, no?
>
> If it is not, I am uncertain how this would work. If the connection one is
> larger than the sum of the streams one, I am not sure how can the sender
> use the remaining buffer, since it cannot allocate it any existing stream.
> Is this meant to the be used for new streams? If the sum of the streams one
> is higher than the connection buffer, then i guess the sender needs to to a
> double check, first for available rcv window in the stream and then
> available rcv window in the connection. Not sure what is the gain of having
> both. Maybe it is worthwhile making explicit in this section why this is
> useful?
>

Yes, like HTTP/2 flow control
<http://httpwg.org/specs/rfc7540.html#rfc.section.5.2>, the offset is
announced at both the connection-level and the stream-level. The general
intuition is exactly the same as with HTTP/2 -- that the connection-level
window provides an upper bound on the memory commitment at the receiver for
this connection. The expectation is that this window will be larger than
any one stream's window. This is so that even if one stream were to be
blocked (meaning that the receiver was not draining a stream), that stream
would not consume the entire receiver buffer and allow other stream(s) to
proceed.

Does this make sense? This is basically what the text in the draft says,
but if you think it can be clarified further, let me know.


Section 3.7 on Resilience to NAT Rebinding.
>
> I assume that the same mechanisms that allow the connection to survive a
> nat rebinding would be also useful to support some simple forms of mobility?
>
> I mean, probably these mechanisms are enough to support the client
> changing IP address (for instance when doing a handover from cellular to
> wifi)?
>
> I guess they are not enough to support more complex mobility scenarios,
> such as simultaneous movement of both endpoints, but in any case, it may be
> worthwhile mentioning this mobility support in this section?
>

That's correct -- this is enough to handle what we call "connection
migration". I was considering making that section heading "Connection
Migration and Resilience to NAT Rebinding", so I'll do that.

On simultaneous mobility -- I have to think more about this, but it seems
plausible with the connection ID as well. Basically, since the key still
protects the header (including the connection ID), and proof of ownership
of the key should be enough to authenticate the endpoint, so I think this
is reasonable. I'll add a note about this to the spec as well.

This mechanism isn't adequate to do general-purpose multipath (MPTCP-style)
however. That will need more bells and whistles.


Section 4.1 on Version Negotiation
>
> I understand that during version negotiation, the following information is
> exchanged: version, version flag, connection ID, and in some cases the
> server's supported versions. I understand that all this information is
> exchanged before the crypto handshake, so it is exchange in clear text and
> without authentication. Since in section 3.6 it is claimed that packets are
> always authenticated, I guess we have a (mild) contradiction here (i
> understand you verify this later on once the crypto has been exchanged). In
> any case, maybe it is worthwhile to explicitly state in this section that
> this information is exchanged in plain text and non authenticated.


Yes, this is correct. Clarified text: "The version negotiation packet is
unencrypted and exchanged without packet-level authentication. Therefore,
in order to avoid downgrade attacks, the client and server MUST include
their supported versions later in their corresponding crypto handshake
data."

I've also modified text in 3.6: "Generally, QUIC packets are always
authenticated and typically the payload is fully encrypted. The parts of
the packet header which are not encrypted are still authenticated by the
receiver, so as to thwart any packet injection or manipulation by third
parties. Some early handshake packets, such as the version negotiation
packet, are not encrypted, but information sent in these unencrypted
handshake packets is later verified under crypto cover."

Also, the last paragraph of this section states that there are other
> transport parameters exchanged in the connection establishment. I assume
> that those are exchanged after the crypto, so they are encrypted and
> authenticated? Is there any other transport parameter exchanged in clear
> text? Probably it would be useful to make this explicit?
>

Yeah, currently the transport params sent by the client are sent in clear
text, but again, later verified since the key derivation includes the
handshake packets. This would be a requirement of the handshake protocol.
I've added some text in the Crypto Handshake Protocol Features section:
"Transport Negotiation: The crypto handshake MUST provide a mechanism for
the transport component to exchange transport parameters and Source Address
Tokens. To avoid downgrade attacks, the transport parameters sent and
received MUST be verified before the handshake completes successfully."


> In section 4.2.1.2 on Required Transport Parameters.
>
> One thing I am missing here (probably because suffering a TCP bias) is the
> initial packet sequence number to be used in the connection.
>
> Do you always use 0 as initial PSN? If so, this is because you assume that
> the crypto mechanisms established during the crypto handshake will prevent
> packet form previous connections to re-emerge later on and be confused for
> packets in the new connection? If this is the case, maybe you should state
> it explicitly in section 4.2.3?
>

Correct. Zero is always the initial PSN, and yes, there should be no
confusion with packets from other (previous) connections, since the key for
this connection should be different. This should be a requirement of the
crypto handshake, so thanks for pointing it out. I've added the following
to section 4.2.3: "The crypto handshake MUST ensure that the final
negotiated key is distinct for every connection between two endpoints."


> However, from reading section 4.2.2 it seems that multiple connections
> between two endpoints can reuse the same crypto material, is that the case?
>
> In this case, does each connection uses 0 as initial PSN? How can you
> guarantee in this case that old packets with same PSN will no re surface
> and appear as packet of a new connection, since they can be using the same
> PSN?
>

Multiple connections do not share the same final crypto key. What in
Section 4.2.2 led you to believe that?

I'll respond to the rest of the comments tomorrow.

Thanks again,
- jana

--001a11452b167e937a053f457c4a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra">Hi Marcelo,</div><div class=3D"=
gmail_extra"><br></div><div class=3D"gmail_extra">Thanks much again for you=
r detailed read of the document. Reponses inline:</div><div class=3D"gmail_=
extra"><br><div class=3D"gmail_quote">On Thu, Oct 6, 2016 at 2:37 AM, marce=
lo bagnulo braun <span dir=3D"ltr">&lt;<a href=3D"mailto:marcelo@it.uc3m.es=
" target=3D"_blank">marcelo@it.uc3m.es</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px so=
lid rgb(204,204,204);padding-left:1ex">Hi,<br>
<br>
<br>
Thanks for the document. I have been reading draft-hamilton-quic-transport-=
<wbr>protocol-00 and I find it not easy to read, probably because it is an =
early version and mostly because of my own limitations.<br>
<br>
I attach a few comments and questions regarding the draft. Hope you find th=
em useful.<br>
<br>
<br>
Section 3.5 on Stream and Connection Flow control.<br>
<br>
Is the offset announced for the connection the sum of the offsets announced=
 for each of the stream of that connection?<br>
<br>
If so, it seems redundant to announce both of them, no?<br>
<br>
If it is not, I am uncertain how this would work. If the connection one is =
larger than the sum of the streams one, I am not sure how can the sender us=
e the remaining buffer, since it cannot allocate it any existing stream. Is=
 this meant to the be used for new streams? If the sum of the streams one i=
s higher than the connection buffer, then i guess the sender needs to to a =
double check, first for available rcv window in the stream and then availab=
le rcv window in the connection. Not sure what is the gain of having both. =
Maybe it is worthwhile making explicit in this section why this is useful?<=
br></blockquote><div><br></div><div>Yes, like=C2=A0<a href=3D"http://httpwg=
.org/specs/rfc7540.html#rfc.section.5.2">HTTP/2 flow control</a>, the offse=
t is announced at both the connection-level and the stream-level. The gener=
al intuition is exactly the same as with HTTP/2 -- that the connection-leve=
l window provides an upper bound on the memory commitment at the receiver f=
or this connection. The expectation is that this window will be larger than=
 any one stream&#39;s window. This is so that even if one stream were to be=
 blocked (meaning that the receiver was not draining a stream), that stream=
 would not consume the entire receiver buffer and allow other stream(s) to =
proceed.</div><div><br></div><div>Does this make sense? This is basically w=
hat the text in the draft says, but if you think it can be clarified furthe=
r, let me know.</div><div><br></div><div><br></div><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,2=
04,204);padding-left:1ex">
Section 3.7 on Resilience to NAT Rebinding.<br>
<br>
I assume that the same mechanisms that allow the connection to survive a na=
t rebinding would be also useful to support some simple forms of mobility?<=
br>
<br>
I mean, probably these mechanisms are enough to support the client changing=
 IP address (for instance when doing a handover from cellular to wifi)?<br>
<br>
I guess they are not enough to support more complex mobility scenarios, suc=
h as simultaneous movement of both endpoints, but in any case, it may be wo=
rthwhile mentioning this mobility support in this section?<br></blockquote>=
<div><br></div><div>That&#39;s correct -- this is enough to handle what we =
call &quot;connection migration&quot;. I was considering making that sectio=
n heading &quot;Connection Migration and=C2=A0Resilience to NAT Rebinding&q=
uot;, so I&#39;ll do that.=C2=A0</div><div><br></div><div>On simultaneous m=
obility -- I have to think more about this, but it seems plausible with the=
 connection ID as well. Basically, since the key still protects the header =
(including the connection ID), and proof of ownership of the key should be =
enough to authenticate the endpoint, so I think this is reasonable. I&#39;l=
l add a note about this to the spec as well.</div><div><br></div><div>This =
mechanism isn&#39;t adequate to do general-purpose multipath (MPTCP-style) =
however. That will need more bells and whistles.</div><div><br></div><div><=
br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex">
Section 4.1 on Version Negotiation<br>
<br>
I understand that during version negotiation, the following information is =
exchanged: version, version flag, connection ID, and in some cases the serv=
er&#39;s supported versions. I understand that all this information is exch=
anged before the crypto handshake, so it is exchange in clear text and with=
out authentication. Since in section 3.6 it is claimed that packets are alw=
ays authenticated, I guess we have a (mild) contradiction here (i understan=
d you verify this later on once the crypto has been exchanged). In any case=
, maybe it is worthwhile to explicitly state in this section that this info=
rmation is exchanged in plain text and non authenticated. </blockquote><div=
><br></div><div>Yes, this is correct. Clarified text: &quot;The version neg=
otiation packet is unencrypted and exchanged without packet-level authentic=
ation. Therefore, in order to avoid downgrade attacks, the client and serve=
r MUST include their supported versions later in their corresponding crypto=
 handshake data.&quot;</div><div><br></div><div>I&#39;ve also modified text=
 in 3.6: &quot;Generally, QUIC packets are always authenticated and typical=
ly the payload is fully encrypted. The parts of the packet header which are=
 not encrypted are still authenticated by the receiver, so as to thwart any=
 packet injection or manipulation by third parties. Some early handshake pa=
ckets, such as the version negotiation packet, are not encrypted, but infor=
mation sent in these unencrypted handshake packets is later verified under =
crypto cover.&quot;</div><div><br></div><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">Also, the last paragraph of this section states that there a=
re other transport parameters exchanged in the connection establishment. I =
assume that those are exchanged after the crypto, so they are encrypted and=
 authenticated? Is there any other transport parameter exchanged in clear t=
ext? Probably it would be useful to make this explicit?<br></blockquote><di=
v><br></div><div>Yeah, currently the transport params sent by the client ar=
e sent in clear text, but again, later verified since the key derivation in=
cludes the handshake packets. This would be a requirement of the handshake =
protocol. I&#39;ve added some text in the Crypto Handshake Protocol Feature=
s section: &quot;Transport Negotiation: The crypto handshake MUST provide a=
 mechanism for the transport component to exchange transport parameters and=
 Source Address Tokens. To avoid downgrade attacks, the transport parameter=
s sent and received MUST be verified before the handshake completes success=
fully.&quot;</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">
In section 4.2.1.2 on Required Transport Parameters.<br>
<br>
One thing I am missing here (probably because suffering a TCP bias) is the =
initial packet sequence number to be used in the connection.<br>
<br>
Do you always use 0 as initial PSN? If so, this is because you assume that =
the crypto mechanisms established during the crypto handshake will prevent =
packet form previous connections to re-emerge later on and be confused for =
packets in the new connection? If this is the case, maybe you should state =
it explicitly in section 4.2.3?<br></blockquote><div><br></div><div>Correct=
. Zero is always the initial PSN, and yes, there should be no confusion wit=
h packets from other (previous) connections, since the key for this connect=
ion should be different. This should be a requirement of the crypto handsha=
ke, so thanks for pointing it out. I&#39;ve added the following to section =
4.2.3: &quot;The crypto handshake MUST ensure that the final negotiated key=
 is distinct for every connection between two endpoints.&quot;</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">
However, from reading section 4.2.2 it seems that multiple connections betw=
een two endpoints can reuse the same crypto material, is that the case?<br>
<br>
In this case, does each connection uses 0 as initial PSN? How can you guara=
ntee in this case that old packets with same PSN will no re surface and app=
ear as packet of a new connection, since they can be using the same PSN?<br=
></blockquote><div><br></div><div>Multiple connections do not share the sam=
e final crypto key. What in Section 4.2.2 led you to believe that?</div><di=
v><br></div><div>I&#39;ll respond to the rest of the comments tomorrow.</di=
v><div><br>Thanks again,</div><div>- jana</div></div></div></div>

--001a11452b167e937a053f457c4a--


From nobody Thu Oct 20 03:19:26 2016
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From: Brian Trammell <ietf@trammell.ch>
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To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [QUIC] draft-trammell-plus-statefulness
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hi Martin, all,

> On 20 Oct 2016, at 04:36, Martin Thomson <martin.thomson@gmail.com> =
wrote:
>=20
> You say:
>=20
>   QUIC's cryptographic protocol, to be based on TLS
>   [I-D.thomson-quic-tls], will provide the necessary integrity
>   protection to drive the state machine.
>=20
> This isn't currently true.  Any signal that you create will
> necessarily be outside of the confidentiality protection envelope.  We
> would have to change the packet protection structure so that was added
> to the AAD.  We currently do not use the AAD, so it's a kinda big
> change.

Right, but aren't all the QUIC header bits also integrity-protected?

> I like the formulation of the association signal idea.  This means
> that you only need to include a connection ID in packet 0.

You *could* do this, at the expense of having to do a fast-reset =
(resending the connection ID, to cause a connection ID echo) if you =
detect connection disruption due to state loss (restart) of a device on =
path. This is no better than the behavior of TCP with stateful firewalls =
that need to see a SYN.

> This
> covers the current mechanism (connection ID in initial packets and the
> combination of shorter or absent connection IDs and the 5-tuple in
> later packets).  It also suggests the possibility that you might do
> other things that make things easier or harder depending on where you
> sit.

Yep. We tried to define association signals in the abstract to carry the =
*minimum* possible information. If a protocol piggybacks association on =
other information it radiates, there are of course tradeoffs in the =
other things you want to do with that information.

> One possibility is that we could make it difficult to trace backwards
> along a flow.  We could even make it impossible to trace a flow
> without packet 0..n.  Of course, I can't imagine a use case for
> either.  It does suggest that there's a trade-off here though.

Yep.

> One thing I'd like to see made a lot clearer is that you are talking
> about a single flow.

Hm, I was pretty sure that was already pretty clear, what with leading =
off the terminology section with the definition of a Flow cribbed from =
7011. Text suggestions welcome :)

> There is no requirement for the network to be
> able to identify packets across disparate flows as belonging to the
> same connection (for multipath or even just for mobility).  I can
> imagine no good reason to have on-path adversaries being able to
> correlate flows on different paths (yep, that's a challenge folks, let
> the counterarguments fly). The main reason being that correlation
> across flows would defeat some of the traffic analysis defenses we are
> considering.

Agreed completely, noting that traffic analysis defense is extremely =
hard to do if you have latency and bandwidth efficiency requirements =
too. That's not a reason to make it easier for no reason, though.

Cheers,

Brian


> On 20 October 2016 at 09:26, Brian Trammell <ietf@trammell.ch> wrote:
>> Hi, all,
>>=20
>> A couple of days ago, in a thread on the network management paragraph =
in the charter, I wrote:
>>=20
>>> Indeed, either directionality signaling or end signaling is probably =
enough to do most of what most firewalls need at layer 3/4.
>>=20
>> We've just posted a short draft on the signaling required to drive a =
generic, transport-independent state machine for flow-state-keeping =
network devices, as part of the PLUS effort:
>>=20
>> https://tools.ietf.org/html/draft-trammell-plus-statefulness
>>=20
>> Aside from the ability to associate packets to flows (generally via =
their five-tuple), this approach requires only an association signal (to =
indicate a receiving endpoint is interested in keeping a connection =
open) and a stop signal (to indicate either endpoint is shutting the =
connection down, and that state should be freed). Depending on how the =
association signal is encoded, it might also provide information about =
the directionality of a flow.
>>=20
>> We note that QUIC in its current state would need only very small =
changes to support this state machine: the Connection ID already works =
as an association signal (albeit without directionality), and the Public =
Reset facility could be repurposed as a stop signal, if exposed on every =
association shutdown, as opposed to just abnormal ones.
>>=20
>> This would be a first step toward the fulfillment of the charter goal =
to ensure that QUIC is manageable; especially as "flow setup and =
teardown" signaling is one of the example functions mentioned directly =
in the charter.
>>=20
>> Cheers,
>>=20
>> Brian
>>=20
>> _______________________________________________
>> QUIC mailing list
>> QUIC@ietf.org
>> https://www.ietf.org/mailman/listinfo/quic
>>=20


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Thu, 20 Oct 2016 21:43:16 +1100
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Subject: Re: [QUIC] draft-trammell-plus-statefulness
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Lifting a choice piece.

On 20 October 2016 at 21:19, Brian Trammell <ietf@trammell.ch> wrote:
> Right, but aren't all the QUIC header bits also integrity-protected?

Let's quickly assess these:
1. connection ID - routing critical, so I didn't see a need for
protection (though we would roll it into the handshake if it was
needed - most fall into this category)
2. version - rolled into the handshake
3. sequence number - forms the nonce (see TLS for analysis of this)
4. public reset etc...  - can't

I realize now that I didn't write down the analysis of the first, so
that could be wrong.

Just thinking about this, if you signal end-of-flow in the clear and
use the same techique for integrity as the version (corroboration
rather than a cryptographic device), you can watch for a close_notify
alert in the TLS to corroborate the cleartext signal.

But that leads me to ask: how do you signal end-of-flow here?  The
close_notify will be reliable and therefore subject to repair.  But if
the middlebox sees the first close_notify with the cleartext
end-of-flow marker, your model has it transition immediately to the
zero state.  I don't think that's a good outcome.

If the close_notify is actually lost we don't have an authenticated
end to communications.  Sure, that's much less of a problem from a
robustness perspective with the solid stream semantics in h2/quic, but
it might lead to clients attempting to repair a connection that is
unwanted.

Enter TIME_WAIT.


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On Thu, Oct 20, 2016 at 3:19 AM, Brian Trammell <ietf@trammell.ch> wrote:

> Right, but aren't all the QUIC header bits also integrity-protected?
>

Yes indeed. All of the public QUIC header bits are integrity protected with
the current code. I believe the plan with TLS is to provide the same
integrity protect, just using the key derived during the TLS handshake. Of
course, this only applies to post-handshake packets. =E2=80=8B

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Thu, Oct 20, 2016 at 3:19 AM, Brian Trammell <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:ietf@trammell.ch" target=3D"_blank" class=3D"cremed">ietf@tram=
mell.ch</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 id=3D"=
:18f" class=3D"a3s aXjCH m157e19a4c592a953">Right, but aren&#39;t all the Q=
UIC header bits also integrity-protected?<br></div></blockquote></div><br><=
div class=3D"gmail_default" style=3D"font-family:&quot;trebuchet ms&quot;,s=
ans-serif">Yes indeed. All of the public QUIC header bits are integrity pro=
tected with the current code. I believe the plan with TLS is to provide the=
 same integrity protect, just using the key derived during the TLS handshak=
e. Of course, this only applies to post-handshake packets. =E2=80=8B</div><=
br></div></div>

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From: Jana Iyengar <jri@google.com>
Date: Thu, 20 Oct 2016 19:49:09 -0700
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Hi Marcelo,

Thank you again for your really careful and deep dive in to the spec!
Responses for rest of the comments inline:

Section 4.2.2 on Proof of Source Address Ownership.
>
> I understand that the STK provides less protection than a reachability
> check at the beginning of each connection, correct?
>

The STK provides proof of IP ownsership. A client includes the source
address token (which was issued by the server to it in a previous
connection) to demonstrate ownership of it's IP address. If the client has
moved to a different IP address, if the token is too old, or if the client
doesn=E2=80=99t have a token, then the server rejects the connection and is=
sues a
fresh token to the client. This is very similar to TFO's cookie, as the
draft notes.

I mean, the STK proves that the client was once in that location and it can
> later leave the location and because it still has the STK, it can, by
> spoofing the source address, pretend to still be there, since only the ST=
K
> is enough to pass this test.
>
> This means that for instance, an attacker can download a large file by
> simply sending paced ACKs to any location it once was (given that the PSN
> initial number is always 0, it doesnt even needs to guess the PSNs),
> correct?
>
> Maybe this is acceptable, but maybe it is worthwhile stating it in the
> security considerations section?
>

Yes, this is a legitimate attack vector. I'll note (also in the draft) that
there are two simple mitigations. I've added the following text in Security
Considerations. Make sense?
"
Spoofed Ack Attack

An attacker steals the victim's STK, spoofs the victim's IP, and initiates
a 0-RTT connection with a server on the victim's behalf. The attacker then
spoofs ack packets to the server which cause the server to potentially
drown the victim in data.


There are two possible mitigations to this attack. The simplest one is that
a server can unilaterally create a gap in the PSN space. In the non-attack
scenario, the client will send back NACKs for the PSNs in the gap, but the
server will eventually send a STOP_WAITING frame for those packets. In the
attack scenario, the attacker may ack a packet in the gap. If the server
sees an ack for a packet that was never sent, the connection can be aborted=
.


The second mitigation is that the server can require that acks for sent
packets match the encryption level of the sent packet. This mitigation is
useful if the connection has an ephemeral forward-secure key that is
generated and used for every new connection. If a packet sent is encrypted
with a forward-secure key, then any acks that are received for them must
also be forward-secure encrypted. Since the attacker will not have the
forward secure key, the attacker will not be able to generate
forward-secure encrypted ack packets.
"


Section 4.2.3 on Crypto Handshake Protocol features
>
> In the bullet about Connection establishment in 0-RTT, it requires 0-RTT
> connection establishment for subsequent connections between the same two
> endpoints, in the subsequent connections, the client and server roles mus=
t
> be preserved or they can be changed?

I mean, suppose that host A initiates a quic connection to host b. Does
> this condition requires that a future connection (with the same ports)
> initiated from host b to host A should be established in 0-RTT?
>
> I think this should be made explicitly because mixed with the STK, I
> believe it changes the threat model significantly.
>

The presumption certainly was that the roles are preserved. With QUIC's
current (non-TLS) crypto handshake, there are very clear roles, and I
believe that TLS also has a clear expectation of client/server roles. That
said, I think this consideration belongs in either the TLS mapping document
(or in TLS itself) since that document should handle any threat models that
arise from it's 0-RTT implementation with an STK and/or TFO cookie. Is
there any clarification that you think might be useful to add to the crypto
handshake requirements section?


> Nit: the paragraph on the bullet on Server Config Update is truncated.
>

Removed the truncated sentence -- thanks!

Section 5.1 on Life of a stream.
>
> It states that quic accommodates the possibility of out-of-order delivery=
.
> Do you mean out of order delivery within the same stream or across stream=
s?
> clarify?
>

Across streams. I've changed the text to: "the possibility of out-of-order
delivery due to the use of multiple streams in QUIC"

I understand that the creation of a stream is very lightweight and a new
> stream is established by simply data with a new stream id, correct?
>

Correct.


> I also understand that the initial stream sequence number is always 0,
> correct? (I mean, the offset of the current data to the beginning of the
> streams is included, and the initial packet has offset 0).
>

Correct.


> So, similar to the question i posed about the initial PSN, how does quic
> prevent old packets to reappear and be confused as being part of the new
> stream while they were part of an older stream? It cannot be done using t=
he
> offset, since for all connection it starts in 0. I guess it could be done
> using the PSN, but then the receiver needs to check that all PSN of packe=
ts
> for a new stream are higher than the PSN of the packet that terminated th=
e
> older stream, is this the proposed approach?

Probably it is worthwhile clarifying what are the means that quic deals
> with old packets re emerging after the connection/stream has ended, given
> that it does not relies on the sequence number to do so, afaict.
>

Within the connection, the Section 5.2 says " A QUIC endpoint MUST NOT
reuse a StreamID on a given connection," so such STREAM frames will be
considered duplicates. Do you think this can be clarified better somewhere?


> In particular, in section 5.1.1 it states that a stream is open by sendin=
g
> or receiving a data packet with that particular stream ID. I assume that
> the data packet MUST contain a 0 offset to create the new stream? is this
> correct?
>

Not necessarily, since the packet containing a STREAM frame with a 0 offset
may have been dropped in the network. A STREAM frame with an offset > 0 may
be accepted, but the data will not get delivered to the application until
the intiial gap within the stream is filled.

Section 5.2 on Stream Identifiers.
>
> I guess you should reserve a range of identifiers for future use, in case
> other "special" streams are needed in the future?
>

I'm not sure what the point of reserving any  would be, since if a special
stream is needed in the future, this usage can be negotiated with either an
option or with a version change.

Section 5.4 on Sending and Receiving data.
>
> Is the initial offset for the data packet sent on a stream set to zero? I=
f
> this is the case, would be good to state it.
>

I've added this sentence: "The first byte of data that is sent on a stream
has the stream offset 0."


Section 6. on Packetization and reliability. What is the initial PSN in a
> given connection? State it explicitly.
>

Added "The initial PSN used, at both the client and the server, MUST be 0.
That is, the first packet in both directions of the connection MUST have a
PSN of 0."


> It is stated that a sender MAY wait for a short period of time to bundle
> multiple frames. I guess it would be useful to provide guidance on what y=
ou
> consider a short period of time. Is that 500ms as in the delayed acks or
> are you thinking much less?
>

MUCH less. This is definitely a MAY, and I am not sure if recommending
something here makes sense, since the expectations in any deployment will
be different about how much waiting makes sense. How's the following: "An
implementation may use heuristics about expected application sending
behavior to determine whether and for how long to wait. This waiting period
is an implementation decision, and an implementation should be careful to
delay conservatively, since any delay is likely to increase
application-visible latency."

Later on, it states that only ACK, STOP_WAITING and PADDING frames MUST not
> be retransmitted, but all other frames must be retransmitted. This implie=
s
> than that WINDOW_UPDATE and BLOCKED frames must be retransmitted, but i a=
m
> not sure if this makes sense. I mean, maybe there were other WINDOW_UPDAT=
E
> frames than were sent after this one so retransmitting this one probably
> doesnt make sense, since the information it contains may be stalled, no?
>

You're correct that WINDOW_UPDATE and BLOCKED frames are retransmitted.
It's hard to construct a scenario however where a WINDOW_UPDATE frame keeps
getting dropped, but the client still sends enough data to cause a new
revised WINDOW_UPDATE frame to be sent... I agree that in this case the
sender would be retransmitted useless information, but I don't think we
need to optimize for this case.

I still dont understand why having both connection and stream flow control
> is useful. I would assume that only stream flow control would be useful. =
Or
> having one or the other in a given connection, but i fail to understand w=
hy
> is useful to have both of them simultaneously in the same connection.
>

I've answered this above, let me know if there's more clarification that
can be had in the doc.

Section 6 states
>
>    Regular QUIC Packets are "containers" of frames; a packet is never
>    retransmitted whole, but frames in a lost packet may be rebundled and
>    transmitted in a subsequent packet as necessary.
>
> So, let me see if i understand this.
>
> This means that if the sender sends packet with PSN 5, and if the packet
> is lost (i.e. there is a timeout for that packet), the packet with PSN 5
> NOT retransmitted. the frames in the packet with PSN 5 are then
> retransmitted in other packets with other PSN (other than 5). If this
> correct?
>

Correct.


> This means that there will be holes in the sequence of packets received b=
y
> the receiver i.e. any packet that get's lost, is simply lost and will nev=
er
> be received by the receiver. Am i understanding this correctly?
>

Correct.


>    A receiver acknowledges receipt of a received packet by sending one
>    or more ACK frames containing the PSN of the received packet.  To
>    avoid perpetual acking between endpoints, a receiver MUST NOT
>    generate an ack in response to every packet containing only ACK
>    frames.
>
> Does this means that ACKs transmission is unreliable, correct?
>

Correct.


> Since packets are never retransmitted with the same PSN, this basically
> means that if an ACK is lost, then the sender gives up on this packet and
> eventually will retransmit the data in other packets with a different PSN=
,
> correct?
>

Correct.


>
> Section 7.1.1 on Mid-stream RST_STREAM
>
> After a RST, when the stream identifier can be reused? probably some
> guidance about this would be useful.
>

As I clarified above, a stream ID can never be reused.

Section 7.1.4.
>
> Is the BLOCKED frame reliably transmitted? I assume it is, but, does it
> make sense?
>

Yes, it is reliably transmitted.


> I mean, if a sender sends a blocked frame and it get's lost, but after
> that the receiver open the window, then retransmitting the BLOCKED frame =
is
> not useful (doesnt harm neither i guess)
>
> OTOH, what if the BLOCKED frame get's lost? Do we end up with the silly
> window syndrome?
>

The BLOCKED frame is really a debugging frame... it's good to know at the
receiver that the sender was flow control blocked. Since the receiver is
expected to have sent a WINDOW_UPDATE well in advance, and since the
BLOCKED frame is not a required part of the flow control machinery, it
doesn't cause SWS. It's currently required for the sender to send, but not
required to be processed by the receiver beyond maybe logging it somewhere.
I've modified the text in Section 7.1.4 to say:

"If a sender does not receive a WINDOW_UPDATE frame when it has run out of
flow control credit, the sender will be blocked and MUST send a BLOCKED
frame. A BLOCKED frame is expected to be useful for debugging at the
receiver. A receiver SHOULD NOT wait for a BLOCKED frame before sending
with a WINDOW_UPDATE, since doing so will cause at least one roundtrip of
quiescence. For smooth operation of the congestion controller, it is
generally considered best to not let the sender go into quiescence if
avoidable. To avoid blocking a sender, and to reasonably account for the
possibiity of loss, a receiver should send a WINDOW_UPDATE frame at least
two roundtrips before it expects the sender to get blocked."

Is that clearer?

I guess we could do what TCP does i.e. periodically transmit the BLOCKED
> msg when there is no window (which is different from reliably transmittin=
g
> the BLOCKED message, since it is periodically sent only when the window i=
s
> 0)
>
>
> Section 8 on connection termination
>
> After a connection close, does the same stream identifier can be reused
> for a new connection? shouldnt also be a hold time to avoid delayed packe=
ts
> of the old connection? How much time would be safe?
>

Do you mean Connection ID? Again, since the expectation is that the keys
will be different. Of course that depends on the KDF, but that depends on
the crypto protocol in use. I think the clarification that I added above to
the crypto requirements should be enough. What do you think?


> Section 10.4
>
> I understand ACK frames are unreliably transmitted.
>
> Also, I understand that packets are never retransmitted with the same PSN=
.
> So, if a packet is lost, there will be a hole in the receiver for that PS=
N
> and the data will be sent in other packets with othew PSN. Hence, you
> cannot have cummulative acks.
>

Correct.


> So, this means that it is possible for a packet to be received by the
> receiver but that all the ACKs for that packet to be lost on the way back
> and then the sender will timeout and retransmit, right? (I guess that is
> ok, since data will be retransmitted in other packets)
>

Correct.


> Let me verify that I understand when a stop waiting frames should be sent
> by the sender. I guess that a sender should send a given PSN X in a stop
> waiting frame if all the packet with PSN lower than X have been received =
or
> have timed out, correct?
>

Correct. I would say when all PSNs <=3D X are no longer considered in fligh=
t.


Thanks again,
- jana


> Regards, marcelo
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

--001a113ee41628e139053f5716cd
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">Hi M=
arcelo,</div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote=
">Thank you again for your really careful and deep dive in to the spec! Res=
ponses for rest of the comments inline:</div><div class=3D"gmail_quote"><br=
></div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex">
Section 4.2.2 on Proof of Source Address Ownership.<br>
<br>
I understand that the STK provides less protection than a reachability chec=
k at the beginning of each connection, correct?<br></blockquote><div><br></=
div><div>The STK provides proof of IP ownsership. A client includes the sou=
rce address token (which was issued by the server to it in a previous conne=
ction) to demonstrate ownership of it&#39;s IP address. If the client has m=
oved to a different IP address, if the token is too old, or if the client d=
oesn=E2=80=99t have a token, then the server rejects the connection and iss=
ues a fresh token to the client. This is very similar to TFO&#39;s cookie, =
as the draft notes.</div><div><br></div><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">
I mean, the STK proves that the client was once in that location and it can=
 later leave the location and because it still has the STK, it can, by spoo=
fing the source address, pretend to still be there, since only the STK is e=
nough to pass this test.<br>
<br>
This means that for instance, an attacker can download a large file by simp=
ly sending paced ACKs to any location it once was (given that the PSN initi=
al number is always 0, it doesnt even needs to guess the PSNs), correct?<br=
>
<br>
Maybe this is acceptable, but maybe it is worthwhile stating it in the secu=
rity considerations section?<br></blockquote><div><br></div><div>Yes, this =
is a legitimate attack vector. I&#39;ll note (also in the draft) that there=
 are two simple mitigations. I&#39;ve added the following text in Security =
Considerations. Make sense?</div><div>&quot;</div><div>Spoofed Ack Attack<b=
r><br>An attacker steals the victim&#39;s STK, spoofs the victim&#39;s IP, =
and initiates a 0-RTT connection with a server on the victim&#39;s behalf. =
The attacker then spoofs ack packets to the server which cause the server t=
o potentially drown the victim in data.<br><br><br>There are two possible m=
itigations to this attack. The simplest one is that a server can unilateral=
ly create a gap in the PSN space. In the non-attack scenario, the client wi=
ll send back NACKs for the PSNs in the gap, but the server will eventually =
send a STOP_WAITING frame for those packets. In the attack scenario, the at=
tacker may ack a packet in the gap. If the server sees an ack for a packet =
that was never sent, the connection can be aborted.<br><br><br>The second m=
itigation is that the server can require that acks for sent packets match t=
he encryption level of the sent packet. This mitigation is useful if the co=
nnection has an ephemeral forward-secure key that is generated and used for=
 every new connection. If a packet sent is encrypted with a forward-secure =
key, then any acks that are received for them must also be forward-secure e=
ncrypted. Since the attacker will not have the forward secure key, the atta=
cker will not be able to generate forward-secure encrypted ack packets.</di=
v><div>&quot;<br></div><div><br></div><div><br></div><blockquote class=3D"g=
mail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204=
,204,204);padding-left:1ex">
Section 4.2.3 on Crypto Handshake Protocol features<br>
<br>
In the bullet about Connection establishment in 0-RTT, it requires 0-RTT co=
nnection establishment for subsequent connections between the same two endp=
oints, in the subsequent connections, the client and server roles must be p=
reserved or they can be changed?</blockquote><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204=
);padding-left:1ex">I mean, suppose that host A initiates a quic connection=
 to host b. Does this condition requires that a future connection (with the=
 same ports) initiated from host b to host A should be established in 0-RTT=
?<br>
<br>
I think this should be made explicitly because mixed with the STK, I believ=
e it changes the threat model significantly.<br></blockquote><div><br></div=
><div><div>The presumption certainly was that the roles are preserved. With=
 QUIC&#39;s current (non-TLS) crypto handshake, there are very clear roles,=
 and I believe that TLS also has a clear expectation of client/server roles=
. That said, I think this consideration belongs in either the TLS mapping d=
ocument (or in TLS itself) since that document should handle=C2=A0any threa=
t models that arise from it&#39;s 0-RTT implementation with an STK and/or T=
FO cookie. Is there any clarification that you think might be useful to add=
 to the crypto handshake requirements section?</div><div>=C2=A0</div></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">Nit: the paragraph on the=
 bullet on Server Config Update is truncated.<br></blockquote><div><br></di=
v><div>Removed the truncated sentence -- thanks!</div><div><br></div><block=
quote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1=
px solid rgb(204,204,204);padding-left:1ex">
Section 5.1 on Life of a stream.<br>
<br>
It states that quic accommodates the possibility of out-of-order delivery. =
Do you mean out of order delivery within the same stream or across streams?=
 clarify?<br></blockquote><div><br></div><div>Across streams. I&#39;ve chan=
ged the text to: &quot;the possibility of out-of-order delivery due to the =
use of multiple streams in QUIC&quot;</div><div><br></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">
I understand that the creation of a stream is very lightweight and a new st=
ream is established by simply data with a new stream id, correct?<br></bloc=
kquote><div><br></div><div>Correct.</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">
I also understand that the initial stream sequence number is always 0, corr=
ect? (I mean, the offset of the current data to the beginning of the stream=
s is included, and the initial packet has offset 0).<br></blockquote><div><=
br></div><div>Correct.</div><div>=C2=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204=
);padding-left:1ex">
So, similar to the question i posed about the initial PSN, how does quic pr=
event old packets to reappear and be confused as being part of the new stre=
am while they were part of an older stream? It cannot be done using the off=
set, since for all connection it starts in 0. I guess it could be done usin=
g the PSN, but then the receiver needs to check that all PSN of packets for=
 a new stream are higher than the PSN of the packet that terminated the old=
er stream, is this the proposed approach?</blockquote><blockquote class=3D"=
gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(20=
4,204,204);padding-left:1ex">Probably it is worthwhile clarifying what are =
the means that quic deals with old packets re emerging after the connection=
/stream has ended, given that it does not relies on the sequence number to =
do so, afaict.<br></blockquote><div><br></div><div>Within the connection, t=
he Section 5.2 says &quot; A QUIC endpoint MUST NOT reuse a StreamID on a g=
iven connection,&quot; so such STREAM frames will be considered duplicates.=
 Do you think this can be clarified better somewhere?</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">
In particular, in section 5.1.1 it states that a stream is open by sending =
or receiving a data packet with that particular stream ID. I assume that th=
e data packet MUST contain a 0 offset to create the new stream? is this cor=
rect?<br></blockquote><div><br></div><div>Not necessarily, since the packet=
 containing a STREAM frame with a 0 offset may have been dropped in the net=
work. A STREAM frame with an offset &gt; 0 may be accepted, but the data wi=
ll not get delivered to the application until the intiial gap within the st=
ream is filled.</div><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">
Section 5.2 on Stream Identifiers.<br>
<br>
I guess you should reserve a range of identifiers for future use, in case o=
ther &quot;special&quot; streams are needed in the future?<br></blockquote>=
<div><br></div><div>I&#39;m not sure what the point of reserving any =C2=A0=
would be, since if a special stream is needed in the future, this usage can=
 be negotiated with either an option or with a version change.</div><div><b=
r></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">
Section 5.4 on Sending and Receiving data.<br>
<br>
Is the initial offset for the data packet sent on a stream set to zero? If =
this is the case, would be good to state it.<br></blockquote><div><br></div=
><div>I&#39;ve added this sentence: &quot;The first byte of data that is se=
nt on a stream has the stream offset 0.&quot;</div><div><br></div><div><br>=
</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">
Section 6. on Packetization and reliability. What is the initial PSN in a g=
iven connection? State it explicitly.<br></blockquote><div><br></div><div>A=
dded &quot;The initial PSN used, at both the client and the server, MUST be=
 0. That is, the first packet in both directions of the connection MUST hav=
e a PSN of 0.&quot;</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);p=
adding-left:1ex">
It is stated that a sender MAY wait for a short period of time to bundle mu=
ltiple frames. I guess it would be useful to provide guidance on what you c=
onsider a short period of time. Is that 500ms as in the delayed acks or are=
 you thinking much less?<br></blockquote><div><br></div><div>MUCH less. Thi=
s is definitely a MAY, and I am not sure if recommending something here mak=
es sense, since the expectations in any deployment will be different about =
how much waiting makes sense. How&#39;s the following: &quot;An implementat=
ion may use heuristics about expected application sending behavior=C2=A0to =
determine whether and for how long to wait. This waiting period is an imple=
mentation decision, and an implementation should be careful to delay conser=
vatively, since any delay is likely to increase application-visible latency=
.&quot;</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1e=
x">
Later on, it states that only ACK, STOP_WAITING and PADDING frames MUST not=
 be retransmitted, but all other frames must be retransmitted. This implies=
 than that WINDOW_UPDATE and BLOCKED frames must be retransmitted, but i am=
 not sure if this makes sense. I mean, maybe there were other WINDOW_UPDATE=
 frames than were sent after this one so retransmitting this one probably d=
oesnt make sense, since the information it contains may be stalled, no?<br>=
</blockquote><div><br></div><div>You&#39;re correct that WINDOW_UPDATE and =
BLOCKED frames are retransmitted. It&#39;s hard to construct a scenario how=
ever where a WINDOW_UPDATE frame keeps getting dropped, but the client stil=
l sends enough data to cause a new revised WINDOW_UPDATE frame to be sent..=
. I agree that in this case the sender would be retransmitted useless infor=
mation, but I don&#39;t think we need to optimize for this case.</div><div>=
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
I still dont understand why having both connection and stream flow control =
is useful. I would assume that only stream flow control would be useful. Or=
 having one or the other in a given connection, but i fail to understand wh=
y is useful to have both of them simultaneously in the same connection.<br>=
</blockquote><div><br></div><div>I&#39;ve answered this above, let me know =
if there&#39;s more clarification that can be had in the doc.</div><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">
Section 6 states<br>
<br>
=C2=A0 =C2=A0Regular QUIC Packets are &quot;containers&quot; of frames; a p=
acket is never<br>
=C2=A0 =C2=A0retransmitted whole, but frames in a lost packet may be rebund=
led and<br>
=C2=A0 =C2=A0transmitted in a subsequent packet as necessary.<br>
<br>
So, let me see if i understand this.<br>
<br>
This means that if the sender sends packet with PSN 5, and if the packet is=
 lost (i.e. there is a timeout for that packet), the packet with PSN 5 NOT =
retransmitted. the frames in the packet with PSN 5 are then retransmitted i=
n other packets with other PSN (other than 5). If this correct?<br></blockq=
uote><div><br></div><div>Correct.</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">
This means that there will be holes in the sequence of packets received by =
the receiver i.e. any packet that get&#39;s lost, is simply lost and will n=
ever be received by the receiver. Am i understanding this correctly?<br></b=
lockquote><div><br></div><div>Correct.</div><div>=C2=A0</div><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid=
 rgb(204,204,204);padding-left:1ex">
=C2=A0 =C2=A0A receiver acknowledges receipt of a received packet by sendin=
g one<br>
=C2=A0 =C2=A0or more ACK frames containing the PSN of the received packet.=
=C2=A0 To<br>
=C2=A0 =C2=A0avoid perpetual acking between endpoints, a receiver MUST NOT<=
br>
=C2=A0 =C2=A0generate an ack in response to every packet containing only AC=
K<br>
=C2=A0 =C2=A0frames.<br>
<br>
Does this means that ACKs transmission is unreliable, correct?<br></blockqu=
ote><div><br></div><div>Correct.</div><div>=C2=A0</div><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(2=
04,204,204);padding-left:1ex">
Since packets are never retransmitted with the same PSN, this basically mea=
ns that if an ACK is lost, then the sender gives up on this packet and even=
tually will retransmit the data in other packets with a different PSN, corr=
ect?<br></blockquote><div><br></div><div>Correct.</div><div>=C2=A0</div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-lef=
t:1px solid rgb(204,204,204);padding-left:1ex">
<br>
Section 7.1.1 on Mid-stream RST_STREAM<br>
<br>
After a RST, when the stream identifier can be reused? probably some guidan=
ce about this would be useful.<br></blockquote><div><br></div><div>As I cla=
rified above, a stream ID can never be reused.</div><div><br></div><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px=
 solid rgb(204,204,204);padding-left:1ex">
Section 7.1.4.<br>
<br>
Is the BLOCKED frame reliably transmitted? I assume it is, but, does it mak=
e sense?<br></blockquote><div><br></div><div>Yes, it is reliably transmitte=
d.</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 mean, if a sender sends a blocked frame and it get&#39;s lost, but after =
that the receiver open the window, then retransmitting the BLOCKED frame is=
 not useful (doesnt harm neither i guess)<br>
<br>
OTOH, what if the BLOCKED frame get&#39;s lost? Do we end up with the silly=
 window syndrome?<br></blockquote><div><br></div><div>The BLOCKED frame is =
really a debugging frame... it&#39;s good to know at the receiver that the =
sender was flow control blocked. Since the receiver is expected to have sen=
t a WINDOW_UPDATE well in advance, and since the BLOCKED frame is not a req=
uired part of the flow control machinery, it doesn&#39;t cause SWS. It&#39;=
s currently required for the sender to send, but not required to be process=
ed by the receiver beyond maybe logging it somewhere. I&#39;ve modified the=
 text in Section 7.1.4 to say:=C2=A0</div><div><br></div><div>&quot;If a se=
nder does not receive a WINDOW_UPDATE frame when it has run out of flow con=
trol credit, the sender will be blocked and MUST send a BLOCKED frame. A BL=
OCKED frame is expected to be useful for debugging at the receiver. A recei=
ver SHOULD NOT wait for a BLOCKED frame before sending with a WINDOW_UPDATE=
, since doing so will cause at least one roundtrip of quiescence. For smoot=
h operation of the congestion controller, it is generally considered best t=
o not let the sender go into quiescence if avoidable. To avoid blocking a s=
ender, and to reasonably account for the possibiity of loss, a receiver sho=
uld send a WINDOW_UPDATE frame at least two roundtrips before it expects th=
e sender to get blocked.&quot;</div><div><br></div><div>Is that clearer?</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">
I guess we could do what TCP does i.e. periodically transmit the BLOCKED ms=
g when there is no window (which is different from reliably transmitting th=
e BLOCKED message, since it is periodically sent only when the window is 0)=
<br>
<br>
<br>
Section 8 on connection termination<br>
<br>
After a connection close, does the same stream identifier can be reused for=
 a new connection? shouldnt also be a hold time to avoid delayed packets of=
 the old connection? How much time would be safe?<br></blockquote><div><br>=
</div><div>Do you mean Connection ID? Again, since the expectation is that =
the keys will be different. Of course that depends on the KDF, but that dep=
ends on the crypto protocol in use. I think the clarification that I added =
above to the crypto requirements should be enough. What do you think?</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">
Section 10.4<br>
<br>
I understand ACK frames are unreliably transmitted.<br>
<br>
Also, I understand that packets are never retransmitted with the same PSN. =
So, if a packet is lost, there will be a hole in the receiver for that PSN =
and the data will be sent in other packets with othew PSN. Hence, you canno=
t have cummulative acks.<br></blockquote><div><br></div><div>Correct.</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">
So, this means that it is possible for a packet to be received by the recei=
ver but that all the ACKs for that packet to be lost on the way back and th=
en the sender will timeout and retransmit, right? (I guess that is ok, sinc=
e data will be retransmitted in other packets)<br></blockquote><div><br></d=
iv><div>Correct.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padd=
ing-left:1ex">
Let me verify that I understand when a stop waiting frames should be sent b=
y the sender. I guess that a sender should send a given PSN X in a stop wai=
ting frame if all the packet with PSN lower than X have been received or ha=
ve timed out, correct?<br></blockquote><div><br></div><div>Correct. I would=
 say when all PSNs &lt;=3D X are no longer considered in flight.</div><div>=
<br></div><div><br></div><div>Thanks again,</div><div>- jana</div><div>=C2=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex">
Regards, marcelo<br>
<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org" target=3D"_blank">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/quic</a><br>
</blockquote></div><br></div></div>

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Hi Lars,

Thanks again for your implementation effort and detailed feedback, and
apologies for taking a while to respond. I'm happy to iterate, and I expect
that my turnaround will be much faster (unlike this time :-))

On Fri, Sep 23, 2016 at 2:47 AM, Eggert, Lars <lars@netapp.com> wrote:

> Hi,
>
> I've started to do a little QUIC implementation exercise, to see if
> the current set of documents is sufficient to allow independent
> implementation. As can be expected, they're not, but I wanted to post
> some feedback based on what I encountered at this very, very, VERY
> early phase, so the documents can be improved.
>

Thank you! This is what we're hoping for as well. Thank you for being a
willing early implementer.


> I'm purposefully avoiding to rely at documentation other than the IDs.
> I won't look at the Chromium code, and I'm only looking at the specs
> on Google Docs or wireshark outputs in order to more accurately
> describe what's currently missing in the IDs. However, I run my
> implementation against the Chromium quic_server and quic_client
> implementations, in order to see if they interoperate.
>

I anticipate some trouble in getting it to work with quic_client and
quic_server if you're just building from the spec, since these binaries use
the QUIC crypto handshake
<https://docs.google.com/document/d/1g5nIXAIkN_Y-7XJW5K45IblHd_L2f5LTaDUDwv=
Z5L6g/edit>,
and the spec currently does not specify one.


> (I may open source this pure C implementation at some point; at the
> moment, it's too embarrassing to do so.)
>
> I began by reading draft-hamilton-quic-transport-protocol-00, in order
> to implement the initial handshake. The ID is reasonably long and
> complete-looking. Sections 1, 2 and 3 are reasonable in terms of
> content.
>
> Section 4 (Connection Establishment) is difficult to make sense of
> without first reading Section 9 (Packet Types and Formats), because
> Section 4 refers to header fields, flags and other concepts that are
> only defined in Section 9. The content seems arbitrarily split between
> those sections, e.g., Section 4.1 attempts to specify version
> negotiation, but Section 9.2.1 describes the version negotiation
> packet format. One needs to constantly flip back and forth, which is
> annoying. I'd recommend reordering the content here.
>

I wasn't sure how to organize this content, so it's currently mechanisms in
early sections and wire formats later. From your experience it seems like
it may be better organized if the the wire format of the various packets
and frames were alongside the mechanisms. Does that sound like a good rule
of thumb? Given the size of the draft, that may be better, so I'm happy to
do it.

One thing that threw me was that Section 9.1 states "all QUIC packets
> on the wire begin with a public header" and then has a diagram of what
> that header looks like, only to immediately follow it in Section 9.2
> with special packets that don't fully include that public header.
> That's really confusing. Is there a way to describe the public header
> that would include the format of the special packets? Is there a way
> to make those special packets use the public header? They're so rarely
> sent that deviating to save a few bytes doesn't seem to be a smart
> tradeoff.
>

I think there may be a way to separate the common parts out. Specifically,
the flags and the connection ID are really the "common header", but the
reason for having these three packet types is because they differ in the
parts that are not encrypted (The parts that are encrypted should be
specified, and I'll add that.) Perhaps one way to describe them would be to
say:
- Common Header =3D Flags + (optional) Connection ID
- Regular Packets =3D Common Header + (optional) Version + Packet Number +
Frames
- Version Negotiation Packet =3D Common Header (Version Negotiation bit set=
)
+ Version List
- Public Reset Packet =3D Common Header (Reset bit set) + Optional tag-valu=
es

I realized that we don't describe the format for the optional tag-values in
the spec, but they're potentially useful to help authenticate the Reset.
That's entirely optional at the moment, but really the Public Reset packet
is identified by the bit in the flags and the connection is identified by
the Connection ID.


> Also, the document makes no mention of what endianness the protocol
> fields are supposed to be encoded in.
>

Good point -- I've added a sentence to the Packet Types and Formats saying
"All values are little-endian unless otherwise noted." This sentence will
likely move as the doc gets restructured, but that is the first place a
packet format is currently being described.

The ID mentions QUIC version "Q025", so my code attempts to negotiate
> this version with the Chromium quic_server, but that offers "Q036" and
> indicates support for "Q035Q034Q033Q032" as well. The negotiation
> response from the server does not conform to what's in the ID, since
> it has the "diversification nonce" flag set (0x04), which is not
> specified in Section 9.2.1. It also seems to use only 12 bytes of the
> fixed 32 bytes that the nonce should occupy to indicate support for
> other versions, according to Section 9.1.
>
> Also, that "diversification nonce" is not referred to anywhere else in
> the ID other than Section 9.1. It remains fully unclear what it is.
>

Good catch. The diversification nonce was a need for QUIC crypto, and it
won't be required for TLS or for the general format. I'll remove mentions
of it from the document.


> I change my code to negotiate version "Q036", and am trying to
> figure out which frame types I am supposed to include in the initial
> packet. The various subsections of Section 4.2 list some (and say that
> some are required), but there is not enough information here for me to
> determine which to include and what the initial values should be.
>

So, this initial packet will be a CHLO, which is a crypto handshake packet.
Basically, this is a Regular QUIC Packet, with the crypto handshake bits
being sent as a STREAM frame on stream ID 1. It's clear to me now that
having a detailed description of a handshake packet at this point in the
text is probably helpful. If the reorganization that I suggested earlier
makes sense, this may be around where I'd introduce the Regular Packet and
the STREAM frame format.

So I'm cheating, and am looking at a wireshark dump of the initial
> packet that the Chromium quic_client sends to my server. I notice
> there is a "Message Authentication Hash" following the public header
> that is not mentioned at all in the ID. There is some brief text about
> "AEAD data" in Section 9.3, but nothing that would let me understand
> how that hash is computed. A quick glance at draft-thomson-quic-tls
> doesn't resolve this either. Googling around, I find that older
> versions of the spec on Google Docs have a short paragraph on "Initial
> Packet Null Encryption" that talks about using a FNV1a=C2=AD128 hash, but
> even the current version on Google Docs doesn't have that paragraph
> anymore. There's clearly some content missing here.
>

Right -- this is the hash that is used by the AEAD algo used by QUIC's
current packet crypto. This may change with TLS, and may be a different
hash, but the location of the hash is right after the public header. This
isn't documtned in the spec as you note, so I'll add some text about this
along with the revision of the Packet Formats.

And that's where I'm currently at, after a few hours of coding. Hope
> this is useful.
>

Yes, extremely useful. Thanks again for your effort and feedback -- I'll
try to incorporate as much as I can before the next rev.

- jana


> Lars
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

--001a113f9a622ba5c5053f591ceb
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Lars,<div><br></div><div>Thanks again for your implemen=
tation effort and detailed feedback, and apologies for taking a while to re=
spond. I&#39;m happy to iterate, and I expect that my turnaround will be mu=
ch faster (unlike this time :-))</div><div><br></div><div class=3D"gmail_ex=
tra"><div class=3D"gmail_quote">On Fri, Sep 23, 2016 at 2:47 AM, Eggert, La=
rs <span dir=3D"ltr">&lt;<a href=3D"mailto:lars@netapp.com" target=3D"_blan=
k">lars@netapp.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204)=
;padding-left:1ex">Hi,<br>
<br>
I&#39;ve started to do a little QUIC implementation exercise, to see if<br>
the current set of documents is sufficient to allow independent<br>
implementation. As can be expected, they&#39;re not, but I wanted to post<b=
r>
some feedback based on what I encountered at this very, very, VERY<br>
early phase, so the documents can be improved.<br></blockquote><div><br></d=
iv><div>Thank you! This is what we&#39;re hoping for as well. Thank you for=
 being a willing early implementer.</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">
I&#39;m purposefully avoiding to rely at documentation other than the IDs.<=
br>
I won&#39;t look at the Chromium code, and I&#39;m only looking at the spec=
s<br>
on Google Docs or wireshark outputs in order to more accurately<br>
describe what&#39;s currently missing in the IDs. However, I run my<br>
implementation against the Chromium quic_server and quic_client<br>
implementations, in order to see if they interoperate.<br></blockquote><div=
><br></div><div>I anticipate some trouble in getting it to work with quic_c=
lient and quic_server if you&#39;re just building from the spec, since thes=
e binaries use the <a href=3D"https://docs.google.com/document/d/1g5nIXAIkN=
_Y-7XJW5K45IblHd_L2f5LTaDUDwvZ5L6g/edit">QUIC crypto handshake</a>, and the=
 spec currently does not specify one.=C2=A0</div><div>=C2=A0</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px =
solid rgb(204,204,204);padding-left:1ex">
(I may open source this pure C implementation at some point; at the<br>
moment, it&#39;s too embarrassing to do so.)<br>
<br>
I began by reading draft-hamilton-quic-transport-<wbr>protocol-00, in order=
<br>
to implement the initial handshake. The ID is reasonably long and<br>
complete-looking. Sections 1, 2 and 3 are reasonable in terms of<br>
content.<br>
<br>
Section 4 (Connection Establishment) is difficult to make sense of<br>
without first reading Section 9 (Packet Types and Formats), because<br>
Section 4 refers to header fields, flags and other concepts that are<br>
only defined in Section 9. The content seems arbitrarily split between<br>
those sections, e.g., Section 4.1 attempts to specify version<br>
negotiation, but Section 9.2.1 describes the version negotiation<br>
packet format. One needs to constantly flip back and forth, which is<br>
annoying. I&#39;d recommend reordering the content here.<br></blockquote><d=
iv><br></div><div>I wasn&#39;t sure how to organize this content, so it&#39=
;s currently mechanisms in early sections and wire formats later. From your=
 experience it seems like it may be better organized if the the wire format=
 of the various packets and frames were alongside the mechanisms. Does that=
 sound like a good rule of thumb? Given the size of the draft, that may be =
better, so I&#39;m happy to do it.</div><div><br></div><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(2=
04,204,204);padding-left:1ex">
One thing that threw me was that Section 9.1 states &quot;all QUIC packets<=
br>
on the wire begin with a public header&quot; and then has a diagram of what=
<br>
that header looks like, only to immediately follow it in Section 9.2<br>
with special packets that don&#39;t fully include that public header.<br>
That&#39;s really confusing. Is there a way to describe the public header<b=
r>
that would include the format of the special packets? Is there a way<br>
to make those special packets use the public header? They&#39;re so rarely<=
br>
sent that deviating to save a few bytes doesn&#39;t seem to be a smart<br>
tradeoff.<br></blockquote><div><br></div><div>I think there may be a way to=
 separate the common parts out. Specifically, the flags and the connection =
ID are really the &quot;common header&quot;, but the reason for having thes=
e three packet types is because they differ in the parts that are not encry=
pted (The parts that are encrypted should be specified, and I&#39;ll add th=
at.) Perhaps one way to describe them would be to say:</div><div>- Common H=
eader =3D Flags + (optional) Connection ID</div><div>- Regular Packets =3D =
Common Header + (optional) Version + Packet Number + Frames</div><div>- Ver=
sion Negotiation Packet =3D Common Header (Version Negotiation bit set) + V=
ersion List</div><div>- Public Reset Packet =3D Common Header (Reset bit se=
t) + Optional tag-values</div><div><br></div><div>I realized that we don&#3=
9;t describe the format for the optional tag-values in the spec, but they&#=
39;re potentially useful to help authenticate the Reset. That&#39;s entirel=
y optional at the moment, but really the Public Reset packet is identified =
by the bit in the flags and the connection is identified by the Connection =
ID.</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"=
>
Also, the document makes no mention of what endianness the protocol<br>
fields are supposed to be encoded in.<br></blockquote><div><br></div><div>G=
ood point -- I&#39;ve added a sentence to the Packet Types and Formats sayi=
ng &quot;All values are little-endian unless otherwise noted.&quot; This se=
ntence will likely move as the doc gets restructured, but that is the first=
 place a packet format is currently being described.</div><div><br></div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-le=
ft:1px solid rgb(204,204,204);padding-left:1ex">
The ID mentions QUIC version &quot;Q025&quot;, so my code attempts to negot=
iate<br>
this version with the Chromium quic_server, but that offers &quot;Q036&quot=
; and<br>
indicates support for &quot;Q035Q034Q033Q032&quot; as well. The negotiation=
<br>
response from the server does not conform to what&#39;s in the ID, since<br=
>
it has the &quot;diversification nonce&quot; flag set (0x04), which is not<=
br>
specified in Section 9.2.1. It also seems to use only 12 bytes of the<br>
fixed 32 bytes that the nonce should occupy to indicate support for<br>
other versions, according to Section 9.1.<br>
<br>
Also, that &quot;diversification nonce&quot; is not referred to anywhere el=
se in<br>
the ID other than Section 9.1. It remains fully unclear what it is.<br></bl=
ockquote><div><br></div><div>Good catch. The diversification nonce was a ne=
ed for QUIC crypto, and it won&#39;t be required for TLS or for the general=
 format. I&#39;ll remove mentions of it from the document.</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 change my code to negotiate version &quot;Q036&quot;, and am trying to<br=
>
figure out which frame types I am supposed to include in the initial<br>
packet. The various subsections of Section 4.2 list some (and say that<br>
some are required), but there is not enough information here for me to<br>
determine which to include and what the initial values should be.<br></bloc=
kquote><div><br></div><div>So, this initial packet will be a CHLO, which is=
 a crypto handshake packet. Basically, this is a Regular QUIC Packet, with =
the crypto handshake bits being sent as a STREAM frame on stream ID 1. It&#=
39;s clear to me now that having a detailed description of a handshake pack=
et at this point in the text is probably helpful. If the reorganization tha=
t I suggested earlier makes sense, this may be around where I&#39;d introdu=
ce the Regular Packet and the STREAM frame format.</div><div><br></div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left=
:1px solid rgb(204,204,204);padding-left:1ex">
So I&#39;m cheating, and am looking at a wireshark dump of the initial<br>
packet that the Chromium quic_client sends to my server. I notice<br>
there is a &quot;Message Authentication Hash&quot; following the public hea=
der<br>
that is not mentioned at all in the ID. There is some brief text about<br>
&quot;AEAD data&quot; in Section 9.3, but nothing that would let me underst=
and<br>
how that hash is computed. A quick glance at draft-thomson-quic-tls<br>
doesn&#39;t resolve this either. Googling around, I find that older<br>
versions of the spec on Google Docs have a short paragraph on &quot;Initial=
<br>
Packet Null Encryption&quot; that talks about using a FNV1a=C2=AD128 hash, =
but<br>
even the current version on Google Docs doesn&#39;t have that paragraph<br>
anymore. There&#39;s clearly some content missing here.<br></blockquote><di=
v><br></div><div>Right -- this is the hash that is used by the AEAD algo us=
ed by QUIC&#39;s current packet crypto. This may change with TLS, and may b=
e a different hash, but the location of the hash is right after the public =
header. This isn&#39;t documtned in the spec as you note, so I&#39;ll add s=
ome text about this along with the revision of the Packet Formats.</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">
And that&#39;s where I&#39;m currently at, after a few hours of coding. Hop=
e<br>
this is useful.<br></blockquote><div><br></div><div>Yes, extremely useful. =
Thanks again for your effort and feedback -- I&#39;ll try to incorporate as=
 much as I can before the next rev.</div><div><br></div><div>- jana</div><d=
iv>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0p=
x 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
Lars<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
</blockquote></div><br></div></div>

--001a113f9a622ba5c5053f591ceb--


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From: "Eggert, Lars" <lars@netapp.com>
To: Jana Iyengar <jri@google.com>
Thread-Topic: [QUIC] Early implementation feedback on draft-hamilton-quic-transport-protocol-00
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From nobody Fri Oct 21 05:05:08 2016
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From: "Eggert, Lars" <lars@netapp.com>
To: "quic@ietf.org" <quic@ietf.org>
Thread-Topic: Multiple frames of the same type per packet?
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Subject: [QUIC] Multiple frames of the same type per packet?
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Hi,

Section 10 says:

> 10.  Frame Types and Formats
>=20
>    A single Regular Packet MAY contain multiple frames and multiple
>    frame types.

However, it remains unclear if a regular packet is permitted to contain mul=
tiple frames of the same type. I guess this should be permitted for STREAM =
frames. But is there any reason to permit this for any of the other frame t=
ypes?

Thanks,
Lars=


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Cc: "quic@ietf.org" <quic@ietf.org>
Subject: Re: [QUIC] Multiple frames of the same type per packet?
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There are good reasons to have multiple stream and stream control frames. No=
t much point in having multiple ACK or similar frames, but we have to weight=
 the cost. Forbidding it means more code. The natural implementation is to p=
rocess a queue of frames as they arrive in one or multiple packets. To forbi=
d it you need to add code, and that can be a source of bugs.

Now, there is another issue. Frames can be received and processed out of ord=
er. We need to analyze for each frame what the consequences are.

-- Christian Huitema=20

> On Oct 21, 2016, at 5:04 AM, Eggert, Lars <lars@netapp.com> wrote:
>=20
> Hi,
>=20
> Section 10 says:
>=20
>> 10.  Frame Types and Formats
>>=20
>>   A single Regular Packet MAY contain multiple frames and multiple
>>   frame types.
>=20
> However, it remains unclear if a regular packet is permitted to contain mu=
ltiple frames of the same type. I guess this should be permitted for STREAM f=
rames. But is there any reason to permit this for any of the other frame typ=
es?
>=20
> Thanks,
> Lars
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic


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Subject: Re: [QUIC] Multiple frames of the same type per packet?
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--089e013d1448c51e6e053f5fd1a3
Content-Type: text/plain; charset=UTF-8

I think for any of the stream-specific frame types it makes sense to allow
multiples in one packet. For example if an endpoint wants to reset N
streams, they should be able to send a single packet with as many
RST_STREAM frames packed in as possible, rather than sending N packets each
with a single frame. The same argument applies to WINDOW_UPDATE, BLOCKED,
and of course STREAM frames.

On Fri, Oct 21, 2016 at 8:04 AM, Eggert, Lars <lars@netapp.com> wrote:

> Hi,
>
> Section 10 says:
>
> > 10.  Frame Types and Formats
> >
> >    A single Regular Packet MAY contain multiple frames and multiple
> >    frame types.
>
> However, it remains unclear if a regular packet is permitted to contain
> multiple frames of the same type. I guess this should be permitted for
> STREAM frames. But is there any reason to permit this for any of the other
> frame types?
>
> Thanks,
> Lars
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

--089e013d1448c51e6e053f5fd1a3
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<div dir=3D"ltr">I think for any of the stream-specific frame types it make=
s sense to allow multiples in one packet. For example if an endpoint wants =
to reset N streams, they should be able to send a single packet with as man=
y RST_STREAM frames packed in as possible, rather than sending N packets ea=
ch with a single frame. The same argument applies to WINDOW_UPDATE, BLOCKED=
, and of course STREAM frames.</div><div class=3D"gmail_extra"><br><div cla=
ss=3D"gmail_quote">On Fri, Oct 21, 2016 at 8:04 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>
Section 10 says:<br>
<br>
&gt; 10.=C2=A0 Frame Types and Formats<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 A single Regular Packet MAY contain multiple frames and m=
ultiple<br>
&gt;=C2=A0 =C2=A0 frame types.<br>
<br>
However, it remains unclear if a regular packet is permitted to contain mul=
tiple frames of the same type. I guess this should be permitted for STREAM =
frames. But is there any reason to permit this for any of the other frame t=
ypes?<br>
<br>
Thanks,<br>
Lars<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
</blockquote></div><br></div>

--089e013d1448c51e6e053f5fd1a3--


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Subject: Re: [QUIC] Multiple frames of the same type per packet?
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There are good reasons to have multiple stream and stream control frames. No=
t much point in having multiple ACK or similar frames, but we have to weight=
 the cost. Forbidding it means more code. The natural implementation is to p=
rocess a queue of frames as they arrive in one or multiple packets. To forbi=
d it you need to add code, and that can be a source of bugs.

Now, there is another issue. Frames can be received and processed out of ord=
er. We need to analyze for each frame what the consequences are.

-- Christian Huitema=20

> On Oct 21, 2016, at 5:04 AM, Eggert, Lars <lars@netapp.com> wrote:
>=20
> Hi,
>=20
> Section 10 says:
>=20
>> 10.  Frame Types and Formats
>>=20
>>   A single Regular Packet MAY contain multiple frames and multiple
>>   frame types.
>=20
> However, it remains unclear if a regular packet is permitted to contain mu=
ltiple frames of the same type. I guess this should be permitted for STREAM f=
rames. But is there any reason to permit this for any of the other frame typ=
es?
>=20
> Thanks,
> Lars
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic


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From: Ian Swett <ianswett@google.com>
Date: Fri, 21 Oct 2016 09:44:43 -0400
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Subject: Re: [QUIC] Multiple frames of the same type per packet?
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--94eb2c086e6cdac24e053f603ff5
Content-Type: text/plain; charset=UTF-8

Multiple reset stream, window update, or blocked frames would be valid as
well.  For other frame types, I believe it would not.

It's likely worth clarifying this?

On Fri, Oct 21, 2016 at 8:04 AM, Eggert, Lars <lars@netapp.com> wrote:

> Hi,
>
> Section 10 says:
>
> > 10.  Frame Types and Formats
> >
> >    A single Regular Packet MAY contain multiple frames and multiple
> >    frame types.
>
> However, it remains unclear if a regular packet is permitted to contain
> multiple frames of the same type. I guess this should be permitted for
> STREAM frames. But is there any reason to permit this for any of the other
> frame types?
>
> Thanks,
> Lars
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

--94eb2c086e6cdac24e053f603ff5
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<div dir=3D"ltr">Multiple reset stream, window update, or blocked frames wo=
uld be valid as well.=C2=A0 For other frame types, I believe it would not.<=
div><br><div>It&#39;s likely worth clarifying this?</div></div></div><div c=
lass=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Oct 21, 2016 at=
 8:04 AM, Eggert, Lars <span dir=3D"ltr">&lt;<a href=3D"mailto:lars@netapp.=
com" target=3D"_blank">lars@netapp.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">Hi,<br>
<br>
Section 10 says:<br>
<br>
&gt; 10.=C2=A0 Frame Types and Formats<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 A single Regular Packet MAY contain multiple frames and m=
ultiple<br>
&gt;=C2=A0 =C2=A0 frame types.<br>
<br>
However, it remains unclear if a regular packet is permitted to contain mul=
tiple frames of the same type. I guess this should be permitted for STREAM =
frames. But is there any reason to permit this for any of the other frame t=
ypes?<br>
<br>
Thanks,<br>
Lars<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
</blockquote></div><br></div>

--94eb2c086e6cdac24e053f603ff5--


From nobody Fri Oct 21 06:52:43 2016
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Subject: Re: [QUIC] Multiple frames of the same type per packet?
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Christian's point that forbidding certain combinations requires checking
and appropriate reaction and thus needs code and increases complexity is
well taken.

If multiple frames of the same type pertaining, e.g., to the same stream
or connection, are included, it may make sense to clearly specify that
must be handled in the order they appear in the packet and that state
updates are performed on each such frame so that the next one only acts
upon the outcome.  This would lead to frame being treated the same no
matter if they are in the same packet or not.  (But I am not sure if
this works for all the QUIC protocol machinery.)

Jörg

On 21.10.16 15:44, Ian Swett wrote:
> Multiple reset stream, window update, or blocked frames would be valid
> as well.  For other frame types, I believe it would not.
>
> It's likely worth clarifying this?
>
> On Fri, Oct 21, 2016 at 8:04 AM, Eggert, Lars <lars@netapp.com
> <mailto:lars@netapp.com>> wrote:
>
>     Hi,
>
>     Section 10 says:
>
>     > 10.  Frame Types and Formats
>     >
>     >    A single Regular Packet MAY contain multiple frames and multiple
>     >    frame types.
>
>     However, it remains unclear if a regular packet is permitted to
>     contain multiple frames of the same type. I guess this should be
>     permitted for STREAM frames. But is there any reason to permit this
>     for any of the other frame types?
>
>     Thanks,
>     Lars
>     _______________________________________________
>     QUIC mailing list
>     QUIC@ietf.org <mailto:QUIC@ietf.org>
>     https://www.ietf.org/mailman/listinfo/quic
>     <https://www.ietf.org/mailman/listinfo/quic>
>
>
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>


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From: Jana Iyengar <jri@google.com>
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Subject: [QUIC] Drafts structure and split
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--001a114aacbc9640c4053f64d5bd
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Hello folks,

There's been some really useful feedback on some of the input drafts from
folks on the list, and I'd like to update them before the drafts deadline,
so the changes are part of the wg discussions in Seoul. I also wondered if
folks had any specific comments on draft structure and split at this stage,
so the authors of the input drafts could get those in before the
conversation about adoption.

Thanks for all the feedback so far,
- jana

--001a114aacbc9640c4053f64d5bd
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<div dir=3D"ltr">Hello folks,<br><br>There&#39;s been some really useful fe=
edback on some of the input drafts from folks on the list, and I&#39;d like=
 to update them before the drafts deadline, so the changes are part of the =
wg discussions in Seoul. I also wondered if folks had any specific comments=
 on draft structure and split at this stage, so the authors of the input dr=
afts could get those in before the conversation about adoption.<div><br></d=
iv><div>Thanks for all the feedback so far,</div><div>- jana</div></div>

--001a114aacbc9640c4053f64d5bd--


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From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Hi,

I been wondering about this. Sometimes, i feel that the loss detection 
and reliability belongs to the protocol document (at least the basic 
loss detection and retransmision) and leave the congestion control in a 
separated document. I mean, it seems that loss detection and 
retransmission is too closely related to the basic protocol.

But i dont really have a strong opinion.
Regards, marcelo



El 21/10/16 a las 21:13, Jana Iyengar escribió:
> Hello folks,
>
> There's been some really useful feedback on some of the input drafts 
> from folks on the list, and I'd like to update them before the drafts 
> deadline, so the changes are part of the wg discussions in Seoul. I 
> also wondered if folks had any specific comments on draft structure 
> and split at this stage, so the authors of the input drafts could get 
> those in before the conversation about adoption.
>
> Thanks for all the feedback so far,
> - jana
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic



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There is a natural layering of stream controls on top of a packet transmissi=
on layer, which itself depends on security negotiations performed on stream 1=
.

The packet transmission layer would specify packet transmission, ack and err=
or detection, and packet level flow control and connection control. In my mi=
nd, it should not specify retransmission, since that is stream specific, but=
 it should notify the stream level when a stream frame has been lost, or ack=
nowledged.

Congestion control should be pluggable.

-- Christian Huitema=20

> On Oct 21, 2016, at 1:25 PM, marcelo bagnulo braun <marcelo@it.uc3m.es> wr=
ote:
>=20
> Hi,
>=20
> I been wondering about this. Sometimes, i feel that the loss detection and=
 reliability belongs to the protocol document (at least the basic loss detec=
tion and retransmision) and leave the congestion control in a separated docu=
ment. I mean, it seems that loss detection and retransmission is too closely=
 related to the basic protocol.
>=20
> But i dont really have a strong opinion.
> Regards, marcelo
>=20
>=20
>=20
>> El 21/10/16 a las 21:13, Jana Iyengar escribi=C3=B3:
>> Hello folks,
>>=20
>> There's been some really useful feedback on some of the input drafts from=
 folks on the list, and I'd like to update them before the drafts deadline, s=
o the changes are part of the wg discussions in Seoul. I also wondered if fo=
lks had any specific comments on draft structure and split at this stage, so=
 the authors of the input drafts could get those in before the conversation a=
bout adoption.
>>=20
>> Thanks for all the feedback so far,
>> - jana
>>=20
>>=20
>> _______________________________________________
>> QUIC mailing list
>> QUIC@ietf.org
>> https://www.ietf.org/mailman/listinfo/quic
>=20
>=20
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic


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From: Jana Iyengar <jri@google.com>
Date: Fri, 21 Oct 2016 14:53:32 -0700
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To: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Subject: Re: [QUIC] Drafts structure and split
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I think perhaps what the document should really be called is not "loss
recovery", but "loss detection". Pretty much everything in the doc is
expected to be about how to detect losses, and not so much about
retransmitting. What gets retransmitted and/or rebundled in actually
specified currently in the main protocol spec. Since loss detection is
intertwined with the currently specified congestion controllers for TCP, it
seems natural to combine them in a single doc.

There were two reasons for moving the loss detection and CC out into a
separate doc: so the main spec could be smaller, and so the loss detection
mechanisms could be independently evolved (later) as updates to the loss
recovery spec. Much like 793 and 2581 (though admittedly separate evolution
was not the concern then.) It might also help with reviews later on, since,
for example, folks in tcpm ought to be able to read, review, and understand
the similarities with and differences from TCP in terms of loss detection
without having to closely follow the rest of the QUIC spec.


On Fri, Oct 21, 2016 at 1:25 PM, marcelo bagnulo braun <marcelo@it.uc3m.es>
wrote:

> Hi,
>
> I been wondering about this. Sometimes, i feel that the loss detection an=
d
> reliability belongs to the protocol document (at least the basic loss
> detection and retransmision) and leave the congestion control in a
> separated document. I mean, it seems that loss detection and retransmissi=
on
> is too closely related to the basic protocol.
>
> But i dont really have a strong opinion.
> Regards, marcelo
>
>
>
> El 21/10/16 a las 21:13, Jana Iyengar escribi=C3=B3:
>
>> Hello folks,
>>
>> There's been some really useful feedback on some of the input drafts fro=
m
>> folks on the list, and I'd like to update them before the drafts deadlin=
e,
>> so the changes are part of the wg discussions in Seoul. I also wondered =
if
>> folks had any specific comments on draft structure and split at this sta=
ge,
>> so the authors of the input drafts could get those in before the
>> conversation about adoption.
>>
>> Thanks for all the feedback so far,
>> - jana
>>
>>
>> _______________________________________________
>> QUIC mailing list
>> QUIC@ietf.org
>> https://www.ietf.org/mailman/listinfo/quic
>>
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

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<div dir=3D"ltr"><div class=3D"gmail_extra">I think perhaps what the docume=
nt should really be called is not &quot;loss recovery&quot;, but &quot;loss=
 detection&quot;. Pretty much everything in the doc is expected to be about=
 how to detect losses, and not so much about retransmitting. What gets retr=
ansmitted and/or rebundled in actually specified currently in the main prot=
ocol spec. Since loss detection is intertwined with the currently specified=
 congestion controllers for TCP, it seems natural to combine them in a sing=
le doc.<br></div><div><br></div>There were two reasons for moving the loss =
detection and CC out into a separate doc: so the main spec could be smaller=
, and so the loss detection mechanisms could be independently evolved (late=
r) as updates to the loss recovery spec. Much like 793 and 2581 (though adm=
ittedly separate evolution was not the concern then.) It might also help wi=
th reviews later on, since, for example, folks in tcpm ought to be able to =
read, review, and understand the similarities with and differences from TCP=
 in terms of loss detection without having to closely follow the rest of th=
e QUIC spec.<div><br></div><div><br></div><div><div class=3D"gmail_extra"><=
div class=3D"gmail_quote">On Fri, Oct 21, 2016 at 1:25 PM, marcelo bagnulo =
braun <span dir=3D"ltr">&lt;<a href=3D"mailto:marcelo@it.uc3m.es" target=3D=
"_blank">marcelo@it.uc3m.es</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204=
,204,204);padding-left:1ex">Hi,<br>
<br>
I been wondering about this. Sometimes, i feel that the loss detection and =
reliability belongs to the protocol document (at least the basic loss detec=
tion and retransmision) and leave the congestion control in a separated doc=
ument. I mean, it seems that loss detection and retransmission is too close=
ly related to the basic protocol.<br>
<br>
But i dont really have a strong opinion.<br>
Regards, marcelo<br>
<br>
<br>
<br>
El 21/10/16 a las 21:13, Jana Iyengar escribi=C3=B3:<br>
<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><div class=3D"gmail-=
h5">
Hello folks,<br>
<br>
There&#39;s been some really useful feedback on some of the input drafts fr=
om folks on the list, and I&#39;d like to update them before the drafts dea=
dline, so the changes are part of the wg discussions in Seoul. I also wonde=
red if folks had any specific comments on draft structure and split at this=
 stage, so the authors of the input drafts could get those in before the co=
nversation about adoption.<br>
<br>
Thanks for all the feedback so far,<br>
- jana<br>
<br>
<br></div></div><span class=3D"gmail-">
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org" target=3D"_blank">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/quic</a><br>
</span></blockquote><div class=3D"gmail-HOEnZb"><div class=3D"gmail-h5">
<br>
<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org" target=3D"_blank">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/quic</a><br>
</div></div></blockquote></div><br></div></div></div>

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From: Jana Iyengar <jri@google.com>
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Christian,

I definitely like your layering in terms of an intellectual separation.

Tell me if you think differently, but I think the documents roughly reflect
some of the separation you suggest, but we've tried to keep the number of
docs from becoming too large (we already have 4). The stream abstraction
and the general protocol mechanisms for packet transmission are described
in the core protocol spec. Since streams and connection signaling may
require reliability, retransmissions and rebundling logic is described in
this doc.

The loss detection and congestion control spec describes how to use various
signals to detect losses, and (will describe) how to modulate the
congestion controller based on losses and received acks. This makes both
the congestion controller and the loss detection pluggable. To be clear,
even if these were to become a part of the core protocol document, I would
have a strong recommendation in there for implementers to keep their
congestion controller and loss detection machinery separate and pluggable.
I think keeping the docs separate helps for reviews and later updates,
i.e., "document pluggability."

- jana

On Fri, Oct 21, 2016 at 1:35 PM, Christian Huitema <huitema@huitema.net>
wrote:

> There is a natural layering of stream controls on top of a packet
> transmission layer, which itself depends on security negotiations perform=
ed
> on stream 1.
>
> The packet transmission layer would specify packet transmission, ack and
> error detection, and packet level flow control and connection control. In
> my mind, it should not specify retransmission, since that is stream
> specific, but it should notify the stream level when a stream frame has
> been lost, or acknowledged.
>
> Congestion control should be pluggable.
>
> -- Christian Huitema
>
> > On Oct 21, 2016, at 1:25 PM, marcelo bagnulo braun <marcelo@it.uc3m.es>
> wrote:
> >
> > Hi,
> >
> > I been wondering about this. Sometimes, i feel that the loss detection
> and reliability belongs to the protocol document (at least the basic loss
> detection and retransmision) and leave the congestion control in a
> separated document. I mean, it seems that loss detection and retransmissi=
on
> is too closely related to the basic protocol.
> >
> > But i dont really have a strong opinion.
> > Regards, marcelo
> >
> >
> >
> >> El 21/10/16 a las 21:13, Jana Iyengar escribi=C3=B3:
> >> Hello folks,
> >>
> >> There's been some really useful feedback on some of the input drafts
> from folks on the list, and I'd like to update them before the drafts
> deadline, so the changes are part of the wg discussions in Seoul. I also
> wondered if folks had any specific comments on draft structure and split =
at
> this stage, so the authors of the input drafts could get those in before
> the conversation about adoption.
> >>
> >> Thanks for all the feedback so far,
> >> - jana
> >>
> >>
> >> _______________________________________________
> >> QUIC mailing list
> >> QUIC@ietf.org
> >> https://www.ietf.org/mailman/listinfo/quic
> >
> >
> > _______________________________________________
> > QUIC mailing list
> > QUIC@ietf.org
> > https://www.ietf.org/mailman/listinfo/quic
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

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<div dir=3D"ltr"><div>Christian,</div><div><br></div>I definitely like your=
 layering in terms of an intellectual separation.<div><br></div><div>Tell m=
e if you think differently, but I think the documents roughly reflect some =
of the separation you suggest, but we&#39;ve tried to keep the number of do=
cs from becoming too large (we already have 4). The stream abstraction and =
the general protocol mechanisms for packet transmission are described in th=
e core protocol spec. Since streams and connection signaling may require re=
liability, retransmissions and rebundling logic is described in this doc.</=
div><div><br></div><div>The loss detection and congestion control spec desc=
ribes how to use various signals to detect losses, and (will describe) how =
to modulate the congestion controller based on losses and received acks. Th=
is makes both the congestion controller and the loss detection pluggable. T=
o be clear, even if these were to become a part of the core protocol docume=
nt, I would have a strong recommendation in there for implementers to keep =
their congestion controller and loss detection machinery separate and plugg=
able. I think keeping the docs separate helps for reviews and later updates=
, i.e., &quot;document pluggability.&quot;<br><div><div><br></div><div>- ja=
na</div><div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On F=
ri, Oct 21, 2016 at 1:35 PM, Christian Huitema <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:huitema@huitema.net" target=3D"_blank">huitema@huitema.net</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Ther=
e is a natural layering of stream controls on top of a packet transmission =
layer, which itself depends on security negotiations performed on stream 1.=
<br>
<br>
The packet transmission layer would specify packet transmission, ack and er=
ror detection, and packet level flow control and connection control. In my =
mind, it should not specify retransmission, since that is stream specific, =
but it should notify the stream level when a stream frame has been lost, or=
 acknowledged.<br>
<br>
Congestion control should be pluggable.<br>
<span class=3D"gmail-HOEnZb"><font color=3D"#888888"><br>
-- Christian Huitema<br>
</font></span><div class=3D"gmail-HOEnZb"><div class=3D"gmail-h5"><br>
&gt; On Oct 21, 2016, at 1:25 PM, marcelo bagnulo braun &lt;<a href=3D"mail=
to:marcelo@it.uc3m.es">marcelo@it.uc3m.es</a>&gt; wrote:<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; I been wondering about this. Sometimes, i feel that the loss detection=
 and reliability belongs to the protocol document (at least the basic loss =
detection and retransmision) and leave the congestion control in a separate=
d document. I mean, it seems that loss detection and retransmission is too =
closely related to the basic protocol.<br>
&gt;<br>
&gt; But i dont really have a strong opinion.<br>
&gt; Regards, marcelo<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;&gt; El 21/10/16 a las 21:13, Jana Iyengar escribi=C3=B3:<br>
&gt;&gt; Hello folks,<br>
&gt;&gt;<br>
&gt;&gt; There&#39;s been some really useful feedback on some of the input =
drafts from folks on the list, and I&#39;d like to update them before the d=
rafts deadline, so the changes are part of the wg discussions in Seoul. I a=
lso wondered if folks had any specific comments on draft structure and spli=
t at this stage, so the authors of the input drafts could get those in befo=
re the conversation about adoption.<br>
&gt;&gt;<br>
&gt;&gt; Thanks for all the feedback so far,<br>
&gt;&gt; - jana<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; ______________________________<wbr>_________________<br>
&gt;&gt; QUIC mailing list<br>
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&gt;<br>
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<br>
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--001a1143ee3e1dc41c053f676110--


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Subject: Re: [QUIC] Drafts structure and split
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--001a1143ee3e821d78053f677fd6
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One thing I meant to but forgot to mention:

Tell me if you think differently, but I think the documents roughly reflect
> some of the separation you suggest, but we've tried to keep the number of
> docs from becoming too large (we already have 4). The stream abstraction
> and the general protocol mechanisms for packet transmission are described
> in the core protocol spec. Since streams and connection signaling may
> require reliability, retransmissions and rebundling logic is described in
> this doc.
>

(Since this doc has streams, it also has Flow Control.) I can see this doc
being separated into two parts, split at the interface between the stream
abstractions and the packet transport logic. I wonder how useful it would
be to split this into two docs. We could point out that there are these two
distinct parts to the transport (which we can do within the core protocol
spec.)

--001a1143ee3e821d78053f677fd6
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<div dir=3D"ltr">One thing I meant to but forgot to mention:<div><br><div c=
lass=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div dir=3D"ltr"><div>Tell me if you think differently, but I think the=
 documents roughly reflect some of the separation you suggest, but we&#39;v=
e tried to keep the number of docs from becoming too large (we already have=
 4). The stream abstraction and the general protocol mechanisms for packet =
transmission are described in the core protocol spec. Since streams and con=
nection signaling may require reliability, retransmissions and rebundling l=
ogic is described in this doc.</div></div></blockquote><div><br></div><div>=
(Since this doc has streams, it also has Flow Control.) I can see this doc =
being separated into two parts, split at the interface between the stream a=
bstractions and the packet transport logic. I wonder how useful it would be=
 to split this into two docs. We could point out that there are these two d=
istinct parts to the transport (which we can do within the core protocol sp=
ec.)</div><div>=C2=A0</div></div></div></div></div>

--001a1143ee3e821d78053f677fd6--


From nobody Fri Oct 21 16:23:33 2016
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From: "\"IETF Secretariat\"" <agenda@ietf.org>
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Subject: [QUIC] quic - Requested session has been scheduled for IETF 97
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Dear Cindy Morgan,

The session(s) that you have requested have been scheduled.
Below is the scheduled session information followed by
the original request. 

quic Session 1 (2:30:00)
    Tuesday, Morning Session I 0930-1200
    Room Name: Grand Ballroom 1 size: 300
    ---------------------------------------------
    


Request Information:


---------------------------------------------------------
Working Group Name: QUIC
Area Name: Transport Area
Session Requester: Cindy Morgan

Number of Sessions: 1
Length of Session(s):  2.5 Hours
Number of Attendees: 250
Conflicts to Avoid: 
 First Priority: irtfopen artarea ianaplan t2trg taps mptcp tcpinc tcpm iccrg tsvwg maprg dispatch httpbis tls saag tsvarea
 Second Priority: rtcweb webpush acme



Special Requests:
  If the schedule is tight, this could go in a 2-hour slot.
---------------------------------------------------------


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From: "Eggert, Lars" <lars@netapp.com>
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Thread-Topic: [QUIC] Drafts structure and split
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Cc: Mark Nottingham <mnot@mnot.net>, "quic@ietf.org" <quic@ietf.org>, Spencer Dawkins <spencerdawkins.ietf@gmail.com>
Subject: Re: [QUIC] Drafts structure and split
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On 2016-10-21, at 21:13, Jana Iyengar <jri@google.com> wrote:
> I also wondered if folks had any specific comments on draft structure and=
 split at this stage, so the authors of the input drafts could get those in=
 before the conversation about adoption.

Without my chair hat on, I had been wondering if it makes sense to break ou=
t the overview parts from core-protocols and stick into a new "start readin=
g here" document.

Lars=


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From: Martin Thomson <martin.thomson@gmail.com>
Date: Sat, 22 Oct 2016 19:53:14 +1100
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Subject: Re: [QUIC] Multiple frames of the same type per packet?
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On 22 October 2016 at 00:52, Joerg Ott <ott@in.tum.de> wrote:
> Christian's point that forbidding certain combinations requires checking
> and appropriate reaction and thus needs code and increases complexity is
> well taken.

You don't have to insist on the receiver enforcing this, all you have
to do is permit them to.  Of course, the most is gained when every
implementation checks and reports it (by killing the connection or
something equally visible, of course), then you find that mistakes are
quickly identified.

I get Christian's point, but the amount of extra code is worth it.
Then you don't end up with systemic tolerance for bad implementations.


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Subject: Re: [QUIC] Drafts structure and split
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> On 22 Oct 2016, at 09:28, Eggert, Lars <lars@netapp.com> wrote:
>=20
> On 2016-10-21, at 21:13, Jana Iyengar <jri@google.com> wrote:
>> I also wondered if folks had any specific comments on draft structure =
and split at this stage, so the authors of the input drafts could get =
those in before the conversation about adoption.
>=20
> Without my chair hat on, I had been wondering if it makes sense to =
break out the overview parts from core-protocols and stick into a new =
"start reading here" document.

FWIW, I had the same thought.

Cheers,

Brian

>=20
> Lars
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic


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From: "Christian Huitema" <huitema@huitema.net>
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Subject: Re: [QUIC] Multiple frames of the same type per packet?
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On October 22, 2016 1:53 AM, Martin Thomson wrote:
> On 22 October 2016 at 00:52, Joerg Ott <ott@in.tum.de> wrote:
>> Christian's point that forbidding certain combinations requires checking
>> and appropriate reaction and thus needs code and increases complexity is
>> well taken.
>
> You don't have to insist on the receiver enforcing this, all you have
> to do is permit them to.  Of course, the most is gained when every
> implementation checks and reports it (by killing the connection or
> something equally visible, of course), then you find that mistakes are
> quickly identified.

Yes. Crowd immunity and all that. 

> I get Christian's point, but the amount of extra code is worth it.
> Then you don't end up with systemic tolerance for bad implementations.

I would be OK with something like "must not include the same frame type
related to the same stream more than once per packet, may drop connections
if encountering a packet that violates that rule." The point is that
compliance is easy, it is quite natural to only process a particular stream
once per packet, but enforcement is somewhat onerous. I would not want to
end up with some kind of compliance test in which the tester sends malformed
packets and flags the implementation as non-conforming because it did not
implement the verification code.

-- Christian Huitema




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Cc: marcelo bagnulo braun <marcelo@it.uc3m.es>, quic@ietf.org, Christian Huitema <huitema@huitema.net>
Subject: Re: [QUIC] Drafts structure and split
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My intuitive split would be:
* Overview document
* Packet layer document -- describes packet wire format, record-level
protection, ACK/STOP_WAITING, loss detection and congestion control.
* Stream layer document -- describes streams, flow control, and other
higher-level issues.
* Session establishment using TLS 1.3
* HTTP/2 mapping

I believe this split would be useful for the implementers given that this
layering seems to have worked out well in Google's current QUIC
implementation.

On Fri, Oct 21, 2016 at 6:23 PM, Jana Iyengar <jri@google.com> wrote:

> One thing I meant to but forgot to mention:
>
> Tell me if you think differently, but I think the documents roughly
>> reflect some of the separation you suggest, but we've tried to keep the
>> number of docs from becoming too large (we already have 4). The stream
>> abstraction and the general protocol mechanisms for packet transmission are
>> described in the core protocol spec. Since streams and connection signaling
>> may require reliability, retransmissions and rebundling logic is described
>> in this doc.
>>
>
> (Since this doc has streams, it also has Flow Control.) I can see this doc
> being separated into two parts, split at the interface between the stream
> abstractions and the packet transport logic. I wonder how useful it would
> be to split this into two docs. We could point out that there are these two
> distinct parts to the transport (which we can do within the core protocol
> spec.)
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>
>

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<div dir=3D"ltr">My intuitive split would be:<div>* Overview document</div>=
<div>* Packet layer document -- describes packet wire format, record-level =
protection, ACK/STOP_WAITING, loss detection and congestion control.</div><=
div>* Stream layer document -- describes streams, flow control, and other h=
igher-level issues.</div><div>* Session establishment using TLS 1.3</div><d=
iv>* HTTP/2 mapping</div><div><br></div><div>I believe this split would be =
useful for the implementers given that this layering seems to have worked o=
ut well in Google&#39;s current QUIC implementation.</div></div><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Oct 21, 2016 at 6:2=
3 PM, Jana Iyengar <span dir=3D"ltr">&lt;<a href=3D"mailto:jri@google.com" =
target=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;padd=
ing-left:1ex"><div dir=3D"ltr">One thing I meant to but forgot to mention:<=
div><br><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span class=
=3D""><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"><div>Tell me if you t=
hink differently, but I think the documents roughly reflect some of the sep=
aration you suggest, but we&#39;ve tried to keep the number of docs from be=
coming too large (we already have 4). The stream abstraction and the genera=
l protocol mechanisms for packet transmission are described in the core pro=
tocol spec. Since streams and connection signaling may require reliability,=
 retransmissions and rebundling logic is described in this doc.</div></div>=
</blockquote><div><br></div></span><div>(Since this doc has streams, it als=
o has Flow Control.) I can see this doc being separated into two parts, spl=
it at the interface between the stream abstractions and the packet transpor=
t logic. I wonder how useful it would be to split this into two docs. We co=
uld point out that there are these two distinct parts to the transport (whi=
ch we can do within the core protocol spec.)</div><div>=C2=A0</div></div></=
div></div></div>
<br>______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
<br></blockquote></div><br></div>

--94eb2c05a6fa7f4c4b053f7b373c--


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From: Kyle Rose <krose@krose.org>
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Subject: Re: [QUIC] Drafts structure and split
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On Fri, Oct 21, 2016 at 6:15 PM, Jana Iyengar <jri@google.com> wrote:

> Tell me if you think differently, but I think the documents roughly
> reflect some of the separation you suggest, but we've tried to keep the
> number of docs from becoming too large (we already have 4). The stream
> abstraction and the general protocol mechanisms for packet transmission are
> described in the core protocol spec. Since streams and connection signaling
> may require reliability, retransmissions and rebundling logic is described
> in this doc.
>

Even though FEC implementations are out of scope for the current charter,
it may make sense to structure the documents and the interfaces between
layers such that loss detection and ECN are fully independent of correction
and compensation mechanisms: these could be retransmits or FEC, but they
could also be AQM or notifications to the application to, for example,
reduce media quality.

Kyle

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Oct 21, 2016 at 6:15 PM, Jana Iyengar <span dir=3D"ltr">&lt;<a href=3D"=
mailto:jri@google.com" target=3D"_blank">jri@google.com</a>&gt;</span> wrot=
e:<br><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">Tell me if you think =
differently, but I think the documents roughly reflect some of the separati=
on you suggest, but we&#39;ve tried to keep the number of docs from becomin=
g too large (we already have 4). The stream abstraction and the general pro=
tocol mechanisms for packet transmission are described in the core protocol=
 spec. Since streams and connection signaling may require reliability, retr=
ansmissions and rebundling logic is described in this doc.</div></blockquot=
e><div><br></div><div>Even though FEC implementations are out of scope for =
the current charter, it may make sense to structure the documents and the i=
nterfaces between layers such that loss detection and ECN are fully indepen=
dent of correction and compensation mechanisms: these could be retransmits =
or FEC, but they could also be AQM or notifications to the application to, =
for example, reduce media quality.<br><br></div><div>Kyle<br></div></div></=
div></div>

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On 23 October 2016 at 05:06, Christian Huitema <huitema@huitema.net> wrote:
> I would be OK with something like "must not include the same frame type
> related to the same stream more than once per packet, may drop connections
> if encountering a packet that violates that rule." The point is that
> compliance is easy, it is quite natural to only process a particular stream
> once per packet, but enforcement is somewhat onerous. I would not want to
> end up with some kind of compliance test in which the tester sends malformed
> packets and flags the implementation as non-conforming because it did not
> implement the verification code.

Right, this is also my preference.  Having the escape valve is
particularly relevant in cases where your implementation is
constrained in some way.


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From: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
Date: Sun, 23 Oct 2016 21:50:57 -0500
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To: Kyle Rose <krose@krose.org>
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Cc: marcelo bagnulo braun <marcelo@it.uc3m.es>, Jana Iyengar <jri@google.com>, quic@ietf.org, Christian Huitema <huitema@huitema.net>
Subject: Re: [QUIC] Drafts structure and split
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--001a114e519625ebcf053f9376bf
Content-Type: text/plain; charset=UTF-8

Speaking as an AD who has been wrong before (references available on
request),

On Oct 23, 2016 13:21, "Kyle Rose" <krose@krose.org> wrote:
>
> On Fri, Oct 21, 2016 at 6:15 PM, Jana Iyengar <jri@google.com> wrote:
>>
>> Tell me if you think differently, but I think the documents roughly
reflect some of the separation you suggest, but we've tried to keep the
number of docs from becoming too large (we already have 4). The stream
abstraction and the general protocol mechanisms for packet transmission are
described in the core protocol spec. Since streams and connection signaling
may require reliability, retransmissions and rebundling logic is described
in this doc.
>
>
> Even though FEC implementations are out of scope for the current charter,
it may make sense to structure the documents and the interfaces between
layers such that loss detection and ECN are fully independent of correction
and compensation mechanisms: these could be retransmits or FEC, but they
could also be AQM or notifications to the application to, for example,
reduce media quality.
>
> Kyle
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

Whether or not the QUIC protocol ever does FEC, I'm thinking that the
division between loss detection and loss recovery would be helpful.

The observations that TCP ended up with the same split by accident that you
folks are considering doing intentionally was helpful for me.

Thanks,

Spencer

--001a114e519625ebcf053f9376bf
Content-Type: text/html; charset=UTF-8
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<p dir=3D"ltr">Speaking as an AD who has been wrong before (references avai=
lable on request), </p>
<p dir=3D"ltr">On Oct 23, 2016 13:21, &quot;Kyle Rose&quot; &lt;<a href=3D"=
mailto:krose@krose.org">krose@krose.org</a>&gt; wrote:<br>
&gt;<br>
&gt; On Fri, Oct 21, 2016 at 6:15 PM, Jana Iyengar &lt;<a href=3D"mailto:jr=
i@google.com">jri@google.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; Tell me if you think differently, but I think the documents roughl=
y reflect some of the separation you suggest, but we&#39;ve tried to keep t=
he number of docs from becoming too large (we already have 4). The stream a=
bstraction and the general protocol mechanisms for packet transmission are =
described in the core protocol spec. Since streams and connection signaling=
 may require reliability, retransmissions and rebundling logic is described=
 in this doc.<br>
&gt;<br>
&gt;<br>
&gt; Even though FEC implementations are out of scope for the current chart=
er, it may make sense to structure the documents and the interfaces between=
 layers such that loss detection and ECN are fully independent of correctio=
n and compensation mechanisms: these could be retransmits or FEC, but they =
could also be AQM or notifications to the application to, for example, redu=
ce media quality.<br>
&gt;<br>
&gt; Kyle<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; QUIC mailing list<br>
&gt; <a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/quic">https://www.iet=
f.org/mailman/listinfo/quic</a><br>
&gt;</p>
<p dir=3D"ltr">Whether or not the QUIC protocol ever does FEC, I&#39;m thin=
king that the division between loss detection and loss recovery would be he=
lpful.</p>
<p dir=3D"ltr">The observations that TCP ended up with the same split by ac=
cident that you folks are considering doing intentionally was helpful for m=
e.</p>
<p dir=3D"ltr">Thanks,</p>
<p dir=3D"ltr">Spencer</p>

--001a114e519625ebcf053f9376bf--


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Subject: Re: [QUIC] Drafts structure and split
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--f403045c68e2d01557053f93feb3
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First off, it is super exciting to me personally to see QUIC progressing so
well in the IETF!!!

The Design and Rationale of QUIC stressed several things:

a) The goal of QUIC was to reduce latency
b) There were numerous gains to be made by integrating elements of highly
evolved portions of traditional layered protocols.

As a result, I'm hopefully to see fewer, rather than more, documents. I'm
hoping to see tight and clear integration, rather than a layered protocol,
with layered documentation, falling victim to the many latency losses of
historical layers such HTTP over SPDY over TLS over TCP over IP.

That cross layer integration has already shown remarkable gains in QUIC.
For example, the integration of the crypto with awareness about
packetization boundaries was a great boon for both connection establishment
(low-latency handshakes; key change; etc.), as well as precluding of head
of line blocking (induced at the level of IP packet loss, but historically
surfacing in block encryption, if not TCP sequential streaming).

A second example is the clarity of packet loss information provided by QUIC
stream-content-rebundling (rather than packet re-transmission) has
apparently enhanced even traditional TCP (RENO?) congestion control results.

I'm very excited about the (finally published in ACM Queue) details of
Google's Congestion Control algorithm entitled BBR (Buffer Bloat
Reduction), which provides tremendous latency savings to TCP by minimizing
buffer size, and hence buffer latency.  I'm am filled with anticipation
about the integrated benefits with QUIC (and its primary goal, to reduce
latency), which should provide more precise transit-time information atop
BBRs incredible published results, and hence still better monitoring of
bottleneck buffer variations. With latency as a goal for QUIC, it is very
desirable to tightly integrate such latency-enhancing advances.

As a result, I'm hoping to see fewer, rather than more, independent
documents.  Each separate document will seemingly make it more difficult to
discuss variations, and potential tight integration in the QUIC protocol.

I do think it makes sense to separate discussion of packet loss detection
(which serves as one of the signals to any congestion control algorithm)
 from stream-content restransmission (which is surely dependent on the
application which QUIC serves), I don't see that such separation requires a
separate document.  I would however like some plugable API (for instance)
to support unreliable transport (probably outside of standardization of
existing practice).  That said, I would like to see direct (status quo)
discussion of how QUIC responds to retranmitted stream content, independent
of cause/motivation (loss recovery?  external IP packet replication due to
transmission media events?).

Bottom line: Hopefully fewer, well integrated documents.

I'm certainly willing to help edit and work on any of these documents.

Thanks,

Jim Roskind (who is only an IETF noob, and could be getting confused by
this discussion)

On Sun, Oct 23, 2016 at 11:21 AM, Kyle Rose <krose@krose.org> wrote:

> On Fri, Oct 21, 2016 at 6:15 PM, Jana Iyengar <jri@google.com> wrote:
>
>> Tell me if you think differently, but I think the documents roughly
>> reflect some of the separation you suggest, but we've tried to keep the
>> number of docs from becoming too large (we already have 4). The stream
>> abstraction and the general protocol mechanisms for packet transmission are
>> described in the core protocol spec. Since streams and connection signaling
>> may require reliability, retransmissions and rebundling logic is described
>> in this doc.
>>
>
> Even though FEC implementations are out of scope for the current charter,
> it may make sense to structure the documents and the interfaces between
> layers such that loss detection and ECN are fully independent of correction
> and compensation mechanisms: these could be retransmits or FEC, but they
> could also be AQM or notifications to the application to, for example,
> reduce media quality.
>
> Kyle
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>
>

--f403045c68e2d01557053f93feb3
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">First off, it is super exciting to me personally to see QU=
IC progressing so well in the IETF!!!<div><br></div><div>The Design and Rat=
ionale of QUIC stressed several things:<div><br></div><div>a) The goal of Q=
UIC was to reduce latency</div><div>b) There were numerous gains to be made=
 by integrating elements of highly evolved portions of traditional layered =
protocols.</div><div><br></div><div>As a result, I&#39;m hopefully to see f=
ewer, rather than more, documents. I&#39;m hoping to see tight and clear in=
tegration, rather than a layered protocol, with layered documentation, fall=
ing victim to the many latency losses of historical layers such HTTP over S=
PDY over TLS over TCP over IP.</div><div><br></div><div>That cross layer in=
tegration has already shown remarkable gains in QUIC. For example, the inte=
gration of the crypto with awareness about packetization boundaries was a g=
reat boon for both connection establishment (low-latency handshakes; key ch=
ange; etc.), as well as precluding of head of line blocking (induced at the=
 level of IP packet loss, but historically surfacing in block encryption, i=
f not TCP sequential streaming).</div><div><br></div><div>A second example =
is the clarity of packet loss information provided by QUIC stream-content-r=
ebundling (rather than packet re-transmission) has apparently enhanced even=
 traditional TCP (RENO?) congestion control results.</div><div><br></div><d=
iv>I&#39;m very excited about the (finally published in ACM Queue) details =
of Google&#39;s Congestion Control algorithm entitled BBR (Buffer Bloat Red=
uction), which provides tremendous latency savings to TCP by minimizing buf=
fer size, and hence buffer latency.=C2=A0 I&#39;m am filled with anticipati=
on about the integrated benefits with QUIC (and its primary goal, to reduce=
 latency), which should provide more precise transit-time information atop =
BBRs incredible published results, and hence still better monitoring of bot=
tleneck buffer variations. With latency as a goal for QUIC, it is very desi=
rable to tightly integrate such latency-enhancing advances.</div><div><br><=
/div><div>As a result, I&#39;m hoping to see fewer, rather than more, indep=
endent documents.=C2=A0 Each separate document will seemingly make it more =
difficult to discuss variations, and potential tight integration in the QUI=
C protocol.</div><div><br></div><div>I do think it makes sense to separate =
discussion of packet loss detection (which serves as one of the signals to =
any congestion control algorithm) =C2=A0from stream-content restransmission=
 (which is surely dependent on the application which QUIC serves), I don&#3=
9;t see that such separation requires a separate document.=C2=A0 I would ho=
wever like some plugable API (for instance) to support unreliable transport=
 (probably outside of standardization of existing practice).=C2=A0 That sai=
d, I would like to see direct (status quo) discussion of how QUIC responds =
to retranmitted stream content, independent of cause/motivation (loss recov=
ery? =C2=A0external IP packet replication due to transmission media events?=
).</div><div><br></div><div>Bottom line: Hopefully fewer, well integrated d=
ocuments.</div><div><br></div><div>I&#39;m certainly willing to help edit a=
nd work on any of these documents.</div><div><br></div><div>Thanks,</div><d=
iv><br></div><div>Jim Roskind (who is only an IETF noob, and could be getti=
ng confused by this discussion)</div></div></div><div class=3D"gmail_extra"=
><br><div class=3D"gmail_quote">On Sun, Oct 23, 2016 at 11:21 AM, Kyle Rose=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:krose@krose.org" target=3D"_blank"=
>krose@krose.org</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"><d=
iv dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span =
class=3D"">On Fri, Oct 21, 2016 at 6:15 PM, Jana Iyengar <span dir=3D"ltr">=
&lt;<a href=3D"mailto:jri@google.com" target=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">Tell m=
e if you think differently, but I think the documents roughly reflect some =
of the separation you suggest, but we&#39;ve tried to keep the number of do=
cs from becoming too large (we already have 4). The stream abstraction and =
the general protocol mechanisms for packet transmission are described in th=
e core protocol spec. Since streams and connection signaling may require re=
liability, retransmissions and rebundling logic is described in this doc.</=
div></blockquote><div><br></div></span><div>Even though FEC implementations=
 are out of scope for the current charter, it may make sense to structure t=
he documents and the interfaces between layers such that loss detection and=
 ECN are fully independent of correction and compensation mechanisms: these=
 could be retransmits or FEC, but they could also be AQM or notifications t=
o the application to, for example, reduce media quality.<span class=3D"HOEn=
Zb"><font color=3D"#888888"><br><br></font></span></div><span class=3D"HOEn=
Zb"><font color=3D"#888888"><div>Kyle<br></div></font></span></div></div></=
div>
<br>______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
<br></blockquote></div><br></div>

--f403045c68e2d01557053f93feb3--


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From: Kyle Rose <krose@krose.org>
Date: Mon, 24 Oct 2016 10:24:10 -0400
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Cc: marcelo bagnulo braun <marcelo@it.uc3m.es>, Jana Iyengar <jri@google.com>, quic@ietf.org, Christian Huitema <huitema@huitema.net>
Subject: Re: [QUIC] Drafts structure and split
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On Sun, Oct 23, 2016 at 11:29 PM, Jim Roskind <JimRoskind@gmail.com> wrote:

> As a result, I'm hopefully to see fewer, rather than more, documents. I'm
> hoping to see tight and clear integration, rather than a layered protocol,
> with layered documentation, falling victim to the many latency losses of
> historical layers such HTTP over SPDY over TLS over TCP over IP.
>

Layering definitely requires more effort to get right (for some definition
of "right": even the very wise cannot see all ends). Intentionally building
a monolith creates a different set of problems.


> For example, the integration of the crypto with awareness about
> packetization boundaries was a great boon for both connection establishment
> (low-latency handshakes; key change; etc.), as well as precluding of head
> of line blocking (induced at the level of IP packet loss, but historically
> surfacing in block encryption, if not TCP sequential streaming).
>

To get this right, one had to notice the tight interaction between the
crypto and packetization in advance. No matter how the documents are
structured, there are still two layers there and therefore still an
interface between them: the question is whether to make the interface
explicit and pluggable, or implicit and difficult to reuse.

I don't think the number of documents actually matters all that much, but
defining clear, bounded layers, along with appropriate information flow
between them to enable current design goals (e.g., low latency) and
foreseeable expanded functionality, does: putting effort in now will
hopefully land the inevitable ossification in places that are easier for
future protocol designers to deal with.

Kyle

--001a11470c3a40bfef053f9d250c
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On S=
un, Oct 23, 2016 at 11:29 PM, Jim Roskind <span dir=3D"ltr">&lt;<a href=3D"=
mailto:JimRoskind@gmail.com" target=3D"_blank">JimRoskind@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">As a resu=
lt, I&#39;m hopefully to see fewer, rather than more, documents. I&#39;m ho=
ping to see tight and clear integration, rather than a layered protocol, wi=
th layered documentation, falling victim to the many latency losses of hist=
orical layers such HTTP over SPDY over TLS over TCP over IP.</div></blockqu=
ote><div><br></div><div>Layering definitely requires more effort to get rig=
ht (for some definition of &quot;right&quot;: even the very wise cannot see=
 all ends). Intentionally building a monolith creates a different set of pr=
oblems.<br></div><div>=C2=A0</div><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></div><div>For example, the integration of the crypto wi=
th awareness about packetization boundaries was a great boon for both conne=
ction establishment (low-latency handshakes; key change; etc.), as well as =
precluding of head of line blocking (induced at the level of IP packet loss=
, but historically surfacing in block encryption, if not TCP sequential str=
eaming).</div></div></div></blockquote><div><br></div><div>To get this righ=
t, one had to notice the tight interaction between the crypto and packetiza=
tion in advance. No matter how the documents are structured, there are stil=
l two layers there and therefore still an interface between them: the quest=
ion is whether to make the interface explicit and pluggable, or implicit an=
d difficult to reuse.<br><br></div><div>I don&#39;t think the number  of do=
cuments actually matters all that much, but defining clear, bounded layers,=
 along with appropriate information flow between them to enable current des=
ign goals (e.g., low latency) and foreseeable expanded functionality, does:=
 putting effort in now will hopefully land the inevitable ossification in p=
laces that are easier for future protocol designers to deal with.<br><br></=
div><div>Kyle<br></div><br></div></div></div>

--001a11470c3a40bfef053f9d250c--


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From: Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>
Date: Mon, 24 Oct 2016 18:06:58 -0500
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Subject: Re: [QUIC] Drafts structure and split
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--001a1142cf12ee131b053fa4727e
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If I might offer an opinion, speaking as an individual ...

On Mon, Oct 24, 2016 at 9:24 AM, Kyle Rose <krose@krose.org> wrote:

> On Sun, Oct 23, 2016 at 11:29 PM, Jim Roskind <JimRoskind@gmail.com>
> wrote:
>
>> As a result, I'm hopefully to see fewer, rather than more, documents. I'm
>> hoping to see tight and clear integration, rather than a layered protocol,
>> with layered documentation, falling victim to the many latency losses of
>> historical layers such HTTP over SPDY over TLS over TCP over IP.
>>
>
> Layering definitely requires more effort to get right (for some definition
> of "right": even the very wise cannot see all ends). Intentionally building
> a monolith creates a different set of problems.
>
>
>> For example, the integration of the crypto with awareness about
>> packetization boundaries was a great boon for both connection establishment
>> (low-latency handshakes; key change; etc.), as well as precluding of head
>> of line blocking (induced at the level of IP packet loss, but historically
>> surfacing in block encryption, if not TCP sequential streaming).
>>
>
> To get this right, one had to notice the tight interaction between the
> crypto and packetization in advance. No matter how the documents are
> structured, there are still two layers there and therefore still an
> interface between them: the question is whether to make the interface
> explicit and pluggable, or implicit and difficult to reuse.
>
> I don't think the number of documents actually matters all that much, but
> defining clear, bounded layers, along with appropriate information flow
> between them to enable current design goals (e.g., low latency) and
> foreseeable expanded functionality, does: putting effort in now will
> hopefully land the inevitable ossification in places that are easier for
> future protocol designers to deal with.
>

If people are expecting a loss recovery mechanism beyond the current
version of BBR to ever appear in QUIC, what I would love is a plugable
*document structure*, where the loss recovery text that is being updated,
obsoleted, or will just be competing with existing mechanism(s) is clear,
without opening parts of the spec that aren't changing.

Thanks,

Spencer, speaking as an individual

--001a1142cf12ee131b053fa4727e
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">If I might offer an opinion, speaking as an individual ...=
<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, Oct 24, 2=
016 at 9:24 AM, Kyle Rose <span dir=3D"ltr">&lt;<a href=3D"mailto:krose@kro=
se.org" target=3D"_blank">krose@krose.org</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div cla=
ss=3D"gmail_quote"><span class=3D"">On Sun, Oct 23, 2016 at 11:29 PM, Jim R=
oskind <span dir=3D"ltr">&lt;<a href=3D"mailto:JimRoskind@gmail.com" target=
=3D"_blank">JimRoskind@gmail.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr">As a result, I&#39;m hopefully to see fewer,=
 rather than more, documents. I&#39;m hoping to see tight and clear integra=
tion, rather than a layered protocol, with layered documentation, falling v=
ictim to the many latency losses of historical layers such HTTP over SPDY o=
ver TLS over TCP over IP.</div></blockquote><div><br></div></span><div>Laye=
ring definitely requires more effort to get right (for some definition of &=
quot;right&quot;: even the very wise cannot see all ends). Intentionally bu=
ilding a monolith creates a different set of problems.<br></div><span class=
=3D""><div>=C2=A0</div><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></div><div>For example, the integration of the crypto with awareness =
about packetization boundaries was a great boon for both connection establi=
shment (low-latency handshakes; key change; etc.), as well as precluding of=
 head of line blocking (induced at the level of IP packet loss, but histori=
cally surfacing in block encryption, if not TCP sequential streaming).</div=
></div></div></blockquote><div><br></div></span><div>To get this right, one=
 had to notice the tight interaction between the crypto and packetization i=
n advance. No matter how the documents are structured, there are still two =
layers there and therefore still an interface between them: the question is=
 whether to make the interface explicit and pluggable, or implicit and diff=
icult to reuse.<br><br></div><div>I don&#39;t think the number  of document=
s actually matters all that much, but defining clear, bounded layers, along=
 with appropriate information flow between them to enable current design go=
als (e.g., low latency) and foreseeable expanded functionality, does: putti=
ng effort in now will hopefully land the inevitable ossification in places =
that are easier for future protocol designers to deal with.</div></div></di=
v></div></blockquote><div><br></div><div>If people are expecting a loss rec=
overy mechanism beyond the current version of BBR to ever appear in QUIC, w=
hat I would love is a plugable *document structure*, where the loss recover=
y text that is being updated, obsoleted, or will just be competing with exi=
sting mechanism(s) is clear, without opening parts of the spec that aren&#3=
9;t changing.=C2=A0</div><div><br></div><div>Thanks,</div><div><br></div><d=
iv>Spencer, speaking as an individual=C2=A0</div></div></div></div>

--001a1142cf12ee131b053fa4727e--


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From: Victor Vasiliev <vasilvv@google.com>
Date: Mon, 24 Oct 2016 19:15:59 -0400
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Subject: Re: [QUIC] draft-trammell-plus-statefulness
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On Thu, Oct 20, 2016 at 6:43 AM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> Lifting a choice piece.
>
> On 20 October 2016 at 21:19, Brian Trammell <ietf@trammell.ch> wrote:
> > Right, but aren't all the QUIC header bits also integrity-protected?
>
> Let's quickly assess these:
> 1. connection ID - routing critical, so I didn't see a need for
> protection (though we would roll it into the handshake if it was
> needed - most fall into this category)
> 2. version - rolled into the handshake
> 3. sequence number - forms the nonce (see TLS for analysis of this)
> 4. public reset etc...  - can't
>
> I realize now that I didn't write down the analysis of the first, so
> that could be wrong.


Cryptographically binding QUIC connection ID would be nice as an extra
protection
against possible confusion attacks.

I agree with you that for the most of the values in the public header
altering
them would result in either routing or decryption failure.  In the future,
we
might want to add values which we want everyone to read but want to
authenticate between endpoints too (like the connection close signal).  I
feel
like setting the AAD to the bytes of the public header is a fairly robust
construction that solves this problem for any of the potential changes we
make.
It should be easy to implement, and most of the implementations already
have to
do AAD in TLS 1.2, so presumably there should not be any extra crypto code
to
write here.

--001a114044603043b5053fa4939a
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Oct 20, 2016 at 6:43 AM, 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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:=
1ex">Lifting a choice piece.<br>
<span class=3D"gmail-"><br>
On 20 October 2016 at 21:19, Brian Trammell &lt;<a href=3D"mailto:ietf@tram=
mell.ch">ietf@trammell.ch</a>&gt; wrote:<br>
&gt; Right, but aren&#39;t all the QUIC header bits also integrity-protecte=
d?<br>
<br>
</span>Let&#39;s quickly assess these:<br>
1. connection ID - routing critical, so I didn&#39;t see a need for<br>
protection (though we would roll it into the handshake if it was<br>
needed - most fall into this category)<br>
2. version - rolled into the handshake<br>
3. sequence number - forms the nonce (see TLS for analysis of this)<br>
4. public reset etc...=C2=A0 - can&#39;t<br>
<br>
I realize now that I didn&#39;t write down the analysis of the first, so<br=
>
that could be wrong.</blockquote></div><br></div><div class=3D"gmail_extra"=
>Cryptographically binding QUIC connection ID would be nice as an extra pro=
tection</div><div class=3D"gmail_extra">against possible confusion attacks.=
</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra"><div =
class=3D"gmail_extra">I agree with you that for the most of the values in t=
he public header altering</div><div class=3D"gmail_extra">them would result=
 in either routing or decryption failure.=C2=A0 In the future, we</div><div=
 class=3D"gmail_extra">might want to add values which we want everyone to r=
ead but want to</div><div class=3D"gmail_extra">authenticate between endpoi=
nts too (like the connection close signal).=C2=A0 I feel</div><div class=3D=
"gmail_extra">like setting the AAD to the bytes of the public header is a f=
airly robust</div><div class=3D"gmail_extra">construction that solves this =
problem for any of the potential changes we make.</div><div class=3D"gmail_=
extra">It should be easy to implement, and most of the implementations alre=
ady have to</div><div class=3D"gmail_extra">do AAD in TLS 1.2, so presumabl=
y there should not be any extra crypto code to</div><div class=3D"gmail_ext=
ra">write here.</div><div><br></div></div></div>

--001a114044603043b5053fa4939a--


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Subject: Re: [QUIC] Drafts structure and split
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--001a1147eb280eb4fa053fa5a1ab
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Agreed, I think that structure makes sense, and as Kyle noted, makes
experimentation with FEC more straightforward.  One can easily imagine a
loss recovery approach where a given stream frame is never retransmitted,
but coded information is.

It's worth noting that BBR is only a change to the congestion controller,
and RACK is designed as a change the loss recovery.

On Mon, Oct 24, 2016 at 7:06 PM, Spencer Dawkins at IETF <
spencerdawkins.ietf@gmail.com> wrote:

> If I might offer an opinion, speaking as an individual ...
>
> On Mon, Oct 24, 2016 at 9:24 AM, Kyle Rose <krose@krose.org> wrote:
>
>> On Sun, Oct 23, 2016 at 11:29 PM, Jim Roskind <JimRoskind@gmail.com>
>> wrote:
>>
>>> As a result, I'm hopefully to see fewer, rather than more, documents.
>>> I'm hoping to see tight and clear integration, rather than a layered
>>> protocol, with layered documentation, falling victim to the many latency
>>> losses of historical layers such HTTP over SPDY over TLS over TCP over IP.
>>>
>>
>> Layering definitely requires more effort to get right (for some
>> definition of "right": even the very wise cannot see all ends).
>> Intentionally building a monolith creates a different set of problems.
>>
>>
>>> For example, the integration of the crypto with awareness about
>>> packetization boundaries was a great boon for both connection establishment
>>> (low-latency handshakes; key change; etc.), as well as precluding of head
>>> of line blocking (induced at the level of IP packet loss, but historically
>>> surfacing in block encryption, if not TCP sequential streaming).
>>>
>>
>> To get this right, one had to notice the tight interaction between the
>> crypto and packetization in advance. No matter how the documents are
>> structured, there are still two layers there and therefore still an
>> interface between them: the question is whether to make the interface
>> explicit and pluggable, or implicit and difficult to reuse.
>>
>> I don't think the number of documents actually matters all that much, but
>> defining clear, bounded layers, along with appropriate information flow
>> between them to enable current design goals (e.g., low latency) and
>> foreseeable expanded functionality, does: putting effort in now will
>> hopefully land the inevitable ossification in places that are easier for
>> future protocol designers to deal with.
>>
>
> If people are expecting a loss recovery mechanism beyond the current
> version of BBR to ever appear in QUIC, what I would love is a plugable
> *document structure*, where the loss recovery text that is being updated,
> obsoleted, or will just be competing with existing mechanism(s) is clear,
> without opening parts of the spec that aren't changing.
>
> Thanks,
>
> Spencer, speaking as an individual
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>
>

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<div dir=3D"ltr">Agreed, I think that structure makes sense, and as Kyle no=
ted, makes experimentation with FEC more straightforward.=C2=A0 One can eas=
ily imagine a loss recovery approach where a given stream frame is never re=
transmitted, but coded information is.<div><br></div><div>It&#39;s worth no=
ting that BBR is only a change to the congestion controller, and RACK is de=
signed as a change the loss recovery.</div></div><div class=3D"gmail_extra"=
><br><div class=3D"gmail_quote">On Mon, Oct 24, 2016 at 7:06 PM, Spencer Da=
wkins at IETF <span dir=3D"ltr">&lt;<a href=3D"mailto:spencerdawkins.ietf@g=
mail.com" target=3D"_blank">spencerdawkins.ietf@gmail.com</a>&gt;</span> wr=
ote:<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">If I might offer an=
 opinion, speaking as an individual ...<div class=3D"gmail_extra"><br><div =
class=3D"gmail_quote"><span class=3D"">On Mon, Oct 24, 2016 at 9:24 AM, Kyl=
e Rose <span dir=3D"ltr">&lt;<a href=3D"mailto:krose@krose.org" target=3D"_=
blank">krose@krose.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">=
<span>On Sun, Oct 23, 2016 at 11:29 PM, Jim Roskind <span dir=3D"ltr">&lt;<=
a href=3D"mailto:JimRoskind@gmail.com" target=3D"_blank">JimRoskind@gmail.c=
om</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"=
>As a result, I&#39;m hopefully to see fewer, rather than more, documents. =
I&#39;m hoping to see tight and clear integration, rather than a layered pr=
otocol, with layered documentation, falling victim to the many latency loss=
es of historical layers such HTTP over SPDY over TLS over TCP over IP.</div=
></blockquote><div><br></div></span><div>Layering definitely requires more =
effort to get right (for some definition of &quot;right&quot;: even the ver=
y wise cannot see all ends). Intentionally building a monolith creates a di=
fferent set of problems.<br></div><span><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr"><div><div></div><div>For example, the integr=
ation of the crypto with awareness about packetization boundaries was a gre=
at boon for both connection establishment (low-latency handshakes; key chan=
ge; etc.), as well as precluding of head of line blocking (induced at the l=
evel of IP packet loss, but historically surfacing in block encryption, if =
not TCP sequential streaming).</div></div></div></blockquote><div><br></div=
></span><div>To get this right, one had to notice the tight interaction bet=
ween the crypto and packetization in advance. No matter how the documents a=
re structured, there are still two layers there and therefore still an inte=
rface between them: the question is whether to make the interface explicit =
and pluggable, or implicit and difficult to reuse.<br><br></div><div>I don&=
#39;t think the number  of documents actually matters all that much, but de=
fining clear, bounded layers, along with appropriate information flow betwe=
en them to enable current design goals (e.g., low latency) and foreseeable =
expanded functionality, does: putting effort in now will hopefully land the=
 inevitable ossification in places that are easier for future protocol desi=
gners to deal with.</div></div></div></div></blockquote><div><br></div></sp=
an><div>If people are expecting a loss recovery mechanism beyond the curren=
t version of BBR to ever appear in QUIC, what I would love is a plugable *d=
ocument structure*, where the loss recovery text that is being updated, obs=
oleted, or will just be competing with existing mechanism(s) is clear, with=
out opening parts of the spec that aren&#39;t changing.=C2=A0</div><div><br=
></div><div>Thanks,</div><div><br></div><div>Spencer, speaking as an indivi=
dual=C2=A0</div></div></div></div>
<br>______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
<br></blockquote></div><br></div>

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From: Jana Iyengar <jri@google.com>
Date: Mon, 24 Oct 2016 20:08:37 -0700
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Cc: "Eggert, Lars" <lars@netapp.com>, Ian Swett <ianswett@google.com>, quic@ietf.org, Christian Huitema <huitema@huitema.net>, Joerg Ott <ott@in.tum.de>
Subject: Re: [QUIC] Multiple frames of the same type per packet?
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On Sun, Oct 23, 2016 at 4:55 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 23 October 2016 at 05:06, Christian Huitema <huitema@huitema.net>
> wrote:
> > I would be OK with something like "must not include the same frame type
> > related to the same stream more than once per packet, may drop
> connections
> > if encountering a packet that violates that rule." The point is that
> > compliance is easy, it is quite natural to only process a particular
> stream
> > once per packet, but enforcement is somewhat onerous. I would not want to
> > end up with some kind of compliance test in which the tester sends
> malformed
> > packets and flags the implementation as non-conforming because it did not
> > implement the verification code.
>
> Right, this is also my preference.  Having the escape valve is
> particularly relevant in cases where your implementation is
> constrained in some way.
>

Two things are not clear to me:
1. Why are these checks necessary? I don't think they're necessary for
correctness.
2. What's the incentive for a receiver to implement these checks? If it's
work, and if _not implementing_ these checks allows for interop with more
senders, why would a receiver implement these checks?

As Christian said earlier, a natural sender would queue frames and
packetize when deemed necessary. An inefficient sender may pack, for
example, multiple stream frames belonging to the same stream in the same
packet, but that's inefficient, not incorrect. A recommendation here can
point to good practice. It seems reasonable to recommend that a sender
SHOULD "not include the same frame type related to the same stream more
than once per packet." Saying that a receiver MAY drop a connection that
violates this agreement seems unnecessary.

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On S=
un, Oct 23, 2016 at 4:55 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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:=
1ex"><span class=3D"gmail-">On 23 October 2016 at 05:06, Christian Huitema =
&lt;<a href=3D"mailto:huitema@huitema.net">huitema@huitema.net</a>&gt; wrot=
e:<br>
&gt; I would be OK with something like &quot;must not include the same fram=
e type<br>
&gt; related to the same stream more than once per packet, may drop connect=
ions<br>
&gt; if encountering a packet that violates that rule.&quot; The point is t=
hat<br>
&gt; compliance is easy, it is quite natural to only process a particular s=
tream<br>
&gt; once per packet, but enforcement is somewhat onerous. I would not want=
 to<br>
&gt; end up with some kind of compliance test in which the tester sends mal=
formed<br>
&gt; packets and flags the implementation as non-conforming because it did =
not<br>
&gt; implement the verification code.<br>
<br>
</span>Right, this is also my preference.=C2=A0 Having the escape valve is<=
br>
particularly relevant in cases where your implementation is<br>
constrained in some way.<br>
</blockquote></div></div><div class=3D"gmail_extra"><br></div><div class=3D=
"gmail_extra">Two things are not clear to me:</div><div class=3D"gmail_extr=
a">1. Why are these checks necessary? I don&#39;t think they&#39;re necessa=
ry for correctness.</div><div class=3D"gmail_extra">2. What&#39;s the incen=
tive for a receiver to implement these checks? If it&#39;s work, and if _no=
t implementing_ these checks allows for interop with more senders, why woul=
d a receiver implement these checks?</div><div class=3D"gmail_extra"><br></=
div><div class=3D"gmail_extra">As Christian said earlier, a natural sender =
would queue frames and packetize when deemed necessary. An inefficient send=
er may pack, for example, multiple stream frames belonging to the same stre=
am in the same packet, but that&#39;s inefficient, not incorrect. A recomme=
ndation here can point to good practice. It seems reasonable to recommend t=
hat a sender SHOULD &quot;not include the same frame type related to the sa=
me stream more than once per packet.&quot; Saying that a receiver MAY drop =
a connection that violates this agreement seems unnecessary.</div></div>

--001a1143d158287546053fa7d3b6--


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From: "Eggert, Lars" <lars@netapp.com>
To: Jana Iyengar <jri@google.com>
Thread-Topic: [QUIC] Multiple frames of the same type per packet?
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Subject: Re: [QUIC] Multiple frames of the same type per packet?
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On 2016-10-25, at 5:08, Jana Iyengar <jri@google.com> wrote:
> Two things are not clear to me:
> 1. Why are these checks necessary? I don't think they're necessary for co=
rrectness.
> 2. What's the incentive for a receiver to implement these checks? If it's=
 work, and if _not implementing_ these checks allows for interop with more =
senders, why would a receiver implement these checks?
>=20
> As Christian said earlier, a natural sender would queue frames and packet=
ize when deemed necessary. An inefficient sender may pack, for example, mul=
tiple stream frames belonging to the same stream in the same packet, but th=
at's inefficient, not incorrect. A recommendation here can point to good pr=
actice. It seems reasonable to recommend that a sender SHOULD "not include =
the same frame type related to the same stream more than once per packet." =
Saying that a receiver MAY drop a connection that violates this agreement s=
eems unnecessary.

That makes sense, and an uppercase SHOULD may not even be needed, i.e., the=
 explanation can be phrased in terms of the inefficiency you describe above=
.

(I raised the question, because when I read the text, I was wondering if my=
 code should throw a warning when it received such packets.)

Lars=


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From: Jana Iyengar <jri@google.com>
Date: Mon, 24 Oct 2016 20:33:44 -0700
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To: Kyle Rose <krose@krose.org>
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Cc: marcelo bagnulo braun <marcelo@it.uc3m.es>, quic@ietf.org, Christian Huitema <huitema@huitema.net>
Subject: Re: [QUIC] Drafts structure and split
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On Sun, Oct 23, 2016 at 11:21 AM, Kyle Rose <krose@krose.org> wrote:

> On Fri, Oct 21, 2016 at 6:15 PM, Jana Iyengar <jri@google.com> wrote:
>
>> Tell me if you think differently, but I think the documents roughly
>> reflect some of the separation you suggest, but we've tried to keep the
>> number of docs from becoming too large (we already have 4). The stream
>> abstraction and the general protocol mechanisms for packet transmission are
>> described in the core protocol spec. Since streams and connection signaling
>> may require reliability, retransmissions and rebundling logic is described
>> in this doc.
>>
>
> Even though FEC implementations are out of scope for the current charter,
> it may make sense to structure the documents and the interfaces between
> layers such that loss detection and ECN are fully independent of correction
> and compensation mechanisms: these could be retransmits or FEC, but they
> could also be AQM or notifications to the application to, for example,
> reduce media quality.
>

Agreed, to detection vs. recovery. I think we could articulate this more
explicitly in the core spec, so I'll give it a shot. I'll also ask you to
remember this (and maybe suggest text if necessary) when reviewing the next
rev of the document :-)

On AQM/ECN, I expect that the signal will be passed to the congestion
controller, and should therefore form part of the interface between the
core protocol and the congestion controller. Any notificaitons to the
application can be interactions with the congestion controller. Does that
make sense?

--94eb2c122e78f8ddab053fa82cfa
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On S=
un, Oct 23, 2016 at 11:21 AM, Kyle Rose <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:krose@krose.org" target=3D"_blank">krose@krose.org</a>&gt;</span> wrot=
e:<br></div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div =
dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span cla=
ss=3D"">On Fri, Oct 21, 2016 at 6:15 PM, Jana Iyengar <span dir=3D"ltr">&lt=
;<a href=3D"mailto:jri@google.com" target=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">Tell me i=
f you think differently, but I think the documents roughly reflect some of =
the separation you suggest, but we&#39;ve tried to keep the number of docs =
from becoming too large (we already have 4). The stream abstraction and the=
 general protocol mechanisms for packet transmission are described in the c=
ore protocol spec. Since streams and connection signaling may require relia=
bility, retransmissions and rebundling logic is described in this doc.</div=
></blockquote><div><br></div></span><div>Even though FEC implementations ar=
e out of scope for the current charter, it may make sense to structure the =
documents and the interfaces between layers such that loss detection and EC=
N are fully independent of correction and compensation mechanisms: these co=
uld be retransmits or FEC, but they could also be AQM or notifications to t=
he application to, for example, reduce media quality.</div></div></div></di=
v></blockquote><div><br></div><div>Agreed, to detection vs. recovery. I thi=
nk we could articulate this more explicitly in the core spec, so I&#39;ll g=
ive it a shot. I&#39;ll also ask you to remember this (and maybe suggest te=
xt if necessary) when reviewing the next rev of the document :-)</div><div>=
<br></div><div>On AQM/ECN, I expect that the signal will be passed to the c=
ongestion controller, and should therefore form part of the interface betwe=
en the core protocol and the congestion controller. Any notificaitons to th=
e application can be interactions with the congestion controller. Does that=
 make sense?</div></div><br></div></div>

--94eb2c122e78f8ddab053fa82cfa--


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From: Jana Iyengar <jri@google.com>
Date: Mon, 24 Oct 2016 21:55:07 -0700
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To: "Eggert, Lars" <lars@netapp.com>
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Cc: Christian Huitema <huitema@huitema.net>, Ian Swett <ianswett@google.com>, "quic@ietf.org" <quic@ietf.org>, Martin Thomson <martin.thomson@gmail.com>, =?UTF-8?B?SsO2cmcgT3R0?= <ott@in.tum.de>
Subject: Re: [QUIC] Multiple frames of the same type per packet?
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On Mon, Oct 24, 2016 at 8:30 PM, Eggert, Lars <lars@netapp.com> wrote:

> On 2016-10-25, at 5:08, Jana Iyengar <jri@google.com> wrote:
> > Two things are not clear to me:
> > 1. Why are these checks necessary? I don't think they're necessary for
> correctness.
> > 2. What's the incentive for a receiver to implement these checks? If
> it's work, and if _not implementing_ these checks allows for interop with
> more senders, why would a receiver implement these checks?
> >
> > As Christian said earlier, a natural sender would queue frames and
> packetize when deemed necessary. An inefficient sender may pack, for
> example, multiple stream frames belonging to the same stream in the same
> packet, but that's inefficient, not incorrect. A recommendation here can
> point to good practice. It seems reasonable to recommend that a sender
> SHOULD "not include the same frame type related to the same stream more
> than once per packet." Saying that a receiver MAY drop a connection that
> violates this agreement seems unnecessary.
>
> That makes sense, and an uppercase SHOULD may not even be needed, i.e.,
> the explanation can be phrased in terms of the inefficiency you describe
> above.
>

That seems reasonable to me.

- jana


> (I raised the question, because when I read the text, I was wondering if
> my code should throw a warning when it received such packets.)
>
> Lars

--001a114e5bfc0361eb053fa9504f
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Oct 24, 2016 at 8:30 PM, Eggert, Lars <span dir=3D"ltr">&lt;<a =
href=3D"mailto:lars@netapp.com" target=3D"_blank">lars@netapp.com</a>&gt;</=
span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex"><span class=3D"">On 2016-10-=
25, at 5:08, Jana Iyengar &lt;<a href=3D"mailto:jri@google.com">jri@google.=
com</a>&gt; wrote:<br>
&gt; Two things are not clear to me:<br>
&gt; 1. Why are these checks necessary? I don&#39;t think they&#39;re neces=
sary for correctness.<br>
&gt; 2. What&#39;s the incentive for a receiver to implement these checks? =
If it&#39;s work, and if _not implementing_ these checks allows for interop=
 with more senders, why would a receiver implement these checks?<br>
&gt;<br>
&gt; As Christian said earlier, a natural sender would queue frames and pac=
ketize when deemed necessary. An inefficient sender may pack, for example, =
multiple stream frames belonging to the same stream in the same packet, but=
 that&#39;s inefficient, not incorrect. A recommendation here can point to =
good practice. It seems reasonable to recommend that a sender SHOULD &quot;=
not include the same frame type related to the same stream more than once p=
er packet.&quot; Saying that a receiver MAY drop a connection that violates=
 this agreement seems unnecessary.<br>
<br>
</span>That makes sense, and an uppercase SHOULD may not even be needed, i.=
e., the explanation can be phrased in terms of the inefficiency you describ=
e above.<br></blockquote><div><br></div><div>That seems reasonable to me.</=
div><div><br></div><div>- jana</div><div>=C2=A0</div><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">
(I raised the question, because when I read the text, I was wondering if my=
 code should throw a warning when it received such packets.)<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Lars</font></span></blockquote></div><br></div></div>

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From: Kyle Rose <krose@krose.org>
Date: Tue, 25 Oct 2016 15:07:50 -0400
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Subject: Re: [QUIC] Drafts structure and split
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On Mon, Oct 24, 2016 at 11:33 PM, Jana Iyengar <jri@google.com> wrote:

> Agreed, to detection vs. recovery. I think we could articulate this more
> explicitly in the core spec, so I'll give it a shot. I'll also ask you to
> remember this (and maybe suggest text if necessary) when reviewing the next
> rev of the document :-)
>

Will do.


> On AQM/ECN, I expect that the signal will be passed to the congestion
> controller, and should therefore form part of the interface between the
> core protocol and the congestion controller. Any notificaitons to the
> application can be interactions with the congestion controller. Does that
> make sense?
>

Some applications might care only about loss or preventing future loss
("does/will the client have enough information to reconstruct the stream,
or do I need to retransmit/send more ECC?"), while others might care about
congestion separately from loss (the previous question vs. "do I need to
reduce the bitrate?"), so having separate signaling for the two cases makes
a lot of sense.

Kyle

--94eb2c06f542955545053fb53984
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Oct 24, 2016 at 11:33 PM, Jana Iyengar <span dir=3D"ltr">&lt;<a href=3D=
"mailto:jri@google.com" target=3D"_blank">jri@google.com</a>&gt;</span> wro=
te:<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 class=3D"gmail_=
extra"><span class=3D""></span><span class=3D""></span><div>Agreed, to dete=
ction vs. recovery. I think we could articulate this more explicitly in the=
 core spec, so I&#39;ll give it a shot. I&#39;ll also ask you to remember t=
his (and maybe suggest text if necessary) when reviewing the next rev of th=
e document :-)</div></div></div></blockquote><div><br></div><div>Will do.<b=
r>=C2=A0<br></div><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 clas=
s=3D"gmail_extra"><div></div><div>On AQM/ECN, I expect that the signal will=
 be passed to the congestion controller, and should therefore form part of =
the interface between the core protocol and the congestion controller. Any =
notificaitons to the application can be interactions with the congestion co=
ntroller. Does that make sense?</div></div></div>
</blockquote></div><br></div><div class=3D"gmail_extra">Some applications m=
ight care only about loss or preventing future loss (&quot;does/will the cl=
ient have enough information to reconstruct the stream, or do I need to ret=
ransmit/send more ECC?&quot;), while others might care about congestion sep=
arately from loss (the previous question vs. &quot;do I need to reduce the =
bitrate?&quot;), so having separate signaling for the two cases makes a lot=
 of sense.<br><br></div><div class=3D"gmail_extra">Kyle<br></div><div class=
=3D"gmail_extra"><br></div></div>

--94eb2c06f542955545053fb53984--


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From: Jana Iyengar <jri@google.com>
Date: Tue, 25 Oct 2016 12:37:09 -0700
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--001a114410886aea79053fb5a28b
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On Tue, Oct 25, 2016 at 12:07 PM, Kyle Rose <krose@krose.org> wrote:

> On Mon, Oct 24, 2016 at 11:33 PM, Jana Iyengar <jri@google.com> wrote:
>
>> Agreed, to detection vs. recovery. I think we could articulate this more
>> explicitly in the core spec, so I'll give it a shot. I'll also ask you to
>> remember this (and maybe suggest text if necessary) when reviewing the next
>> rev of the document :-)
>>
>
> Will do.
>
>
>> On AQM/ECN, I expect that the signal will be passed to the congestion
>> controller, and should therefore form part of the interface between the
>> core protocol and the congestion controller. Any notificaitons to the
>> application can be interactions with the congestion controller. Does that
>> make sense?
>>
>
> Some applications might care only about loss or preventing future loss
> ("does/will the client have enough information to reconstruct the stream,
> or do I need to retransmit/send more ECC?"), while others might care about
> congestion separately from loss (the previous question vs. "do I need to
> reduce the bitrate?"), so having separate signaling for the two cases makes
> a lot of sense.
>

ECN signals indicate only congestion, not loss. My point was that
congestion signals - explicit ones via ECN or implicit ones via loss/delay
-- are all fed to the congestion controller. I agree that loss recovery is
an entirely different matter, and that's a signal that comes out of the
Loss Detection component. Make sense?

If you're pointing out that congestion control and loss detection are
separable, I agree. Perhaps this separation could be made explicit in the
Loss Detection and Congestion Control doc? Or would you want to separate
the two out? My earlier inclination was to keep fewer documents until they
become large enough to need separation, but I can see arguments for
document pluggability too.

- jana



>
> Kyle
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>
>

--001a114410886aea79053fb5a28b
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
ue, Oct 25, 2016 at 12:07 PM, Kyle Rose <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:krose@krose.org" target=3D"_blank">krose@krose.org</a>&gt;</span> wrot=
e:<br><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"><div class=3D"gmail_e=
xtra"><div class=3D"gmail_quote"><span class=3D"">On Mon, Oct 24, 2016 at 1=
1:33 PM, Jana Iyengar <span dir=3D"ltr">&lt;<a href=3D"mailto:jri@google.co=
m" target=3D"_blank">jri@google.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><span></span><=
span></span><div>Agreed, to detection vs. recovery. I think we could articu=
late this more explicitly in the core spec, so I&#39;ll give it a shot. I&#=
39;ll also ask you to remember this (and maybe suggest text if necessary) w=
hen reviewing the next rev of the document :-)</div></div></div></blockquot=
e><div><br></div></span><div>Will do.<br>=C2=A0<br></div><span 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"><div class=3D"gmail_extra"><=
div></div><div>On AQM/ECN, I expect that the signal will be passed to the c=
ongestion controller, and should therefore form part of the interface betwe=
en the core protocol and the congestion controller. Any notificaitons to th=
e application can be interactions with the congestion controller. Does that=
 make sense?</div></div></div>
</blockquote></span></div><br></div><div class=3D"gmail_extra">Some applica=
tions might care only about loss or preventing future loss (&quot;does/will=
 the client have enough information to reconstruct the stream, or do I need=
 to retransmit/send more ECC?&quot;), while others might care about congest=
ion separately from loss (the previous question vs. &quot;do I need to redu=
ce the bitrate?&quot;), so having separate signaling for the two cases make=
s a lot of sense.</div></div></blockquote><div><br></div><div>ECN signals i=
ndicate only congestion, not loss. My point was that congestion signals - e=
xplicit ones via ECN or implicit ones via loss/delay -- are all fed to the =
congestion controller. I agree that loss recovery is an entirely different =
matter, and that&#39;s a signal that comes out of the Loss Detection compon=
ent. Make sense?</div><div><br></div><div>If you&#39;re pointing out that c=
ongestion control and loss detection are separable, I agree. Perhaps this s=
eparation could be made explicit in the Loss Detection and Congestion Contr=
ol doc? Or would you want to separate the two out? My earlier inclination w=
as to keep fewer documents until they become large enough to need separatio=
n, but I can see arguments for document pluggability too.</div><div><br></d=
iv><div>- jana</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><span class=3D"HOEnZb">=
<font color=3D"#888888"><br></font></span></div><span class=3D"HOEnZb"><fon=
t color=3D"#888888"><div class=3D"gmail_extra">Kyle<br></div><div class=3D"=
gmail_extra"><br></div></font></span></div>
<br>______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
<br></blockquote></div><br></div></div>

--001a114410886aea79053fb5a28b--


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From: Kyle Rose <krose@krose.org>
Date: Tue, 25 Oct 2016 17:34:30 -0400
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To: Jana Iyengar <jri@google.com>
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Cc: marcelo bagnulo braun <marcelo@it.uc3m.es>, quic@ietf.org, Christian Huitema <huitema@huitema.net>
Subject: Re: [QUIC] Drafts structure and split
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On Tue, Oct 25, 2016 at 3:37 PM, Jana Iyengar <jri@google.com> wrote:

> ECN signals indicate only congestion, not loss. My point was that
> congestion signals - explicit ones via ECN or implicit ones via loss/delay
> -- are all fed to the congestion controller. I agree that loss recovery is
> an entirely different matter, and that's a signal that comes out of the
> Loss Detection component. Make sense?
>

Yep, I agree completely. I think maybe what I meant was more clearly stated
as "Applications may (will?) want to react differently to loss or
prospective loss from congestion, versus loss for other reasons (like poor
wifi signal)." We may currently have difficulty distinguishing the former
from the latter when ECN isn't a universal congestion signal, but having
the proper interface in place allowing applications to tailor behavior
based on improved future telemetry could be useful.


> If you're pointing out that congestion control and loss detection are
> separable, I agree. Perhaps this separation could be made explicit in the
> Loss Detection and Congestion Control doc? Or would you want to separate
> the two out? My earlier inclination was to keep fewer documents until they
> become large enough to need separation, but I can see arguments for
> document pluggability too.
>

I agree with Spencer's point about document pluggability: it's less
important how many documents the WG produces than that they use
encapsulation to make it easy to swap out or replace components later
without having to individually update sentences from 17 different sections.

Kyle

--001a11429ba22346d5053fb7463b
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">On Tue, Oct 25, 2016 at 3:37 PM, 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"><div class=3D"gmail_e=
xtra"><div class=3D"gmail_quote"><span></span>ECN signals indicate only con=
gestion, not loss. My point was that congestion signals - explicit ones via=
 ECN or implicit ones via loss/delay -- are all fed to the congestion contr=
oller. I agree that loss recovery is an entirely different matter, and that=
&#39;s a signal that comes out of the Loss Detection component. Make sense?=
</div></div></div></blockquote><div><br></div><div>Yep, I agree completely.=
 I think maybe what I meant was more clearly stated as &quot;Applications m=
ay (will?) want to react differently to loss or prospective loss from conge=
stion, versus loss for other reasons (like poor wifi signal).&quot; We may =
currently have difficulty distinguishing the former from the latter when EC=
N isn&#39;t a universal congestion signal, but having the proper interface =
in place allowing applications to tailor behavior based on improved future =
telemetry could be useful.<br>=C2=A0<br></div><blockquote class=3D"gmail_qu=
ote" 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 class=3D"gmail_quote"><d=
iv>If you&#39;re pointing out that congestion control and loss detection ar=
e separable, I agree. Perhaps this separation could be made explicit in the=
 Loss Detection and Congestion Control doc? Or would you want to separate t=
he two out? My earlier inclination was to keep fewer documents until they b=
ecome large enough to need separation, but I can see arguments for document=
 pluggability too.</div></div></div></div></blockquote><br></div><div class=
=3D"gmail_quote">I agree with Spencer&#39;s point about document pluggabili=
ty: it&#39;s less important how many documents the WG produces than that th=
ey use encapsulation to make it easy to swap out or replace components late=
r without having to individually update sentences from 17 different section=
s.<br><br></div><div class=3D"gmail_quote">Kyle<br><br></div></div></div>

--001a11429ba22346d5053fb7463b--


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From: Jana Iyengar <jri@google.com>
Date: Tue, 25 Oct 2016 18:18:40 -0700
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To: Kyle Rose <krose@krose.org>
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Subject: Re: [QUIC] Drafts structure and split
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--94eb2c122e78cb1951053fba6736
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On Tue, Oct 25, 2016 at 2:34 PM, Kyle Rose <krose@krose.org> wrote:

> On Tue, Oct 25, 2016 at 3:37 PM, Jana Iyengar <jri@google.com> wrote:
>
>> ECN signals indicate only congestion, not loss. My point was that
>> congestion signals - explicit ones via ECN or implicit ones via loss/delay
>> -- are all fed to the congestion controller. I agree that loss recovery is
>> an entirely different matter, and that's a signal that comes out of the
>> Loss Detection component. Make sense?
>>
>
> Yep, I agree completely. I think maybe what I meant was more clearly
> stated as "Applications may (will?) want to react differently to loss or
> prospective loss from congestion, versus loss for other reasons (like poor
> wifi signal)." We may currently have difficulty distinguishing the former
> from the latter when ECN isn't a universal congestion signal, but having
> the proper interface in place allowing applications to tailor behavior
> based on improved future telemetry could be useful.
>
>
>> If you're pointing out that congestion control and loss detection are
>> separable, I agree. Perhaps this separation could be made explicit in the
>> Loss Detection and Congestion Control doc? Or would you want to separate
>> the two out? My earlier inclination was to keep fewer documents until they
>> become large enough to need separation, but I can see arguments for
>> document pluggability too.
>>
>
> I agree with Spencer's point about document pluggability: it's less
> important how many documents the WG produces than that they use
> encapsulation to make it easy to swap out or replace components later
> without having to individually update sentences from 17 different sections.
>

Sounds good to me.

- jana


>
> Kyle
>
>

--94eb2c122e78cb1951053fba6736
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
ue, Oct 25, 2016 at 2:34 PM, Kyle Rose <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:krose@krose.org" target=3D"_blank">krose@krose.org</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"">On Tu=
e, Oct 25, 2016 at 3:37 PM, Jana Iyengar <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:jri@google.com" target=3D"_blank">jri@google.com</a>&gt;</span> wrote=
:<br></span><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span cla=
ss=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"><div class=3D"gmail=
_extra"><div class=3D"gmail_quote"><span></span>ECN signals indicate only c=
ongestion, not loss. My point was that congestion signals - explicit ones v=
ia ECN or implicit ones via loss/delay -- are all fed to the congestion con=
troller. I agree that loss recovery is an entirely different matter, and th=
at&#39;s a signal that comes out of the Loss Detection component. Make sens=
e?</div></div></div></blockquote><div><br></div></span><div>Yep, I agree co=
mpletely. I think maybe what I meant was more clearly stated as &quot;Appli=
cations may (will?) want to react differently to loss or prospective loss f=
rom congestion, versus loss for other reasons (like poor wifi signal).&quot=
; We may currently have difficulty distinguishing the former from the latte=
r when ECN isn&#39;t a universal congestion signal, but having the proper i=
nterface in place allowing applications to tailor behavior based on improve=
d future telemetry could be useful.<br>=C2=A0<br></div><span class=3D""><bl=
ockquote 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"><di=
v class=3D"gmail_quote"><div>If you&#39;re pointing out that congestion con=
trol and loss detection are separable, I agree. Perhaps this separation cou=
ld be made explicit in the Loss Detection and Congestion Control doc? Or wo=
uld you want to separate the two out? My earlier inclination was to keep fe=
wer documents until they become large enough to need separation, but I can =
see arguments for document pluggability too.</div></div></div></div></block=
quote><br></span></div><div class=3D"gmail_quote">I agree with Spencer&#39;=
s point about document pluggability: it&#39;s less important how many docum=
ents the WG produces than that they use encapsulation to make it easy to sw=
ap out or replace components later without having to individually update se=
ntences from 17 different sections.</div></div></div></blockquote><div><br>=
</div><div>Sounds good to me.</div><div><br></div><div>- jana</div><div>=C2=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmai=
l_extra"><div class=3D"gmail_quote"><span class=3D"HOEnZb"><font color=3D"#=
888888"><br></font></span></div><span class=3D"HOEnZb"><font color=3D"#8888=
88"><div class=3D"gmail_quote">Kyle<br><br></div></font></span></div></div>
</blockquote></div><br></div></div>

--94eb2c122e78cb1951053fba6736--


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From: Jim Roskind <JimRoskind@gmail.com>
Date: Tue, 25 Oct 2016 19:13:11 -0700
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Subject: Re: [QUIC] Drafts structure and split
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On Mon, Oct 24, 2016 at 5:31 PM, Ian Swett <ianswett@google.com> wrote:

> ...
> It's worth noting that BBR is only a change to the congestion controller,
> ....
>
>
>
BBR relies on getting estimates of transit time, so as to estimate
variations in buffer sizes.  As a relatively rough estimate, I believe BBR
in TCP relies on traditional-TCP RTT calculations, which includes the round
trip sum of transit times in both directions.  I expect that better one-way
transit time estimates can be harvested in QUIC, by changing the data
included in ACK frames to better illuminate transit times in each
directions (or more precisely, illuminate changes, from one packet to the
next, in one-way transit times).

Such modifications appear at first blush to be notably larger than just a
change to the congestion controller.

Similarly, I'd expect modifications to support ECN in QUIC to run much
deeper than a congestion controller.

Jim

--94eb2c07c916c086e0053fbb2a55
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Oct 24, 2016 at 5:31 PM, Ian Swett <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:ianswett@google.com" target=3D"_blank">ianswett@google.com</a>&gt;</spa=
n> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;b=
order-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>...</div>=
<div>It&#39;s worth noting that BBR is only a change to the congestion cont=
roller, ....</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_=
quote"><div><div class=3D"h5"><br></div></div></div></div></blockquote><div=
><br></div><div>BBR relies on getting estimates of transit time, so as to e=
stimate variations in buffer sizes.=C2=A0 As a relatively rough estimate, I=
 believe BBR in TCP relies on traditional-TCP RTT calculations, which inclu=
des the round trip sum of transit times in both directions.=C2=A0 I expect =
that better one-way transit time estimates can be harvested in QUIC, by cha=
nging the data included in ACK frames to better illuminate transit times in=
 each directions (or more precisely, illuminate changes, from one packet to=
 the next, in one-way transit times).</div><div><br></div><div>Such modific=
ations appear at first blush to be notably larger than just a change to the=
 congestion controller.</div><div><br></div><div>Similarly, I&#39;d expect =
modifications to support ECN in QUIC to run much deeper than a congestion c=
ontroller.</div><div><br></div><div>Jim=C2=A0</div></div></div></div>

--94eb2c07c916c086e0053fbb2a55--


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From: Ryan Hamilton <rch@google.com>
Date: Tue, 25 Oct 2016 21:09:28 -0700
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To: Jim Roskind <JimRoskind@gmail.com>
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Subject: Re: [QUIC] Drafts structure and split
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On Tue, Oct 25, 2016 at 7:13 PM, Jim Roskind <JimRoskind@gmail.com> wrote:

> On Mon, Oct 24, 2016 at 5:31 PM, Ian Swett <ianswett@google.com> wrote:
>
>> ...
>> It's worth noting that BBR is only a change to the congestion controller=
,
>> ....
>>
>>
>>
> BBR relies on getting estimates of transit time, so as to estimate
> variations in buffer sizes.  As a relatively rough estimate, I believe BB=
R
> in TCP relies on traditional-TCP RTT calculations, which includes the rou=
nd
> trip sum of transit times in both directions.  I expect that better one-w=
ay
> transit time estimates can be harvested in QUIC, by changing the data
> included in ACK frames to better illuminate transit times in each
> directions (or more precisely, illuminate changes, from one packet to the
> next, in one-way transit times).
>

=E2=80=8BCorrect me if I'm wrong, but QUIC ACK frames already have this
information. The contain the arrival time of each packet at the peer, which
provides the one-way transfer time.
=E2=80=8B


>  Such modifications appear at first blush to be notably larger than just =
a
> change to the congestion controller.
>

> Similarly, I'd expect modifications to support ECN in QUIC to run much
> deeper than a congestion controller.
>
> Jim
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>
>

--047d7b873864a1c4b5053fbcca4d
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:trebuche=
t ms,sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On Tue, Oct 25, 2016 at 7:13 PM, Jim Roskind <span dir=3D"ltr">&l=
t;<a href=3D"mailto:JimRoskind@gmail.com" target=3D"_blank" class=3D"cremed=
">JimRoskind@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:1=
ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">=
On Mon, Oct 24, 2016 at 5:31 PM, Ian Swett <span dir=3D"ltr">&lt;<a href=3D=
"mailto:ianswett@google.com" target=3D"_blank" class=3D"cremed">ianswett@go=
ogle.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><div>It&#39;s worth noting that BBR is only a change=
 to the congestion controller, ....</div></div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote"><div><div class=3D"m_-2646942042275758490h5">=
<br></div></div></div></div></blockquote><div><br></div><div>BBR relies on =
getting estimates of transit time, so as to estimate variations in buffer s=
izes.=C2=A0 As a relatively rough estimate, I believe BBR in TCP relies on =
traditional-TCP RTT calculations, which includes the round trip sum of tran=
sit times in both directions.=C2=A0 I expect that better one-way transit ti=
me estimates can be harvested in QUIC, by changing the data included in ACK=
 frames to better illuminate transit times in each directions (or more prec=
isely, illuminate changes, from one packet to the next, in one-way transit =
times).</div></div></div></div></blockquote><div><br></div><div><div class=
=3D"gmail_default" style=3D"font-family:&quot;trebuchet ms&quot;,sans-serif=
;display:inline">=E2=80=8BCorrect me if I&#39;m wrong, but QUIC ACK frames =
already have this information. The contain the arrival time of each packet =
at the peer, which provides the one-way transfer time.</div></div><div><div=
 class=3D"gmail_default" style=3D"font-family:&quot;trebuchet ms&quot;,sans=
-serif;display:inline">=E2=80=8B</div>=C2=A0</div><blockquote class=3D"gmai=
l_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 class=3D"gmail_quote=
"><div>=C2=A0Such modifications appear at first blush to be notably larger =
than just a change to the congestion controller.</div></div></div></div></b=
lockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gma=
il_extra"><div class=3D"gmail_quote"><div><br></div><div>Similarly, I&#39;d=
 expect modifications to support ECN in QUIC to run much deeper than a cong=
estion controller.</div><span class=3D"HOEnZb"><font color=3D"#888888"><div=
><br></div><div>Jim=C2=A0</div></font></span></div></div></div>
<br>______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org" class=3D"cremed">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank" class=3D"cremed">https://www.ietf.org/mailman/<wbr>listinf=
o/quic</a><br>
<br></blockquote></div><br></div></div>

--047d7b873864a1c4b5053fbcca4d--


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From: "Eggert, Lars" <lars@netapp.com>
To: "quic@ietf.org" <quic@ietf.org>
Thread-Topic: IETF-97 agenda planning
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Archived-At: <https://mailarchive.ietf.org/arch/msg/quic/xMlOZk4PeLxFPQJrWEMOo2dS_tc>
Subject: [QUIC] IETF-97 agenda planning
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Hi,

we are TENTATIVELY scheduled for Tuesday, November 15, 2016 in the 9:30-12:=
00 "Tuesday Morning session I" slot.	         =20

Our current agenda is evolving on https://github.com/quicwg/wg-materials/bl=
ob/master/ietf97/agenda.md.

If you would like to present during the session, please send an email to qu=
ic-chairs@ietf.org, include a few sentences outlining your talk and give us=
 an estimate for how much presentation + discussion time you think is neede=
d.

Lars

PS: ID authors, please note the upcoming ID submission deadline: 2016-07-08=
 (Friday): Internet Draft submission cut-off (for all drafts, including -00=
) by UTC 23:59, upload using IETF ID Submission Tool.=


From nobody Wed Oct 26 00:43:41 2016
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Cc: Lars Eggert <lars@netapp.com>
Subject: [QUIC] First Interim planning -- availability poll
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Everyone,

We'd like to get our first interim meeting nailed down soon. We have a host w=
ith with what we believe to be a suitable space in Tokyo, Japan on 24-26 Jan=
uary.

To make sure that it's suitable for a good portion of the WG, as well as to a=
ssist planning, please fill out this poll to indicate your availability:
  http://doodle.com/poll/ams8q39eecud4zih

In particular, if you'd like to actively participate but the cannot make tho=
se dates, please indicate that with "Cannot make it."

We'll have more details soon.

Cheers,

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



Sent from my iPhone=


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From: "Eggert, Lars" <lars@netapp.com>
To: "quic@ietf.org" <quic@ietf.org>
Thread-Topic: [QUIC] IETF-97 agenda planning
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Subject: Re: [QUIC] IETF-97 agenda planning
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On 2016-10-26, at 8:56, Eggert, Lars <lars@netapp.com> wrote:
> PS: ID authors, please note the upcoming ID submission deadline: 2016-07-=
08 (Friday): Internet Draft submission cut-off (for all drafts, including -=
00) by UTC 23:59, upload using IETF ID Submission Tool.

Eagle-eyed participants (or some that can actually read, rather than appare=
ntly your co-chair) have spotted that the date above is wrong and should be=
: 2016-10-31 (Monday): Internet Draft submission cut-off (for all drafts, i=
ncluding -00) by UTC 23:59, upload using IETF ID Submission Tool.

Sorry,
Lars=


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Ryan Hamilton <rch@google.com>, Jim Roskind <JimRoskind@gmail.com>
Thread-Topic: QUIC ACK frames and one-way transfer time (was RE: [QUIC] Drafts structure and split)
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Subject: [QUIC] QUIC ACK frames and one-way transfer time (was RE: Drafts structure and split)
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From: Ryan Hamilton <rch@google.com>
Date: Wed, 26 Oct 2016 06:46:51 -0700
Message-ID: <CAJ_4DfQZw3dOjutqxbtOcAut_iM4FVbYEh2EUNsE9XbPChHiWw@mail.gmail.com>
To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
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Cc: Ian Swett <ianswett@google.com>, Kyle Rose <krose@krose.org>, Christian Huitema <huitema@huitema.net>, Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>, Jana Iyengar <jri@google.com>, marcelo bagnulo braun <marcelo@it.uc3m.es>, Jim Roskind <JimRoskind@gmail.com>, "quic@ietf.org" <quic@ietf.org>
Subject: Re: [QUIC] QUIC ACK frames and one-way transfer time (was RE: Drafts structure and split)
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Yes, clock synchronization is an issue, but QUIC makes an attempt to deal
with it. The receipt "time" in the ACK frame is not an actual wall time,
but is instead a time delta since the connection is established. Assuming
that the clocks are moving forward at roughly the same rate, this means
that the relative skew in these deltas is the time it took for the CHLO to
reach the server, and the peers can account for this in subsequent
calculations.


On Wed, Oct 26, 2016 at 4:06 AM, Ingemar Johansson S <
ingemar.s.johansson@ericsson.com> wrote:

> Hi
>
>
>
> Ryan. You are right that the QUIC ACK frames already contains the
> necessary information to compute the one way delay. It is of course not
> possible to deduce the true one way delay unless the sender and receiver
> clocks are synchronized. What is easy to determine is the queue delay tha=
t
> can be used in a way similar to how LEDBAT does it, there are some pros a=
nd
> cons with that approach.
>
> https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/ describes
> a modification to Cubic that is based on this principle.
>
>
>
> /Ingemar
>
>
>
>
>
> *From:* Ryan Hamilton [mailto:rch@google.com]
> *Sent:* den 26 oktober 2016 06:09
> *To:* Jim Roskind <JimRoskind@gmail.com>
> *Cc:* Ian Swett <ianswett@google.com>; Kyle Rose <krose@krose.org>;
> Christian Huitema <huitema@huitema.net>; Spencer Dawkins at IETF <
> spencerdawkins.ietf@gmail.com>; marcelo bagnulo braun <marcelo@it.uc3m.es=
>;
> Jana Iyengar <jri@google.com>; quic@ietf.org
> *Subject:* Re: [QUIC] Drafts structure and split
>
>
>
>
>
>
>
> On Tue, Oct 25, 2016 at 7:13 PM, Jim Roskind <JimRoskind@gmail.com> wrote=
:
>
> On Mon, Oct 24, 2016 at 5:31 PM, Ian Swett <ianswett@google.com> wrote:
>
> ...
>
> It's worth noting that BBR is only a change to the congestion controller,
> ....
>
>
>
>
>
>
>
> BBR relies on getting estimates of transit time, so as to estimate
> variations in buffer sizes.  As a relatively rough estimate, I believe BB=
R
> in TCP relies on traditional-TCP RTT calculations, which includes the rou=
nd
> trip sum of transit times in both directions.  I expect that better one-w=
ay
> transit time estimates can be harvested in QUIC, by changing the data
> included in ACK frames to better illuminate transit times in each
> directions (or more precisely, illuminate changes, from one packet to the
> next, in one-way transit times).
>
>
>
> =E2=80=8BCorrect me if I'm wrong, but QUIC ACK frames already have this
> information. The contain the arrival time of each packet at the peer, whi=
ch
> provides the one-way transfer time.
>
> =E2=80=8B
>
>
>
>  Such modifications appear at first blush to be notably larger than just =
a
> change to the congestion controller.
>
>
>
> Similarly, I'd expect modifications to support ECN in QUIC to run much
> deeper than a congestion controller.
>
>
>
> Jim
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>
>

--047d7bb70e2e7cc845053fc4db16
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:trebuche=
t ms,sans-serif">Yes, clock synchronization is an issue, but QUIC makes an =
attempt to deal with it. The receipt &quot;time&quot; in the ACK frame is n=
ot an actual wall time, but is instead a time delta since the connection is=
 established. Assuming that the clocks are moving forward at roughly the sa=
me rate, this means that the relative skew in these deltas is the time it t=
ook for the CHLO to reach the server, and the peers can account for this in=
 subsequent calculations.</div><div class=3D"gmail_default" style=3D"font-f=
amily:trebuchet ms,sans-serif"><br></div></div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Wed, Oct 26, 2016 at 4:06 AM, Ingemar Joha=
nsson S <span dir=3D"ltr">&lt;<a href=3D"mailto:ingemar.s.johansson@ericsso=
n.com" target=3D"_blank">ingemar.s.johansson@ericsson.com</a>&gt;</span> wr=
ote:<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"blue" vlink=3D"purple">
<div class=3D"m_8778874064686280382WordSection1">
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,sans-serif">Hi<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,sans-serif"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,sans-serif">Ryan.
</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans=
-serif">You are right that the QUIC ACK frames already contains the necessa=
ry information to compute the one way delay. It is of course not possible t=
o deduce the true one way delay unless the sender
 and receiver clocks are synchronized. What is easy to determine is the que=
ue delay that can be used in a way similar to how LEDBAT does it, there are=
 some pros and cons with that approach.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif"><a href=3D"https://datatracker.ietf.org/doc/draft-j=
ohansson-cc-for-4g-5g/" target=3D"_blank">https://datatracker.ietf.org/<wbr=
>doc/draft-johansson-cc-for-4g-<wbr>5g/</a> describes a modification to Cub=
ic
 that is based on this principle.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif">/Ingemar<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif"><u></u>=C2=A0<u></u></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #e1e1e1 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"SV" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,sans-serif">From:</span></b><span lang=3D"SV" st=
yle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif"> Ryan H=
amilton [mailto:<a href=3D"mailto:rch@google.com" target=3D"_blank">rch@goo=
gle.com</a>]
<br>
<b>Sent:</b> den 26 oktober 2016 06:09<br>
<b>To:</b> Jim Roskind &lt;<a href=3D"mailto:JimRoskind@gmail.com" target=
=3D"_blank">JimRoskind@gmail.com</a>&gt;<br>
<b>Cc:</b> </span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri=
&quot;,sans-serif">Ian Swett &lt;<a href=3D"mailto:ianswett@google.com" tar=
get=3D"_blank">ianswett@google.com</a>&gt;; Kyle Rose &lt;<a href=3D"mailto=
:krose@krose.org" target=3D"_blank">krose@krose.org</a>&gt;; Christian Huit=
ema &lt;<a href=3D"mailto:huitema@huitema.net" target=3D"_blank">huitema@hu=
itema.net</a>&gt;; Spencer Dawkins at IETF &lt;<a href=3D"mailto:spencerdaw=
kins.ietf@gmail.com" target=3D"_blank">spencerdawkins.ietf@gmail.com</a><wb=
r>&gt;; marcelo bagnulo
 braun &lt;<a href=3D"mailto:marcelo@it.uc3m.es" target=3D"_blank">marcelo@=
it.uc3m.es</a>&gt;; Jana Iyengar &lt;<a href=3D"mailto:jri@google.com" targ=
et=3D"_blank">jri@google.com</a>&gt;; <a href=3D"mailto:quic@ietf.org" targ=
et=3D"_blank">quic@ietf.org</a><br>
<b>Subject:</b> Re: [QUIC] Drafts structure and split<u></u><u></u></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Trebuchet MS&quot;,=
sans-serif"><u></u>=C2=A0<u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Tue, Oct 25, 2016 at 7:13 PM, Jim Roskind &lt;<a =
href=3D"mailto:JimRoskind@gmail.com" target=3D"_blank">JimRoskind@gmail.com=
</a>&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<p class=3D"MsoNormal">On Mon, Oct 24, 2016 at 5:31 PM, Ian Swett &lt;<a hr=
ef=3D"mailto:ianswett@google.com" target=3D"_blank">ianswett@google.com</a>=
&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal">...<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">It&#39;s worth noting that BBR is only a change to t=
he congestion controller, ....<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">BBR relies on getting estimates of transit time, so =
as to estimate variations in buffer sizes.=C2=A0 As a relatively rough esti=
mate, I believe BBR in TCP relies on traditional-TCP RTT calculations, whic=
h includes the round trip sum of transit
 times in both directions.=C2=A0 I expect that better one-way transit time =
estimates can be harvested in QUIC, by changing the data included in ACK fr=
ames to better illuminate transit times in each directions (or more precise=
ly, illuminate changes, from one packet
 to the next, in one-way transit times).<u></u><u></u></p>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,sans-se=
rif">=E2=80=8B</span><span style=3D"font-family:&quot;Trebuchet MS&quot;,sa=
ns-serif">Correct me if I&#39;m wrong, but QUIC ACK frames already have thi=
s information. The contain the arrival time of each packet at the peer,
 which provides the one-way transfer time.<u></u><u></u></span></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,sans-se=
rif">=E2=80=8B</span><span style=3D"font-family:&quot;Trebuchet MS&quot;,sa=
ns-serif"><u></u><u></u></span></p>
</div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal">=C2=A0Such modifications appear at first blush to be=
 notably larger than just a change to the congestion controller.<u></u><u><=
/u></p>
</div>
</div>
</div>
</div>
</blockquote>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Similarly, I&#39;d expect modifications to support E=
CN in QUIC to run much deeper than a congestion controller.<u></u><u></u></=
p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888"><u></u>=C2=A0<u></u></=
span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Jim=C2=A0<u></u><u></u=
></span></p>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org" target=3D"_blank">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" target=3D"_blank">ht=
tps://www.ietf.org/mailman/<wbr>listinfo/quic</a><u></u><u></u></p>
</blockquote>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>
</div>
</div>

<br>______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
<br></blockquote></div><br></div>

--047d7bb70e2e7cc845053fc4db16--


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From: Ted Hardie <ted.ietf@gmail.com>
Date: Wed, 26 Oct 2016 06:52:08 -0700
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To: Mark Nottingham <mnot@mnot.net>
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Subject: Re: [QUIC] First Interim planning -- availability poll
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Howdy,

Two thoughts:  our first working group meeting will be in Asia and our
second in the U.S.--would you like a space in Europe for the interim?   For
remote attendees, will we use meetecho or something else?

best,

Ted

On Wed, Oct 26, 2016 at 12:43 AM, Mark Nottingham <mnot@mnot.net> wrote:

> Everyone,
>
> We'd like to get our first interim meeting nailed down soon. We have a
> host with with what we believe to be a suitable space in Tokyo, Japan on
> 24-26 January.
>
> To make sure that it's suitable for a good portion of the WG, as well as
> to assist planning, please fill out this poll to indicate your availability:
>   http://doodle.com/poll/ams8q39eecud4zih
>
> In particular, if you'd like to actively participate but the cannot make
> those dates, please indicate that with "Cannot make it."
>
> We'll have more details soon.
>
> Cheers,
>
> --
> Mark Nottingham   https://www.mnot.net/
>
>
>
> Sent from my iPhone
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

--001a114242ae2120ae053fc4f095
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<div dir=3D"ltr"><div><div><div>Howdy,<br><br></div>Two thoughts:=C2=A0 our=
 first working group meeting will be in Asia and our second in the U.S.--wo=
uld you like a space in Europe for the interim?=C2=A0=C2=A0 For remote atte=
ndees, will we use meetecho or something else?<br><br></div>best,<br></div>=
<div><br></div>Ted<br></div><div class=3D"gmail_extra"><br><div class=3D"gm=
ail_quote">On Wed, Oct 26, 2016 at 12:43 AM, Mark Nottingham <span dir=3D"l=
tr">&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">Everyone,<br>
<br>
We&#39;d like to get our first interim meeting nailed down soon. We have a =
host with with what we believe to be a suitable space in Tokyo, Japan on 24=
-26 January.<br>
<br>
To make sure that it&#39;s suitable for a good portion of the WG, as well a=
s to assist planning, please fill out this poll to indicate your availabili=
ty:<br>
=C2=A0 <a href=3D"http://doodle.com/poll/ams8q39eecud4zih" rel=3D"noreferre=
r" target=3D"_blank">http://doodle.com/poll/<wbr>ams8q39eecud4zih</a><br>
<br>
In particular, if you&#39;d like to actively participate but the cannot mak=
e those dates, please indicate that with &quot;Cannot make it.&quot;<br>
<br>
We&#39;ll have more details soon.<br>
<br>
Cheers,<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>
<br>
<br>
Sent from my iPhone<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
</blockquote></div><br></div>

--001a114242ae2120ae053fc4f095--


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Subject: Re: [QUIC] First Interim planning -- availability poll
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And another half-thought: the poll seems to provide no way to indicate =
availability/preference for in-person versus remote attendance. I'd =
think that would be a useful piece of input for the chairs to have.

Cheers,

Brian


> On 26 Oct 2016, at 15:52, Ted Hardie <ted.ietf@gmail.com> wrote:
>=20
> Howdy,
>=20
> Two thoughts:  our first working group meeting will be in Asia and our =
second in the U.S.--would you like a space in Europe for the interim?   =
For remote attendees, will we use meetecho or something else?
>=20
> best,
>=20
> Ted
>=20
> On Wed, Oct 26, 2016 at 12:43 AM, Mark Nottingham <mnot@mnot.net> =
wrote:
> Everyone,
>=20
> We'd like to get our first interim meeting nailed down soon. We have a =
host with with what we believe to be a suitable space in Tokyo, Japan on =
24-26 January.
>=20
> To make sure that it's suitable for a good portion of the WG, as well =
as to assist planning, please fill out this poll to indicate your =
availability:
>   http://doodle.com/poll/ams8q39eecud4zih
>=20
> In particular, if you'd like to actively participate but the cannot =
make those dates, please indicate that with "Cannot make it."
>=20
> We'll have more details soon.
>=20
> Cheers,
>=20
> --
> Mark Nottingham   https://www.mnot.net/
>=20
>=20
>=20
> Sent from my iPhone
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>=20
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic


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Subject: Re: [QUIC] First Interim planning -- availability poll
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I would second both Ted and Brian's comments.

On Wed, Oct 26, 2016 at 9:55 AM, Brian Trammell <ietf@trammell.ch> wrote:

> And another half-thought: the poll seems to provide no way to indicate
> availability/preference for in-person versus remote attendance. I'd think
> that would be a useful piece of input for the chairs to have.
>
> Cheers,
>
> Brian
>
>
> > On 26 Oct 2016, at 15:52, Ted Hardie <ted.ietf@gmail.com> wrote:
> >
> > Howdy,
> >
> > Two thoughts:  our first working group meeting will be in Asia and our
> second in the U.S.--would you like a space in Europe for the interim?   For
> remote attendees, will we use meetecho or something else?
> >
> > best,
> >
> > Ted
> >
> > On Wed, Oct 26, 2016 at 12:43 AM, Mark Nottingham <mnot@mnot.net> wrote:
> > Everyone,
> >
> > We'd like to get our first interim meeting nailed down soon. We have a
> host with with what we believe to be a suitable space in Tokyo, Japan on
> 24-26 January.
> >
> > To make sure that it's suitable for a good portion of the WG, as well as
> to assist planning, please fill out this poll to indicate your availability:
> >   http://doodle.com/poll/ams8q39eecud4zih
> >
> > In particular, if you'd like to actively participate but the cannot make
> those dates, please indicate that with "Cannot make it."
> >
> > We'll have more details soon.
> >
> > Cheers,
> >
> > --
> > Mark Nottingham   https://www.mnot.net/
> >
> >
> >
> > Sent from my iPhone
> > _______________________________________________
> > QUIC mailing list
> > QUIC@ietf.org
> > https://www.ietf.org/mailman/listinfo/quic
> >
> > _______________________________________________
> > QUIC mailing list
> > QUIC@ietf.org
> > https://www.ietf.org/mailman/listinfo/quic
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>
>

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<div dir=3D"ltr">I would second both Ted and Brian&#39;s comments.</div><di=
v class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Oct 26, 2016=
 at 9:55 AM, Brian Trammell <span dir=3D"ltr">&lt;<a href=3D"mailto:ietf@tr=
ammell.ch" target=3D"_blank">ietf@trammell.ch</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">And another half-thought: the poll seems to prov=
ide no way to indicate availability/preference for in-person versus remote =
attendance. I&#39;d think that would be a useful piece of input for the cha=
irs to have.<br>
<br>
Cheers,<br>
<br>
Brian<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
&gt; On 26 Oct 2016, at 15:52, Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gm=
ail.com">ted.ietf@gmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Howdy,<br>
&gt;<br>
&gt; Two thoughts:=C2=A0 our first working group meeting will be in Asia an=
d our second in the U.S.--would you like a space in Europe for the interim?=
=C2=A0 =C2=A0For remote attendees, will we use meetecho or something else?<=
br>
&gt;<br>
&gt; best,<br>
&gt;<br>
&gt; Ted<br>
&gt;<br>
&gt; On Wed, Oct 26, 2016 at 12:43 AM, Mark Nottingham &lt;<a href=3D"mailt=
o:mnot@mnot.net">mnot@mnot.net</a>&gt; wrote:<br>
&gt; Everyone,<br>
&gt;<br>
&gt; We&#39;d like to get our first interim meeting nailed down soon. We ha=
ve a host with with what we believe to be a suitable space in Tokyo, Japan =
on 24-26 January.<br>
&gt;<br>
&gt; To make sure that it&#39;s suitable for a good portion of the WG, as w=
ell as to assist planning, please fill out this poll to indicate your avail=
ability:<br>
&gt;=C2=A0 =C2=A0<a href=3D"http://doodle.com/poll/ams8q39eecud4zih" rel=3D=
"noreferrer" target=3D"_blank">http://doodle.com/poll/<wbr>ams8q39eecud4zih=
</a><br>
&gt;<br>
&gt; In particular, if you&#39;d like to actively participate but the canno=
t make those dates, please indicate that with &quot;Cannot make it.&quot;<b=
r>
&gt;<br>
&gt; We&#39;ll have more details soon.<br>
&gt;<br>
&gt; Cheers,<br>
&gt;<br>
&gt; --<br>
&gt; Mark Nottingham=C2=A0 =C2=A0<a href=3D"https://www.mnot.net/" rel=3D"n=
oreferrer" target=3D"_blank">https://www.mnot.net/</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Sent from my iPhone<br>
&gt; ______________________________<wbr>_________________<br>
&gt; QUIC mailing list<br>
&gt; <a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferr=
er" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><b=
r>
&gt;<br>
&gt; ______________________________<wbr>_________________<br>
&gt; QUIC mailing list<br>
&gt; <a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferr=
er" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><b=
r>
<br>
</div></div><br>______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
<br></blockquote></div><br></div>

--f403045dc5d254dde0053fc533d9--


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From: Jim Roskind <JimRoskind@gmail.com>
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To: Ryan Hamilton <rch@google.com>
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Cc: Ian Swett <ianswett@google.com>, Kyle Rose <krose@krose.org>, Christian Huitema <huitema@huitema.net>, Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>, marcelo bagnulo braun <marcelo@it.uc3m.es>, Jana Iyengar <jri@google.com>, quic@ietf.org
Subject: Re: [QUIC] Drafts structure and split
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On Tue, Oct 25, 2016 at 9:09 PM, Ryan Hamilton <rch@google.com> wrote:

>
>
> On Tue, Oct 25, 2016 at 7:13 PM, Jim Roskind <JimRoskind@gmail.com> wrote=
:
>
>> On Mon, Oct 24, 2016 at 5:31 PM, Ian Swett <ianswett@google.com> wrote:
>>
>>> ...
>>> It's worth noting that BBR is only a change to the congestion
>>> controller, ....
>>>
>>>
>>>
>> BBR relies on getting estimates of transit time, so as to estimate
>> variations in buffer sizes.  As a relatively rough estimate, I believe B=
BR
>> in TCP relies on traditional-TCP RTT calculations, which includes the ro=
und
>> trip sum of transit times in both directions.  I expect that better one-=
way
>> transit time estimates can be harvested in QUIC, by changing the data
>> included in ACK frames to better illuminate transit times in each
>> directions (or more precisely, illuminate changes, from one packet to th=
e
>> next, in one-way transit times).
>>
>
> =E2=80=8BCorrect me if I'm wrong, but QUIC ACK frames already have this
> information. The contain the arrival time of each packet at the peer, whi=
ch
> provides the one-way transfer time.
>

Absolutely (and since you implemented and are probably maintaining this in
the Google implementation, you'd know if there was any change in that
status quo). The cross-stack requirements in the QUIC Rationale and Design
Doc motivated the placement and use of timestamps in ACK frames for
evaluating bottleneck bandwidth back in 2012. All that "frame level" design
was part of congestion control design, even though at the time we were only
using traditional TCP-style congestion control (which did not call for the
time extra timestamps).

All of this is indeed a reflection of the tight integration between the
elements, and across the stack. That is why I'd argue that it is difficult
to divide a QUIC standard (into a multitude of separate docs) across
layers, when delicate details in one layer are heavily depended upon, and
perchance varied greatly, to better suit needs of would-be "other layers."

For instance, consideration may be given to adjust the resolution, range,
or coding of those ACK-frame timestamps specifically for congestion
algorithms like BBR, but such a change would sadly not be confined to
otherwise isolated docs about congestion avoidance. :-(  As other folks
look at, and experiment with, other variants of congestion control, such as
ECN, it becomes plausible that different metrics will be needed at
different levels of the protocol.

It would be a shame if standardization in one "layer" made it more
difficult to rapidly evolve broader advances for QUIC.

I guess on the other side, it sure would be nice if we were all brilliant
enough to standardize ALL required hooks...but that seems more than hard.

If we can find nice partitioning points, that would be great, but I was
hard pressed to expect much more than the crypto separation, and still
allow for rapid ongoing evolution. Even with crypto, there were interesting
cross-stack requirements, but getting crypto right is soooo hard, that
separation seemed very prudent (along with somewhat freezing supporting
issues elsewhere in the protocol).

Jim

[Opinions expressed are my own, and not that of my employer]

--94eb2c14ea2c853103053fc66e7b
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Tue, Oct 25, 2016 at 9:09 PM, Ryan Hamilton <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:rch@google.com" target=3D"_blank">rch@google.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"><div style=3D"fo=
nt-family:trebuchet ms,sans-serif"><br></div><div class=3D"gmail_extra"><br=
><div class=3D"gmail_quote"><span class=3D"">On Tue, Oct 25, 2016 at 7:13 P=
M, Jim Roskind <span dir=3D"ltr">&lt;<a href=3D"mailto:JimRoskind@gmail.com=
" class=3D"m_5245255675730322491cremed" target=3D"_blank">JimRoskind@gmail.=
com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr=
"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On Mon, Oct 24, 201=
6 at 5:31 PM, Ian Swett <span dir=3D"ltr">&lt;<a href=3D"mailto:ianswett@go=
ogle.com" class=3D"m_5245255675730322491cremed" target=3D"_blank">ianswett@=
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><div>It&#39;s worth noting that BBR is only a change=
 to the congestion controller, ....</div></div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote"><div><div class=3D"m_5245255675730322491m_-26=
46942042275758490h5"><br></div></div></div></div></blockquote><div><br></di=
v><div>BBR relies on getting estimates of transit time, so as to estimate v=
ariations in buffer sizes.=C2=A0 As a relatively rough estimate, I believe =
BBR in TCP relies on traditional-TCP RTT calculations, which includes the r=
ound trip sum of transit times in both directions.=C2=A0 I expect that bett=
er one-way transit time estimates can be harvested in QUIC, by changing the=
 data included in ACK frames to better illuminate transit times in each dir=
ections (or more precisely, illuminate changes, from one packet to the next=
, in one-way transit times).</div></div></div></div></blockquote><div><br><=
/div></span><div><div style=3D"font-family:&quot;trebuchet ms&quot;,sans-se=
rif;display:inline">=E2=80=8BCorrect me if I&#39;m wrong, but QUIC ACK fram=
es already have this information. The contain the arrival time of each pack=
et at the peer, which provides the one-way transfer time.</div></div></div>=
</div></div></blockquote><div><br></div><div>Absolutely (and since you impl=
emented and are probably maintaining this in the Google implementation, you=
&#39;d know if there was any change in that status quo). The cross-stack re=
quirements in the QUIC Rationale and Design Doc motivated the placement and=
 use of timestamps in ACK frames for evaluating bottleneck bandwidth back i=
n 2012. All that &quot;frame level&quot; design was part of congestion cont=
rol design, even though at the time we were only using traditional TCP-styl=
e congestion control (which did not call for the time extra timestamps).</d=
iv><div><br></div><div>All of this is indeed a reflection of the tight inte=
gration between the elements, and across the stack. That is why I&#39;d arg=
ue that it is difficult to divide a QUIC standard (into a multitude of sepa=
rate docs) across layers, when delicate details in one layer are heavily de=
pended upon, and perchance varied greatly, to better suit needs of would-be=
 &quot;other layers.&quot; =C2=A0</div><div><br></div><div>For instance, co=
nsideration may be given to adjust the resolution, range, or coding of thos=
e ACK-frame timestamps specifically for congestion algorithms like BBR, but=
 such a change would sadly not be confined to otherwise isolated docs about=
 congestion avoidance. :-( =C2=A0As other folks look at, and experiment wit=
h, other variants of congestion control, such as ECN, it becomes plausible =
that different metrics will be needed at different levels of the protocol. =
=C2=A0</div><div><br></div><div>It would be a shame if standardization in o=
ne &quot;layer&quot; made it more difficult to rapidly evolve broader advan=
ces for QUIC.</div><div><br></div><div>I guess on the other side, it sure w=
ould be nice if we were all brilliant enough to standardize ALL required ho=
oks...but that seems more than hard.</div><div><br></div><div>If we can fin=
d nice partitioning points, that would be great, but I was hard pressed to =
expect much more than the crypto separation, and still allow for rapid ongo=
ing evolution. Even with crypto, there were interesting cross-stack require=
ments, but getting crypto right is soooo hard, that separation seemed very =
prudent (along with somewhat freezing supporting issues elsewhere in the pr=
otocol).</div><div><br></div><div>Jim</div><div><br></div><div>[Opinions ex=
pressed are my own, and not that of my employer]</div><div><br></div></div>=
</div></div>

--94eb2c14ea2c853103053fc66e7b--


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Ryan Hamilton <rch@google.com>
Thread-Topic: [QUIC] QUIC ACK frames and one-way transfer time (was RE: Drafts structure and split)
Thread-Index: AQHSL499mWSd+VhNgk+Qp+hFPjBxkaC69qZw
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Cc: Ian Swett <ianswett@google.com>, Kyle Rose <krose@krose.org>, Christian Huitema <huitema@huitema.net>, Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>, Jana Iyengar <jri@google.com>, marcelo bagnulo braun <marcelo@it.uc3m.es>, Jim Roskind <JimRoskind@gmail.com>, "quic@ietf.org" <quic@ietf.org>
Subject: Re: [QUIC] QUIC ACK frames and one-way transfer time (was RE: Drafts structure and split)
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Cc: 'Ian Swett' <ianswett@google.com>, 'Kyle Rose' <krose@krose.org>, 'Spencer Dawkins at IETF' <spencerdawkins.ietf@gmail.com>, 'Jim Roskind' <JimRoskind@gmail.com>, 'marcelo bagnulo braun' <marcelo@it.uc3m.es>, 'Jana Iyengar' <jri@google.com>, quic@ietf.org
Subject: Re: [QUIC] QUIC ACK frames and one-way transfer time (was RE: Drafts structure and split)
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On Wednesday, October 26, 2016 10:10 AM, Ingemar Johansson wrote:

> That was interesting.=20
> I probably need some pen and paper exercise to figure out how good =
synchronization
> one can get this. If it is within 20-30ms (rough figure) then it can =
be useful. Otherwise
> it can still be useful as it can help to remedy for instance late =
comers advantage and=20
> base-delay inflation issues that is a known problem for LEDBAT.

If an implementer really feels ambitious, there is enough information in =
the ACK to run an NTP style algorithm and compute skew and lag. That =
will drive the precision in the millisecond range. Of course, that's not =
really necessary for any practical purpose. The receiver measured ACK =
delay is quite sufficient.

-- Christian Huitema





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Subject: Re: [QUIC] First Interim planning -- availability poll
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Hi Ted,

> On 27 Oct. 2016, at 12:52 am, Ted Hardie <ted.ietf@gmail.com> wrote:
>=20
> Howdy,
>=20
> Two thoughts:  our first working group meeting will be in Asia and our =
second in the U.S.--would you like a space in Europe for the interim?=20

We're going to have a broad spread of locations, so we'll get there (and =
Jana made the same offer ;)


>  For remote attendees, will we use meetecho or something else?

We don't have details for this yet, but we'll have some way for people =
to follow the meeting remotely.

Note that actively participating remotely is difficult, and while we'll =
try to accommodate it, if you want to actively participate, we strongly =
encourage you to be there in person.=20

To Brian's question - the poll is just to give us a rough idea of =
people's availability, to make sure that the dates we've got are =
reasonable (and to give people some dates to pencil in). We'll do a more =
formal registration to get exact numbers and in-person vs. remote later.

Cheers,


>=20
> best,
>=20
> Ted
>=20
> On Wed, Oct 26, 2016 at 12:43 AM, Mark Nottingham <mnot@mnot.net> =
wrote:
> Everyone,
>=20
> We'd like to get our first interim meeting nailed down soon. We have a =
host with with what we believe to be a suitable space in Tokyo, Japan on =
24-26 January.
>=20
> To make sure that it's suitable for a good portion of the WG, as well =
as to assist planning, please fill out this poll to indicate your =
availability:
>   http://doodle.com/poll/ams8q39eecud4zih
>=20
> In particular, if you'd like to actively participate but the cannot =
make those dates, please indicate that with "Cannot make it."
>=20
> We'll have more details soon.
>=20
> Cheers,
>=20
> --
> Mark Nottingham   https://www.mnot.net/
>=20
>=20
>=20
> Sent from my iPhone
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>=20

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





From nobody Wed Oct 26 14:48:22 2016
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Subject: Re: [QUIC] First Interim planning -- availability poll
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Hi Mark,

> Am 26.10.2016 um 22:58 schrieb Mark Nottingham <mnot@mnot.net>:
>=20
>> For remote attendees, will we use meetecho or something else?
>=20
> We don't have details for this yet, but we'll have some way for people =
to follow the meeting remotely.
>=20
> Note that actively participating remotely is difficult, and while =
we'll try to accommodate it, if you want to actively participate, we =
strongly encourage you to be there in person.=20

There is always an IETF Webex or Meetecho available to use, so all you =
need is Internet, speakers and a mic or just a phone!

Mirja


From nobody Wed Oct 26 14:56:49 2016
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Subject: Re: [QUIC] First Interim planning -- availability poll
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Hi Mark Agien,

> Am 26.10.2016 um 22:58 schrieb Mark Nottingham <mnot@mnot.net>:
>=20
>> Two thoughts:  our first working group meeting will be in Asia and =
our second in the U.S.--would you like a space in Europe for the =
interim?=20
>=20
> We're going to have a broad spread of locations, so we'll get there =
(and Jana made the same offer ;)

Given we have this standing offer, can we maybe have poll between the =
participants where they prefer an interim end of November?

Thanks,
Mirja=


From nobody Wed Oct 26 15:51:48 2016
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Subject: Re: [QUIC] First Interim planning -- availability poll
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Hi Mirja,

I don't think we have enough lead time for that, and the end of November is g=
oing to be a conflict for pretty much all US people.=20

Sent from my iPhone

> On 27 Oct. 2016, at 8:56 am, Mirja K=C3=BChlewind <mirja.kuehlewind@tik.ee=
.ethz.ch> wrote:
>=20
> Hi Mark Agien,
>=20
>>> Am 26.10.2016 um 22:58 schrieb Mark Nottingham <mnot@mnot.net>:
>>>=20
>>> Two thoughts:  our first working group meeting will be in Asia and our s=
econd in the U.S.--would you like a space in Europe for the interim?=20
>>=20
>> We're going to have a broad spread of locations, so we'll get there (and J=
ana made the same offer ;)
>=20
> Given we have this standing offer, can we maybe have poll between the part=
icipants where they prefer an interim end of November?
>=20
> Thanks,
> Mirja


From nobody Wed Oct 26 15:59:19 2016
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To: Mark Nottingham <mnot@mnot.net>
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Subject: Re: [QUIC] First Interim planning -- availability poll
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Sorry, I meant end of January. The point was to ask the participant =
where they=E2=80=99d prefer a meeting.

Mirja


> Am 27.10.2016 um 00:51 schrieb Mark Nottingham <mnot@mnot.net>:
>=20
> Hi Mirja,
>=20
> I don't think we have enough lead time for that, and the end of =
November is going to be a conflict for pretty much all US people.=20
>=20
> Sent from my iPhone
>=20
>> On 27 Oct. 2016, at 8:56 am, Mirja K=C3=BChlewind =
<mirja.kuehlewind@tik.ee.ethz.ch> wrote:
>>=20
>> Hi Mark Agien,
>>=20
>>>> Am 26.10.2016 um 22:58 schrieb Mark Nottingham <mnot@mnot.net>:
>>>>=20
>>>> Two thoughts:  our first working group meeting will be in Asia and =
our second in the U.S.--would you like a space in Europe for the =
interim?=20
>>>=20
>>> We're going to have a broad spread of locations, so we'll get there =
(and Jana made the same offer ;)
>>=20
>> Given we have this standing offer, can we maybe have poll between the =
participants where they prefer an interim end of November?
>>=20
>> Thanks,
>> Mirja
>=20
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic


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Subject: Re: [QUIC] First Interim planning -- availability poll
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I want to see if the arrangements we have are viable first.=20

Sent from my iPhone

> On 27 Oct. 2016, at 9:59 am, Mirja K=C3=BChlewind <mirja.kuehlewind@tik.ee=
.ethz.ch> wrote:
>=20
> Sorry, I meant end of January. The point was to ask the participant where t=
hey=E2=80=99d prefer a meeting.
>=20
> Mirja
>=20
>=20
>> Am 27.10.2016 um 00:51 schrieb Mark Nottingham <mnot@mnot.net>:
>>=20
>> Hi Mirja,
>>=20
>> I don't think we have enough lead time for that, and the end of November i=
s going to be a conflict for pretty much all US people.=20
>>=20
>> Sent from my iPhone
>>=20
>>> On 27 Oct. 2016, at 8:56 am, Mirja K=C3=BChlewind <mirja.kuehlewind@tik.=
ee.ethz.ch> wrote:
>>>=20
>>> Hi Mark Agien,
>>>=20
>>>>> Am 26.10.2016 um 22:58 schrieb Mark Nottingham <mnot@mnot.net>:
>>>>>=20
>>>>> Two thoughts:  our first working group meeting will be in Asia and our=
 second in the U.S.--would you like a space in Europe for the interim?=20
>>>>=20
>>>> We're going to have a broad spread of locations, so we'll get there (an=
d Jana made the same offer ;)
>>>=20
>>> Given we have this standing offer, can we maybe have poll between the pa=
rticipants where they prefer an interim end of November?
>>>=20
>>> Thanks,
>>> Mirja
>>=20
>> _______________________________________________
>> QUIC mailing list
>> QUIC@ietf.org
>> https://www.ietf.org/mailman/listinfo/quic
>=20


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Subject: Re: [QUIC] First Interim planning -- availability poll
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> Sorry, I meant end of January. The point was to ask the participant =
where they=E2=80=99d prefer a meeting.

End of January? Hawaii would be nice.

-- Christian Huitema




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Subject: Re: [QUIC] First Interim planning -- availability poll
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Following in the path of the IETFs that were conducted in Minneapolis in
March, arguably Japan would be more productive :-)

Seriously though, I expect that we'll likely have several meetings next
year in various locations. Since Japan's being proffered right now by the
chairs, it seems reasonable to see if this can work for many active QUIC
participants.

I'd be interested in seeing remote participation made more workable
however. There are many reasons that even active participants may not be
able to make it to any proposed location. Visa is just one of these
considerations --  it's not the difficulty of getting a visa, the prep and
lead time required for it can be debilitating. Folks who are keen to be
active ought to be able to remotely participate despite the inconvenient
time difference.

On Wed, Oct 26, 2016 at 9:10 PM, Christian Huitema <huitema@huitema.net>
wrote:

> > Sorry, I meant end of January. The point was to ask the participant
> where they=E2=80=99d prefer a meeting.
>
> End of January? Hawaii would be nice.
>
> -- Christian Huitema
>
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

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<div dir=3D"ltr">Following in the path of the IETFs that were conducted in =
Minneapolis in March, arguably Japan would be more productive :-)<br><br>Se=
riously though, I expect that we&#39;ll likely have several meetings next y=
ear in various locations. Since Japan&#39;s being proffered right now by th=
e chairs, it seems reasonable to see if this can work for many active QUIC =
participants.<div><br></div><div>I&#39;d be interested in seeing remote par=
ticipation made more workable however. There are many reasons that even act=
ive participants may not be able to make it to any proposed location. Visa =
is just one of these considerations -- =C2=A0it&#39;s not the difficulty of=
 getting a visa, the prep and lead time required for it can be debilitating=
. Folks who are keen to be active ought to be able to remotely participate =
despite the inconvenient time difference.</div></div><div class=3D"gmail_ex=
tra"><br><div class=3D"gmail_quote">On Wed, Oct 26, 2016 at 9:10 PM, Christ=
ian Huitema <span dir=3D"ltr">&lt;<a href=3D"mailto:huitema@huitema.net" ta=
rget=3D"_blank">huitema@huitema.net</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex"><span class=3D"">&gt; Sorry, I meant end of January. The p=
oint was to ask the participant where they=E2=80=99d prefer a meeting.<br>
<br>
</span>End of January? Hawaii would be nice.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
-- Christian Huitema<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
</div></div></blockquote></div><br></div>

--94eb2c0d06d0b008d3053fd128a1--


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To: Jana Iyengar <jri@google.com>, Christian Huitema <huitema@huitema.net>
References: <4B3A8FA0-F5CE-4320-B4BD-9DF13057918E@mnot.net> <CA+9kkMCiQWSfWxUkKBjsv=xmMeEHeAaQ=NM_u4MyNzKpjUr24w@mail.gmail.com> <65B2FEE9-2CF4-4526-823B-905216897096@mnot.net> <21DE1D15-2C82-42D0-A486-C9C773443B15@tik.ee.ethz.ch> <85A2BDE4-793A-4E47-920F-4EDFF6CA3EE5@mnot.net> <210E0B65-2793-4139-BD19-B63045C30675@tik.ee.ethz.ch> <031d01d23008$006f6e20$014e4a60$@huitema.net> <CAGD1bZbhuWtWBZtHN0FY1vYGxvDxFj6XWyO42Ggzpsq2+q48EA@mail.gmail.com>
From: =?UTF-8?Q?Mirja_K=c3=bchlewind?= <mirja.kuehlewind@tik.ee.ethz.ch>
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Date: Thu, 27 Oct 2016 09:53:54 +0200
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Cc: Brian Trammell <ietf@trammell.ch>, Mark Nottingham <mnot@mnot.net>, quic@ietf.org, Lars Eggert <lars@netapp.com>, Ted Hardie <ted.ietf@gmail.com>
Subject: Re: [QUIC] First Interim planning -- availability poll
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Hi Jana, hi Mark, hi all,

I personally don't really see a reason to make this a QUIC-around-the-world 
tour. We should a meeting location where it is most practical for most of the 
people that want to participate. I can't see who filled the doodle but that 
would probably give some indication about a suitable meeting location. As we 
already have some offers, I 'm sure it wouldn't be a problem to find a room 
in Europe or the US.

Further, I think fully interactive remote participation is a must. The IETF 
provides you WebEx or meetecho, so that shouldn't be a problem. If there is 
any reason at a certain location that you could use it, it is not the right 
location.

And finally, one question: Do you want people to fill the doodle if they 
would be able to participate in person on those dates in Tokyo, or would be 
able to participate on those date but remotely, or when I'm supposed to say 
yes or no?

Mirja


On 27.10.2016 06:27, Jana Iyengar wrote:
> Following in the path of the IETFs that were conducted in Minneapolis in
> March, arguably Japan would be more productive :-)
>
> Seriously though, I expect that we'll likely have several meetings next year
> in various locations. Since Japan's being proffered right now by the chairs,
> it seems reasonable to see if this can work for many active QUIC participants.
>
> I'd be interested in seeing remote participation made more workable however.
> There are many reasons that even active participants may not be able to make
> it to any proposed location. Visa is just one of these considerations --
>  it's not the difficulty of getting a visa, the prep and lead time required
> for it can be debilitating. Folks who are keen to be active ought to be able
> to remotely participate despite the inconvenient time difference.
>
> On Wed, Oct 26, 2016 at 9:10 PM, Christian Huitema <huitema@huitema.net
> <mailto:huitema@huitema.net>> wrote:
>
>     > Sorry, I meant end of January. The point was to ask the participant where they’d prefer a meeting.
>
>     End of January? Hawaii would be nice.
>
>     -- Christian Huitema
>
>
>
>     _______________________________________________
>     QUIC mailing list
>     QUIC@ietf.org <mailto:QUIC@ietf.org>
>     https://www.ietf.org/mailman/listinfo/quic
>     <https://www.ietf.org/mailman/listinfo/quic>
>
>


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From: "Eggert, Lars" <lars@netapp.com>
To: Jana Iyengar <jri@google.com>
Thread-Topic: [QUIC] First Interim planning -- availability poll
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Subject: Re: [QUIC] First Interim planning -- availability poll
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Hi,

On 2016-10-27, at 6:27, Jana Iyengar <jri@google.com> wrote:
> Seriously though, I expect that we'll likely have several meetings next y=
ear in various locations. Since Japan's being proffered right now by the ch=
airs, it seems reasonable to see if this can work for many active QUIC part=
icipants.

this is at least my thinking here as well, and likely Mark's. We're going t=
o be having meetings every 2-3 months (at IETFs and interims), and so we're=
 going to be going places that are inconvenient for some, some of the time.=
 I'd ask that people trust the chairs to select a reasonable mix of locatio=
ns over the coming months.

I really do not think it is a productive use of WG time to meeting-plan by =
consensus. We're very open to offers to host (and already have received a f=
ew off-list - thanks!), and we encourage participants to suggest locations =
that are convenient for them where they can offer to host. For this first m=
eeting, we have a host at a location that has worked before.=20

> I'd be interested in seeing remote participation made more workable howev=
er. There are many reasons that even active participants may not be able to=
 make it to any proposed location. Visa is just one of these considerations=
 --  it's not the difficulty of getting a visa, the prep and lead time requ=
ired for it can be debilitating. Folks who are keen to be active ought to b=
e able to remotely participate despite the inconvenient time difference.

Agreed. There are at least Meetecho and Webex available via the IETF, but i=
f participants have good experiences with other solutions, we're very open =
to use something else.

The main issue we'll be facing is likely going to be with the (lack of) A/V=
 installation at the meeting venue - I expect these interims to be large (>=
50 people) and that's way past where using a single mic and camera is going=
 to work. So even passive remote participation is likely going to be challe=
nge. For active remote participation, add speaker/projector setup into the =
mix. if anyone has suggestions for how we can reliably provide a better rem=
ote experience, please contact the chairs. (If feels like there should be a=
 kickstarter for a "remote-meeting-in-a-box" hardware package.)

Lars=


From nobody Fri Oct 28 21:24:32 2016
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From: Ted Hardie <ted.ietf@gmail.com>
Date: Fri, 28 Oct 2016 09:50:19 -0700
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To: Christian Huitema <huitema@huitema.net>
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Cc: Ian Swett <ianswett@google.com>, Kyle Rose <krose@krose.org>, Ryan Hamilton <rch@google.com>, Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, Spencer Dawkins at IETF <spencerdawkins.ietf@gmail.com>, Jana Iyengar <jri@google.com>, marcelo bagnulo braun <marcelo@it.uc3m.es>, Jim Roskind <JimRoskind@gmail.com>, quic@ietf.org
Subject: Re: [QUIC] QUIC ACK frames and one-way transfer time (was RE: Drafts structure and split)
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On Wed, Oct 26, 2016 at 10:30 AM, Christian Huitema <huitema@huitema.net>
wrote:

> On Wednesday, October 26, 2016 10:10 AM, Ingemar Johansson wrote:
>
> > That was interesting.
> > I probably need some pen and paper exercise to figure out how good
> synchronization
> > one can get this. If it is within 20-30ms (rough figure) then it can be
> useful. Otherwise
> > it can still be useful as it can help to remedy for instance late comers
> advantage and
> > base-delay inflation issues that is a known problem for LEDBAT.
>
> If an implementer really feels ambitious, there is enough information in
> the ACK to run an NTP style algorithm and compute skew and lag. That will
> drive the precision in the millisecond range.


My employer may have IPR that covers this approach.  I'm working with our
legal folks to get a disclosure filed.

Of course, that's not really necessary for any practical purpose. The
> receiver measured ACK delay is quite sufficient.
>
>
I agree with this; while there are cases where you might want and need
this, I don't really think this is one of them.

regards,

Ted


> -- Christian Huitema
>
>
>
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

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<div dir=3D"ltr">On Wed, Oct 26, 2016 at 10:30 AM, Christian Huitema <span =
dir=3D"ltr">&lt;<a href=3D"mailto:huitema@huitema.net" target=3D"_blank">hu=
itema@huitema.net</a>&gt;</span> wrote:<br><div class=3D"gmail_extra"><div =
class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px=
 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><sp=
an class=3D"gmail-">On Wednesday, October 26, 2016 10:10 AM, Ingemar Johans=
son wrote:<br>
<br>
&gt; That was interesting.<br>
&gt; I probably need some pen and paper exercise to figure out how good syn=
chronization<br>
&gt; one can get this. If it is within 20-30ms (rough figure) then it can b=
e useful. Otherwise<br>
&gt; it can still be useful as it can help to remedy for instance late come=
rs advantage and<br>
&gt; base-delay inflation issues that is a known problem for LEDBAT.<br>
<br>
</span>If an implementer really feels ambitious, there is enough informatio=
n in the ACK to run an NTP style algorithm and compute skew and lag. That w=
ill drive the precision in the millisecond range. </blockquote><div><br><sp=
an style=3D"color:rgb(34,34,34);font-family:georgia,serif;font-size:small;f=
ont-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;fon=
t-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text=
-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(25=
5,255,255);display:inline;float:none">My employer may have IPR that covers =
this approach.=C2=A0 I&#39;m working with our legal folks to get a disclosu=
re filed.</span><br><br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:=
1ex">Of course, that&#39;s not really necessary for any practical purpose. =
The receiver measured ACK delay is quite sufficient.<br>
<span class=3D"gmail-HOEnZb"><font color=3D"#888888"><br></font></span></bl=
ockquote><div><br></div><div>I agree with this; while there are cases where=
 you might want and need this, I don&#39;t really think this is one of them=
.<br><br></div><div>regards,<br><br></div><div>Ted<br></div><div>=C2=A0</di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left:1px solid rgb(204,204,204);padding-left:1ex"><span class=3D"gmail-HO=
EnZb"><font color=3D"#888888">
-- Christian Huitema<br>
</font></span><div class=3D"gmail-HOEnZb"><div class=3D"gmail-h5"><br>
<br>
<br>
<br>
______________________________<wbr>_________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/quic</a><br>
</div></div></blockquote></div><br></div></div>

--94eb2c0665c60bb0ac053fefa906--

