Network Working Group                                        A. Frindell
Internet-Draft                                                  Facebook
Intended status: Standards Track                              E. Kinnear
Expires: 7 January 2023                                       Apple Inc.
                                                             V. Vasiliev
                                                             6 July 2022

                        WebTransport over HTTP/3


   WebTransport [OVERVIEW] is a protocol framework that enables clients
   constrained by the Web security model to communicate with a remote
   server using a secure multiplexed transport.  This document describes
   a WebTransport protocol that is based on HTTP/3 [HTTP3] and provides
   support for unidirectional streams, bidirectional streams and
   datagrams, all multiplexed within the same HTTP/3 connection.

Note to Readers

   Discussion of this draft takes place on the WebTransport mailing list
   (, which is archived at

   The repository tracking the issues for this draft can be found at
   issues>.  The web API draft corresponding to this document can be
   found at <>.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on 7 January 2023.

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Copyright Notice

   Copyright (c) 2022 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents (
   license-info) in effect on the date of publication of this document.
   Please review these documents carefully, as they describe your rights
   and restrictions with respect to this document.  Code Components
   extracted from this document must include Revised BSD License text as
   described in Section 4.e of the Trust Legal Provisions and are
   provided without warranty as described in the Revised BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Protocol Overview . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Session Establishment . . . . . . . . . . . . . . . . . . . .   4
     3.1.  Establishing a Transport-Capable HTTP/3 Connection  . . .   4
     3.2.  Extended CONNECT in HTTP/3  . . . . . . . . . . . . . . .   4
     3.3.  Creating a New Session  . . . . . . . . . . . . . . . . .   4
     3.4.  Limiting the Number of Simultaneous Sessions  . . . . . .   5
   4.  WebTransport Features . . . . . . . . . . . . . . . . . . . .   5
     4.1.  Unidirectional streams  . . . . . . . . . . . . . . . . .   6
     4.2.  Bidirectional Streams . . . . . . . . . . . . . . . . . .   6
     4.3.  Resetting Data Streams  . . . . . . . . . . . . . . . . .   7
     4.4.  Datagrams . . . . . . . . . . . . . . . . . . . . . . . .   8
     4.5.  Buffering Incoming Streams and Datagrams  . . . . . . . .   8
     4.6.  Interaction with HTTP/3 GOAWAY frame  . . . . . . . . . .   9
   5.  Session Termination . . . . . . . . . . . . . . . . . . . . .   9
   6.  Negotiating the Draft Version . . . . . . . . . . . . . . . .  10
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  10
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  11
     8.1.  Upgrade Token Registration  . . . . . . . . . . . . . . .  11
     8.2.  HTTP/3 SETTINGS Parameter Registration  . . . . . . . . .  11
     8.3.  Frame Type Registration . . . . . . . . . . . . . . . . .  12
     8.4.  Stream Type Registration  . . . . . . . . . . . . . . . .  12
     8.5.  HTTP/3 Error Code Registration  . . . . . . . . . . . . .  12
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  13
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  13
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  14
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  14

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1.  Introduction

   HTTP/3 [HTTP3] is a protocol defined on top of QUIC [RFC9000] that
   can multiplex HTTP requests over a QUIC connection.  This document
   defines a mechanism for multiplexing non-HTTP data with HTTP/3 in a
   manner that conforms with the WebTransport protocol requirements and
   semantics [OVERVIEW].  Using the mechanism described here, multiple
   WebTransport instances can be multiplexed simultaneously with regular
   HTTP traffic on the same HTTP/3 connection.

1.1.  Terminology

   The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "OPTIONAL" in this document are to be interpreted as described in BCP
   14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown here.

   This document follows terminology defined in Section 1.2 of
   [OVERVIEW].  Note that this document distinguishes between a
   WebTransport server and an HTTP/3 server.  An HTTP/3 server is the
   server that terminates HTTP/3 connections; a WebTransport server is
   an application that accepts WebTransport sessions, which can be
   accessed via an HTTP/3 server.

2.  Protocol Overview

   WebTransport servers in general are identified by a pair of authority
   value and path value (defined in [RFC3986] Sections 3.2 and 3.3

   When an HTTP/3 connection is established, both the client and server
   have to send a SETTINGS_ENABLE_WEBTRANSPORT setting in order to
   indicate that they both support WebTransport over HTTP/3.

   WebTransport sessions are initiated inside a given HTTP/3 connection
   by the client, who sends an extended CONNECT request [RFC8441].  If
   the server accepts the request, an WebTransport session is
   established.  The resulting stream will be further referred to as a
   _CONNECT stream_, and its stream ID is used to uniquely identify a
   given WebTransport session within the connection.  The ID of the
   CONNECT stream that established a given WebTransport session will be
   further referred to as a _Session ID_.

   After the session is established, the peers can exchange data using
   the following mechanisms:

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   *  A client can create a bidirectional stream using a special
      indefinite-length HTTP/3 frame that transfers ownership of the
      stream to WebTransport.

   *  A server can create a bidirectional stream, which is possible
      since HTTP/3 does not define any semantics for server-initiated
      bidirectional streams.

   *  Both client and server can create a unidirectional stream using a
      special stream type.

   *  A datagram can be sent using HTTP Datagrams [HTTP-DATAGRAM].

   An WebTransport session is terminated when the CONNECT stream that
   created it is closed.

3.  Session Establishment

3.1.  Establishing a Transport-Capable HTTP/3 Connection

   In order to indicate support for WebTransport, both the client and
   the server MUST send a SETTINGS_ENABLE_WEBTRANSPORT value set to "1"
   in their SETTINGS frame.  The SETTINGS_ENABLE_WEBTRANSPORT parameter
   value SHALL be either "0" or "1", with "0" being the default; an
   endpoint that receives a value other than "0" or "1" MUST close the
   connection with the H3_SETTINGS_ERROR error code.

   The client MUST NOT send a WebTransport request until it has received
   the setting indicating WebTransport support from the server.
   Similarly, the server MUST NOT process any incoming WebTransport
   requests until the client settings have been received, as the client
   may be using a version of WebTransport extension that is different
   from the one used by the server.

3.2.  Extended CONNECT in HTTP/3

   [RFC8441] defines an extended CONNECT method in Section 4, enabled by
   the SETTINGS_ENABLE_CONNECT_PROTOCOL parameter.  That parameter is
   only defined for HTTP/2.  This document does not create a new multi-
   purpose parameter to indicate support for extended CONNECT in HTTP/3;
   instead, the SETTINGS_ENABLE_WEBTRANSPORT setting implies that an
   endpoint supports extended CONNECT.

3.3.  Creating a New Session

   As WebTransport sessions are established over HTTP/3, they are
   identified using the https URI scheme ([HTTP], Section 4.2.2).

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   In order to create a new WebTransport session, a client can send an
   HTTP CONNECT request.  The :protocol pseudo-header field ([RFC8441])
   MUST be set to webtransport.  The :scheme field MUST be https.  Both
   the :authority and the :path value MUST be set; those fields indicate
   the desired WebTransport server.  An Origin header [RFC6454] MUST be
   provided within the request.

   Upon receiving an extended CONNECT request with a :protocol field set
   to webtransport, the HTTP/3 server can check if it has a WebTransport
   server associated with the specified :authority and :path values.  If
   it does not, it SHOULD reply with status code 404 (Section 15.5.4,
   [HTTP]).  If it does, it MAY accept the session by replying with a
   2xx series status code, as defined in Section 15.3 of [HTTP].  The
   WebTransport server MUST verify the Origin header to ensure that the
   specified origin is allowed to access the server in question.

   From the client's perspective, a WebTransport session is established
   when the client receives a 2xx response.  From the server's
   perspective, a session is established once it sends a 2xx response.
   WebTransport over HTTP/3 does not support 0-RTT.

   The webtransport HTTP Upgrade Token uses the Capsule Protocol as
   defined in [HTTP-DATAGRAM].

3.4.  Limiting the Number of Simultaneous Sessions

   From the flow control perspective, WebTransport sessions count
   against the stream flow control just like regular HTTP requests,
   since they are established via an HTTP CONNECT request.  This
   document does not make any effort to introduce a separate flow
   control mechanism for sessions, nor to separate HTTP requests from
   WebTransport data streams.  If the server needs to limit the rate of
   incoming requests, it has alternative mechanisms at its disposal:

   *  HTTP_REQUEST_REJECTED error code defined in [HTTP3] indicates to
      the receiving HTTP/3 stack that the request was not processed in
      any way.

   *  HTTP status code 429 indicates that the request was rejected due
      to rate limiting [RFC6585].  Unlike the previous method, this
      signal is directly propagated to the application.

4.  WebTransport Features

   WebTransport over HTTP/3 provides the following features described in
   [OVERVIEW]: unidirectional streams, bidirectional streams and
   datagrams, initiated by either endpoint.

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   Session IDs are used to demultiplex streams and datagrams belonging
   to different WebTransport sessions.  On the wire, session IDs are
   encoded using the QUIC variable length integer scheme described in

   If at any point a session ID is received that cannot a valid ID for a
   client-initiated bidirectional stream, the recepient MUST close the
   connection with an H3_ID_ERROR error code.

4.1.  Unidirectional streams

   Once established, both endpoints can open unidirectional streams.
   The HTTP/3 unidirectional stream type SHALL be 0x54.  The body of the
   stream SHALL be the stream type, followed by the session ID, encoded
   as a variable-length integer, followed by the user-specified stream
   data (Figure 1).

     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     |                           0x54 (i)                          ...
     |                        Session ID (i)                       ...
     |                         Stream Body                         ...

            Figure 1: Unidirectional WebTransport stream format

4.2.  Bidirectional Streams

   WebTransport endpoints can initiate bidirectional streams by opening
   an HTTP/3 bidirectional stream and sending an HTTP/3 frame with type
   WEBTRANSPORT_STREAM (type=0x41).  The format of the frame SHALL be
   the frame type, followed by the session ID, encoded as a variable-
   length integer, followed by the user-specified stream data
   (Figure 2).  The frame SHALL last until the end of the stream.

     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     |                           0x41 (i)                          ...
     |                        Session ID (i)                       ...
     |                         Stream Body                         ...

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                 Figure 2: WEBTRANSPORT_STREAM frame format

   HTTP/3 does not by itself define any semantics for server-initiated
   bidirectional streams.  If WebTransport setting is negotiated by both
   endpoints, the syntax of the server-initiated bidirectional streams
   SHALL be the same as the syntax of client-initated bidirectional
   streams, that is, a sequence of HTTP/3 frames.  The only frame
   defined by this document for use within server-initiated
   bidirectional streams is WEBTRANSPORT_STREAM.

   TODO: move the paragraph above into a separate draft; define what
   happens with already existing HTTP/3 frames on server-initiated
   bidirectional streams.

4.3.  Resetting Data Streams

   A WebTransport endpoint may send a RESET_STREAM or a STOP_SENDING
   frame for a WebTransport data stream.  Those signals are propagated
   by the WebTransport implementation to the application.

   A WebTransport application SHALL provide an error code for those
   operations.  Since WebTransport shares the error code space with
   HTTP/3, WebTransport application errors for streams are limited to an
   unsigned 8-bit integer, assuming values between 0x00 and 0xff.
   WebTransport implementations SHALL remap those error codes into an
   error range where 0x00 corresponds to 0x52e4a40fa8db, and 0xff
   corresponds to 0x52e4a40fa9e2.  Note that there are code points
   inside that range of form "0x1f * N + 0x21" that are reserved by
   Section 8.1 of [HTTP3]; those have to be accounted for when mapping
   the error codes by skipping them (i.e. the two HTTP/3 error
   codepoints adjacent to a GREASE codepoint would map to two adjacent
   WebTransport application error codepoints).  An example pseudocode
   can be seen in Figure 3.

       first = 0x52e4a40fa8db
       last = 0x52e4a40fa9e2

       def webtransport_code_to_http_code(n):
           return first + n + floor(n / 0x1e)

       def http_code_to_webtransport_code(h):
           assert(first <= h <= last)
           assert((h - 0x21) % 0x1f != 0)
           shifted = h - first
           return shifted - shifted // 0x1f

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          Figure 3: Pseudocode for converting between WebTransport
      application errors and HTTP/3 error codes; here, `//` is integer

   WebTransport data streams are associated with sessions through a
   header at the beginning of the stream; resetting a stream may result
   in that data being discarded.  Because of that, WebTransport
   application error codes are best effort, as the WebTransport stack is
   not always capable of associating the reset code with a session.  The
   only exception is the situation where there is only one session on a
   given HTTP/3 connection, and no intermediaries between the client and
   the server.

   WebTransport implementations SHALL forward the error code for a
   stream associated with a known session to the application that owns
   that session; similarly, the intermediaries SHALL reset the streams
   with corresponding error code when receiving a reset from the peer.
   If a WebTransport implementation intentionally allows only one
   session over a given HTTP/3 connection, it SHALL forward the error
   codes within WebTransport application error code range to the
   application that owns the only session on that connection.

4.4.  Datagrams

   Datagrams can be sent using HTTP Datagrams.  The WebTransport
   datagram payload is sent unmodified in the "HTTP Datagram Payload"
   field of an HTTP Datagram.

4.5.  Buffering Incoming Streams and Datagrams

   In WebTransport over HTTP/3, the client MAY send its SETTINGS frame,
   as well as multiple WebTransport CONNECT requests, WebTransport data
   streams and WebTransport datagrams, all within a single flight.  As
   those can arrive out of order, a WebTransport server could be put
   into a situation where it receives a stream or a datagram without a
   corresponding session.  Similarly, a client may receive a server-
   initiated stream or a datagram before receiving the CONNECT response
   headers from the server.

   To handle this case, WebTransport endpoints SHOULD buffer streams and
   datagrams until those can be associated with an established session.
   To avoid resource exhaustion, the endpoints MUST limit the number of
   buffered streams and datagrams.  When the number of buffered streams
   is exceeded, a stream SHALL be closed by sending a RESET_STREAM and/
   error code.  When the number of buffered datagrams is exceeded, a
   datagram SHALL be dropped.  It is up to an implementation to choose
   what stream or datagram to discard.

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4.6.  Interaction with HTTP/3 GOAWAY frame

   HTTP/3 defines a graceful shutdown mechanism (Section 5.2 of [HTTP3])
   that allows a peer to send a GOAWAY frame indicating that it will no
   longer accept any new incoming requests or pushes.  This mechanism
   applies to the CONNECT requests for new WebTransport sessions.  A
   GOAWAY frame does not affect data streams for existing WebTransport
   sessions; those can continue to be opened even after the GOAWAY frame
   has been sent or received.

5.  Session Termination

   A WebTransport session over HTTP/3 is considered terminated when
   either of the following conditions is met:

   *  the CONNECT stream is closed, either cleanly or abruptly, on
      either side; or

   *  a CLOSE_WEBTRANSPORT_SESSION capsule is either sent or received.

   Upon learning that the session has been terminated, the endpoint MUST
   reset the send side and abort reading on the receive side of all of
   the streams associated with the session (see Section 2.4 of
   [RFC9000]) using the H3_WEBTRANSPORT_SESSION_GONE error code; it MUST
   NOT send any new datagrams or open any new streams.

   To terminate a session with a detailed error message, an application
   MAY send an HTTP capsule [HTTP-DATAGRAM] of type
   CLOSE_WEBTRANSPORT_SESSION (0x2843).  The format of the capsule SHALL
   be as follows:

     Length (i),
     Application Error Code (32),
     Application Error Message (..8192),

   CLOSE_WEBTRANSPORT_SESSION has the following fields:

   Application Error Code:  A 32-bit error code provided by the
      application closing the connection.

   Application Error Message:  A UTF-8 encoded error message string
      provided by the application closing the connection.  The message
      takes up the remainer of the capsule, and its length MUST NOT
      exceed 1024 bytes.

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   A CLOSE_WEBTRANSPORT_SESSION capsule MUST be followed by a FIN on the
   sender side.  If any additional stream data is received on the
   CONNECT stream after CLOSE_WEBTRANSPORT_SESSION, the stream MUST be
   reset with code H3_MESSAGE_ERROR.  The recipient MUST close the
   stream upon receiving a FIN.  If the sender of
   CLOSE_WEBTRANSPORT_SESSION does not receive a FIN after some time, it
   SHOULD send STOP_SENDING on the CONNECT stream.

   Cleanly terminating a CONNECT stream without a
   CLOSE_WEBTRANSPORT_SESSION capsule SHALL be semantically equivalent
   to terminating it with a CLOSE_WEBTRANSPORT_SESSION capsule that has
   an error code of 0 and an empty error string.

6.  Negotiating the Draft Version

   [[RFC editor: please remove this section before publication.]]

   WebTransport over HTTP/3 uses two different mechanisms to negotiate
   versions for the different parts of the draft.

   The hop-by-hop wire format aspects of the protocol are negotiated by
   changing the codepoint used for the SETTINGS_ENABLE_WEBTRANSPORT
   parameter.  Because of that, any WebTransport endpoint MUST wait for
   the peer's SETTINGS frame before sending or processing any
   WebTransport traffic.  When multiple versions are supported by both
   of the peers, the most recent version supported by both is selected.

   The data exchanged over the CONNECT stream is transmitted across
   intermediaries, and thus cannot be versioned using a SETTINGS
   parameter.  To indicate support for different versions of the
   protocol defined in this draft, the clients SHALL send a header for
   each version of the draft supported.  The header corresponding to the
   version described in this draft is Sec-Webtransport-Http3-Draft02;
   its value SHALL be 1.  The server SHALL reply with a Sec-
   Webtransport-Http3-Draft header indicating the selected version; its
   value SHALL be draft02 for the version described in this draft.

7.  Security Considerations

   WebTransport over HTTP/3 satisfies all of the security requirements
   imposed by [OVERVIEW] on WebTransport protocols, thus providing a
   secure framework for client-server communication in cases when the
   client is potentially untrusted.

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   WebTransport over HTTP/3 requires explicit opt-in through the use of
   an HTTP/3 setting; this avoids potential protocol confusion attacks
   by ensuring the HTTP/3 server explicitly supports it.  It also
   requires the use of the Origin header, providing the server with the
   ability to deny access to Web-based clients that do not originate
   from a trusted origin.

   Just like HTTP traffic going over HTTP/3, WebTransport pools traffic
   to different origins within a single connection.  Different origins
   imply different trust domains, meaning that the implementations have
   to treat each transport as potentially hostile towards others on the
   same connection.  One potential attack is a resource exhaustion
   attack: since all of the transports share both congestion control and
   flow control context, a single client aggressively using up those
   resources can cause other transports to stall.  The user agent thus
   SHOULD implement a fairness scheme that ensures that each transport
   within connection gets a reasonable share of controlled resources;
   this applies both to sending data and to opening new streams.

8.  IANA Considerations

8.1.  Upgrade Token Registration

   The following entry is added to the "Hypertext Transfer Protocol
   (HTTP) Upgrade Token Registry" registry established by Section 16.7
   of [HTTP].

   The "webtransport" label identifies HTTP/3 used as a protocol for

   Value:  webtransport

   Description:  WebTransport over HTTP/3

   Reference:  This document and [I-D.ietf-webtrans-http2]

8.2.  HTTP/3 SETTINGS Parameter Registration

   The following entry is added to the "HTTP/3 Settings" registry
   established by [HTTP3]:

   The SETTINGS_ENABLE_WEBTRANSPORT parameter indicates that the
   specified HTTP/3 connection is WebTransport-capable.


   Value:  0x2b603742

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   Default:  0

   Specification:  This document

8.3.  Frame Type Registration

   The following entry is added to the "HTTP/3 Frame Type" registry
   established by [HTTP3]:

   The WEBTRANSPORT_STREAM frame allows HTTP/3 client-initiated
   bidirectional streams to be used by WebTransport:

   Code:  0x41


   Specification:  This document

8.4.  Stream Type Registration

   The following entry is added to the "HTTP/3 Stream Type" registry
   established by [HTTP3]:

   The "WebTransport stream" type allows unidirectional streams to be
   used by WebTransport:

   Code:  0x54

   Stream Type:  WebTransport stream

   Specification:  This document

   Sender:  Both

8.5.  HTTP/3 Error Code Registration

   The following entry is added to the "HTTP/3 Error Code" registry
   established by [HTTP3]:


   Value:  0x3994bd84

   Description:  WebTransport data stream rejected due to lack of
      associated session.

   Specification:  This document.

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   Value:  0x170d7b68

   Description:  WebTransport data stream aborted because the associated
      WebTransport session has been closed.

   Specification:  This document.

   In addition, the following range of entries is registered:


   Value:  0x52e4a40fa8db to 0x52e4a40fa9e2 inclusive, with the
      exception of 0x52e4a40fa8f9, 0x52e4a40fa918, 0x52e4a40fa937,
      0x52e4a40fa956, 0x52e4a40fa975, 0x52e4a40fa994, 0x52e4a40fa9b3,
      and 0x52e4a40fa9d2.

   Description:  WebTransport application error codes.

   Specification:  This document.

9.  References

9.1.  Normative References

   [HTTP]     Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,
              Ed., "HTTP Semantics", STD 97, RFC 9110,
              DOI 10.17487/RFC9110, June 2022,

              Schinazi, D. and L. Pardue, "HTTP Datagrams and the
              Capsule Protocol", Work in Progress, Internet-Draft,
              draft-ietf-masque-h3-datagram-11, 17 June 2022,

   [HTTP3]    Bishop, M., Ed., "HTTP/3", RFC 9114, DOI 10.17487/RFC9114,
              June 2022, <>.

   [OVERVIEW] Vasiliev, V., "The WebTransport Protocol Framework", Work
              in Progress, Internet-Draft, draft-ietf-webtrans-overview-
              03, 7 March 2022, <

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   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
              Resource Identifier (URI): Generic Syntax", STD 66,
              RFC 3986, DOI 10.17487/RFC3986, January 2005,

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              DOI 10.17487/RFC6454, December 2011,

   [RFC6585]  Nottingham, M. and R. Fielding, "Additional HTTP Status
              Codes", RFC 6585, DOI 10.17487/RFC6585, April 2012,

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <>.

   [RFC8441]  McManus, P., "Bootstrapping WebSockets with HTTP/2",
              RFC 8441, DOI 10.17487/RFC8441, September 2018,

   [RFC9000]  Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
              Multiplexed and Secure Transport", RFC 9000,
              DOI 10.17487/RFC9000, May 2021,

9.2.  Informative References

              Frindell, A., Kinnear, E., Pauly, T., Thomson, M.,
              Vasiliev, V., and G. Xie, "WebTransport using HTTP/2",
              Work in Progress, Internet-Draft, draft-ietf-webtrans-
              http2-03, 7 March 2022,

Authors' Addresses

   Alan Frindell

Frindell, et al.         Expires 7 January 2023                [Page 14]

Internet-Draft               WebTransport-H3                   July 2022

   Eric Kinnear
   Apple Inc.

   Victor Vasiliev

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