webtrans                                                     A. Frindell
Internet-Draft                                             Facebook Inc.
Intended status: Standards Track                              E. Kinnear
Expires: 31 January 2022                                        T. Pauly
                                                              Apple Inc.
                                                             V. Vasiliev
                                                                  Google
                                                                  G. Xie
                                                           Facebook Inc.
                                                            30 July 2021


                       WebTransport using HTTP/2
                      draft-ietf-webtrans-http2-01

Abstract

   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/2 [RFC7540] and
   provides support for unidirectional streams, bidirectional streams
   and datagrams, all multiplexed within the same HTTP/2 connection.

Note to Readers

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

   The repository tracking the issues for this draft can be found at
   https://github.com/ietf-wg-webtrans/draft-webtransport-http2.  The
   web API draft corresponding to this document can be found at
   https://w3c.github.io/webtransport/.

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 https://datatracker.ietf.org/drafts/current/.






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   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 31 January 2022.

Copyright Notice

   Copyright (c) 2021 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 (https://trustee.ietf.org/
   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 Simplified BSD License text
   as described in Section 4.e of the Trust Legal Provisions and are
   provided without warranty as described in the Simplified 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/2 Connection  . . .   4
     3.2.  Extended CONNECT in HTTP/2  . . . . . . . . . . . . . . .   4
     3.3.  Creating a New Session  . . . . . . . . . . . . . . . . .   4
     3.4.  Limiting the Number of Simultaneous Sessions  . . . . . .   5
   4.  WebTransport Features . . . . . . . . . . . . . . . . . . . .   5
     4.1.  WT_STREAM Frame . . . . . . . . . . . . . . . . . . . . .   6
     4.2.  WT_RST_STREAM Frame . . . . . . . . . . . . . . . . . . .   7
     4.3.  WT_STOP_SENDING Frame . . . . . . . . . . . . . . . . . .   8
     4.4.  WT_DATAGRAM Frame . . . . . . . . . . . . . . . . . . . .   9
   5.  Session Termination . . . . . . . . . . . . . . . . . . . . .  10
   6.  Transport Properties  . . . . . . . . . . . . . . . . . . . .  11
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  12
     8.1.  HTTP/2 SETTINGS Parameter Registration  . . . . . . . . .  12
     8.2.  Frame Type Registration . . . . . . . . . . . . . . . . .  12
     8.3.  HTTP/2 Error Code Registry  . . . . . . . . . . . . . . .  13
     8.4.  Examples  . . . . . . . . . . . . . . . . . . . . . . . .  13
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  15
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  15
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  16
   Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . .  17



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   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  17

1.  Introduction

   Currently, the only mechanism in HTTP/2 for server to client
   communication is server push.  That is, servers can initiate
   unidirectional push promised streams to clients, but clients cannot
   respond to them; they can only accept them or discard them.
   Additionally, intermediaries along the path may have different server
   push policies and may not forward push promised streams to the
   downstream client.  This best effort mechanism is not sufficient to
   reliably deliver messages from servers to clients, limiting server to
   client use-cases such as chat messages or notifications.

   Several techniques have been developed to workaround these
   limitations: long polling [RFC6202], WebSocket [RFC8441], and
   tunneling using the CONNECT method.  All of these approaches have
   limitations.

   This document defines a mechanism for multiplexing non-HTTP data with
   HTTP/2 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/2
   connection.

1.1.  Terminology

   The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "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/2 server.  An HTTP/2 server is the
   server that terminates HTTP/2 connections; a WebTransport server is
   an application that accepts WebTransport sessions, which can be
   accessed via an HTTP/2 server.

2.  Protocol Overview

   WebTransport servers are identified by an HTTPS URI as defined in
   Section 4.2.2 of [HTTP].






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   When an HTTP/2 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/2.

   WebTransport sessions are initiated inside a given HTTP/2 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:

   *  Both client and server can create a bidirectional or
      unidirectional stream using a new HTTP/2 extension frame
      (WT_STREAM)

   *  A datagram can be sent using a new HTTP/2 extension frame
      WT_DATAGRAM.

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

3.  Session Establishment

3.1.  Establishing a Transport-Capable HTTP/2 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.  Endpoints MUST NOT use any WebTransport-
   related functionality unless the parameter has been negotiated.

3.2.  Extended CONNECT in HTTP/2

   [RFC8441] defines an extended CONNECT method in Section 4, enabled by
   the SETTINGS_ENABLE_CONNECT_PROTOCOL parameter.  An endpoint does not
   need to send both SETTINGS_ENABLE_CONNECT_PROTOCOL and
   SETTINGS_ENABLE_WEBTRANSPORT; 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/2, they are
   identified using the "https" URI scheme [RFC7230].




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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" (Section 7.1 of
   [WEBTRANSPORT-H3]).  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/2 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 6.5.4, [RFC7231]).  If it does, it MAY accept the session by
   replying with status code 200.  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 200 response.  From the server's
   perspective, a session is established once it sends a 200 response.
   Both endpoints MUST NOT open any streams or send any datagrams on a
   given session before that session is established.

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_STREAM_REFUSED" error code defined in [RFC7540] indicates to
      the receiving HTTP/2 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/2 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 a 31-bit integer field.

4.1.  WT_STREAM Frame

   A new HTTP/2 frame called WT_STREAM is introduced for either endpoint
   to establish WebTransport streams.  WT_STREAM frames can be sent on a
   stream in the "idle", "reserved (local)", "open", or "half-closed
   (remote)" state.

    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
   +---------------+
   |Pad Length? (8)|
   +-+-------------+-----------------------------------------------+
   |R|                     Session ID (31)                         |
   +-+-------------------------------------------------------------+
   |                           Padding (*)                       ...
   +---------------------------------------------------------------+

                      Figure 1: WT_STREAM Frame Format

   The WT_STREAM frame define the following fields:

   Pad Length: An 8-bit field containing the length of the frame padding
   in units of octets.  This field is conditional (as signified by a "?"
   in the diagram) and is only present if the PADDED flag is set.

   Session ID: An unsigned 31-bit integer that identifies the stream
   Connect Stream for this Web Transport stream.  The Session ID MUST be
   MUST be an open stream negotiated via the extended CONNECT protocol
   with a ":protocol" value of "webtransport".

   The WT_STREAM frame defines the following flags:

   UNIDIRECTIONAL (0x1): When set, the stream begins in the "half-closed
   (remote)" state at the sender, and in the "half-closed (local)" state
   at the receiver.

   As with all HTTP/2 streams, WebTransport streams initiated by a
   client have odd stream IDs and those initiated by a server have even
   stream IDs.








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   The recipient MUST respond with a stream error of type
   WT_STREAM_ERROR if the specified WebTransport Connect Stream does not
   exist, is not a stream established via extended CONNECT to use the
   "webtransport" protocol, or if it is in the "closed" or "half-closed
   (remote)" stream state.

4.2.  WT_RST_STREAM Frame

   The RST_STREAM frame defined in HTTP/2 moves a stream to the "closed"
   state, but WebTransport streams can be reset unidirectionally, as in
   QUIC and HTTP/3.  A new HTTP/2 frame called WT_RST_STREAM is
   introduced for an endpoint to unidirectionally reset a stream for
   writing.  WT_RST_STREAM frames can be sent on a stream in the "open",
   or "half-closed (remote)" state.

    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
   +---------------------------------------------------------------+
   |                        Error Code (32)                        |
   +---------------------------------------------------------------+

                    Figure 2: WT_RST_STREAM Frame Format

   The WT_RST_STREAM frame contains a single unsigned, 32-bit integer
   identifying the error code.  The error code indicates why the stream
   is being abruptly terminated for writing.

   The WT_RST_STREAM frame does not define any flags.

   The WT_RST_STREAM frame half-closes the referenced stream and effects
   the same state machine transitions as sending or receiving the
   END_STREAM flag.  If the sender is in the "open" state, it
   transitions to "half-closed (local)".  If the sender is in the "half-
   closed (remote)" state, it transitions to "closed".  If the receiver
   is in the "open" state, it transitions to "half-closed (remote)".  If
   the receiver is in the "half-closed (local)" state, it transitions to
   "closed".

   After sending a WT_RST_STREAM on a stream, the sender MUST NOT send
   additional DATA frames for that stream.  If another frame is received
   on a stream after receiving a WT_RST_STREAM, the recipient MUST treat
   this as a connection error of type PROTOCOL_ERROR.

   WT_RST_STREAM frames MUST be associated with a WebTransport stream.
   If a WT_RST_STREAM frame is received with a stream identifier of 0x0,
   or a request or push stream, the recipient MUST treat this as a
   connection error of type PROTOCOL_ERROR.




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   WT_RST_STREAM frames MUST NOT be sent for a stream in any state other
   than "open" or "half-closed (remote)".  If a WT_RST_STREAM frame
   identifying a stream in the "idle", "reserved (local)", "reserved
   (remote)" state is received, the recipient MUST treat this as a
   connection error (Section 5.4.1) of type PROTOCOL_ERROR.  Because of
   race conditions with WT_STOP_SENDING, it is possible to receive a
   WT_RST_STREAM in the "half-closed (remote)" or "closed" state.  The
   recipient should ignore the frame in this case.

   A WT_RST_STREAM frame with a length other than 4 octets MUST be
   treated as a connection error (Section 5.4.1) of type
   FRAME_SIZE_ERROR.

4.3.  WT_STOP_SENDING Frame

   The RST_STREAM frame defined in HTTP/2 moves a stream to the "closed"
   state, but WebTransport streams can be reset unidirectionally, as in
   QUIC and HTTP/3.  A new HTTP/2 frame called WT_STOP_SENDING is
   introduced for an endpoint to unidirectionally reset a stream for
   reading.  WT_STOP_SENDING frames can be sent on a stream in the
   "open", or "half-closed (local)" state.

    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
   +---------------------------------------------------------------+
   |                        Error Code (32)                        |
   +---------------------------------------------------------------+

                   Figure 3: WT_STOP_SENDING Frame Format

   The WT_STOP_SENDING frame contains a single unsigned, 32-bit integer
   identifying the error code.  The error code indicates why the reading
   the stream is being abandoned.

   The WT_STOP_SENDING frame does not define any flags.

   The WT_STOP_SENDING frame half-closes the referenced stream and
   effects the same state machine transitions as sending or receiving
   the END_STREAM flag.  If the sender is in the open state, it
   transitions to "half-closed (remote)".  If the sender is in the
   "half-closed (local)" state, it transitions to "closed".  If the
   receiver is in the "open" state, it transitions to "half-closed
   (local)".  If the receiver is in the "half-closed (remote)" state, it
   transitions to "closed".







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   After receiving a WT_STOP_SENDING on a stream, the sender MUST NOT
   send additional frames for that stream.  After sending the
   WT_STOP_SENDING, the sending endpoint MUST be prepared to receive and
   handle additional frames sent on the stream that might have been sent
   by the peer prior to the arrival of the WT_STOP_SENDING.

   WT_STOP_SENDING frames MUST be associated with a WebTransport stream.
   If a WT_STOP_SENDING frame is received with a stream identifier of
   0x0, or a request or push stream, the recipient MUST treat this as a
   connection error of type PROTOCOL_ERROR.

   WT_STOP_SENDING frames MUST NOT be sent for a stream in any state
   other than "open" or "half-closed (local)".  It is possible to
   receive a WT_STOP_SENDING in another state however, because the
   sender might have closed or reset the stream while the
   WT_STOP_SENDING was in flight.  If a WT_STOP_SENDING frame
   identifying a stream that is already "closed" or "half-closed
   (local)", the recipient SHOULD ignore the frame.

   It is also possible to receive DATA frames on a WebTransport stream
   in the "half-closed (remote)" or "closed" states if the stream
   transitioned there via WT_STOP_SENDING.  If a DATA frame is received
   on a WebTransport stream in one of these states, the recipient MUST
   account for its contribution against the connection flow-control
   window and MUST NOT treat it as an error.

   A WT_STOP_SENDING frame with a length other than 4 octets MUST be
   treated as a connection error (Section 5.4.1) of type
   FRAME_SIZE_ERROR.

4.4.  WT_DATAGRAM Frame

   A new HTTP/2 frame called WT_DATAGRAM is introduced for either
   endpoint to transmit a datagram.  WT_DATAGRAM frames are sent with
   Stream Identifier 0.

    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
   +---------------+
   |Pad Length? (8)|
   +-+-------------+-----------------------------------------------+
   |R|                     Session ID (31)                         |
   +-+-------------------------------------------------------------+
   |                            Data (*)                         ...
   +---------------------------------------------------------------+
   |                           Padding (*)                       ...
   +---------------------------------------------------------------+




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                     Figure 4: WT_DATAGRAM Frame Format

   The WT_DATAGRAM frame define the following fields:

   Pad Length: An 8-bit field containing the length of the frame padding
   in units of octets.  This field is conditional (as signified by a "?"
   in the diagram) and is only present if the PADDED flag is set.

   Session ID: An unsigned 31-bit integer that identifies the stream
   Connect Stream for this Web Transport stream.  The Session ID MUST be
   MUST be an open stream negotiated via the extended CONNECT protocol
   with a ":protocol" value of "webtransport".

   Data: Application data.  The amount of data is the remainder of the
   frame payload after subtracting the length of the other fields that
   are present.

   The WT_DATAGRAM frame does not define any flags.

   The recipient MAY respond with a stream error of type WT_STREAM_ERROR
   if the specified WebTransport Connect Stream does not exist, is not a
   stream established via extended CONNECT to use the "webtransport"
   protocol, or if it is in the "closed" or "half-closed (remote)"
   stream state.

   The data in WT_DATAGRAM frames is not subject to flow control.  The
   receiver MAY discard this data if it does not have sufficient space
   to buffer it.

   An intermediary could forward the data in a WT_DATAGRAM frame over
   another protocol, such as WebTransport over HTTP/3.  In QUIC, a
   datagram frame can span at most one packet.  Because of that, the
   applications have to know the maximum size of the datagram they can
   send.  However, when proxying the datagrams, the hop-by-hop MTUs can
   vary.

5.  Session Termination

   An WebTransport session over HTTP/2 is terminated when either
   endpoint closes the stream associated with the CONNECT request that
   initiated the session.  Upon learning about the session being
   terminated, the endpoint MUST stop sending new datagrams and reset
   all of the streams associated with the session.








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6.  Transport Properties

   The WebTransport framework [OVERVIEW] defines a set of optional
   transport properties that clients can use to determine the presence
   of features which might allow additional optimizations beyond the
   common set of properties available via all WebTransport protocols.
   Below are details about support in Http2Transport for those
   properties.

   Stream Independence:  Http2Transport does not support stream
      independence, as HTTP/2 inherently has head of line blocking.

   Partial Reliability:  Http2Transport does not support partial
      reliability, as HTTP/2 retransmits any lost data.  This means that
      any datagrams sent via Http2Transport will be retransmitted
      regardless of the preference of the application.  The receiver is
      permitted to drop them, however, if it is unable to buffer them.

   Pooling Support:  Http2Transport supports pooling, as multiple
      transports using Http2Transport may share the same underlying
      HTTP/2 connection and therefore share a congestion controller and
      other transport context.

   Connection Mobility:  Http2Transport does not support connection
      mobility, unless an underlying transport protocol that supports
      multipath or migration, such as MPTCP [RFC7540], is used
      underneath HTTP/2 and TLS.  Without such support, Http2Transport
      connections cannot survive network transitions.

7.  Security Considerations

   WebTransport over HTTP/2 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.

   WebTransport over HTTP/2 requires explicit opt-in through the use of
   HTTP SETTINGS; this avoids potential protocol confusion attacks by
   ensuring the HTTP/2 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/2, 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



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   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.  HTTP/2 SETTINGS Parameter Registration

   The following entry is added to the "HTTP/2 Settings" registry
   established by [RFC7540]:

   The "SETTINGS_ENABLE_WEBTRANSPORT" parameter indicates that the
   specified HTTP/2 connection is WebTransport-capable.

   Setting Name:  ENABLE_WEBTRANSPORT

   Value:  0x2b603742

   Default:  0

   Specification:  This document

8.2.  Frame Type Registration

   The following entries are added to the "HTTP/2 Frame Type" registry
   established by [RFC7540]:

   The "WT_STREAM" frame allows HTTP/2 client- and server-initiated
   unidirectional and bidirectional streams to be used by WebTransport:

   Code:  0xTBD

   Frame Type:  WEBTRANSPORT_STREAM

   Specification:  This document

   The "WT_RST_STREAM" frame allows HTTP/2 WebTransport streams to be
   unidirectionally reset for writing:

   Code:  0xTBD

   Frame Type:  WEBTRANSPORT_RST_STREAM

   Specification:  This document




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   The "WT_STOP_SENDING" frame allows HTTP/2 WebTransport streams to be
   unidirectionally reset for reading:

   Code:  0xTBD

   Frame Type:  WEBTRANSPORT_STOP_SENDING

   Specification:  This document

   The "WT_DATAGRAM" frame allows HTTP/2 client and server to exchange
   datagrams used by WebTransport:

   Code:  0xTBD

   Frame Type:  WEBTRANSPORT_DATAGRAM

   Specification:  This document

8.3.  HTTP/2 Error Code Registry

   The following entries are added to the "HTTP/2 Error Code" registry
   that was established by Section 11.2 of [RFC7540].

   Name:  WT_STREAM_ERROR

   Code:  0xTBD

   Description:  Invalid use of WT_STREAM frame

   Specification:  _RFC Editor: Please fill in this value with the RFC
      number for this document_

8.4.  Examples

   An example of negotiating a WebTransport Stream on an HTTP/2
   connection follows.  This example is intended to closely follow the
   example in Section 5.1 of [RFC8441] to help illustrate the
   differences defined in this document.













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   [[ From Client ]]                   [[ From Server ]]

   SETTINGS
   SETTINGS_ENABLE_WEBTRANSPORT = 1

                                       SETTINGS
                                       SETTINGS_ENABLE_WEBTRANSPORT = 1

   HEADERS + END_HEADERS
   Stream ID = 3
   :method = CONNECT
   :protocol = webtransport
   :scheme = https
   :path = /
   :authority = server.example.com
   origin: server.example.com

                                       HEADERS + END_HEADERS
                                       Stream ID = 3
                                       :status = 200

   WT_STREAM
   Stream ID = 5
   Session ID = 3


   DATA
   Stream ID = 5
   WebTransport Data

                                       DATA + END_STREAM
                                       Stream ID = 5
                                       WebTransport Data

   DATA + END_STREAM
   Stream ID = 5
   WebTransport Data

   An example of the server initiating a WebTransport Stream follows.
   The only difference here is the endpoint that sends the first
   WT_STREAM frame.










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   [[ From Client ]]                   [[ From Server ]]

   SETTINGS
   SETTINGS_ENABLE_WEBTRANSPORT = 1

                                       SETTINGS
                                       SETTINGS_ENABLE_WEBTRANSPORT = 1

   HEADERS + END_HEADERS
   Stream ID = 3
   :method = CONNECT
   :protocol = webtransport
   :scheme = https
   :path = /
   :authority = server.example.com
   origin: server.example.com
                                       HEADERS + END_HEADERS
                                       Stream ID = 3
                                       :status = 200

                                       WT_STREAM
                                       Stream ID = 2
                                       Session ID = 3

                                       DATA
                                       Stream ID = 2
                                       WebTransport Data

   DATA + END_STREAM
   Stream ID = 2
   WebTransport Data

                                       DATA + END_STREAM
                                       Stream ID = 2
                                       WebTransport Data

9.  References

9.1.  Normative References

   [HTTP]     Fielding, R. T., Nottingham, M., and J. Reschke, "HTTP
              Semantics", Work in Progress, Internet-Draft, draft-ietf-
              httpbis-semantics-17, 25 July 2021,
              <https://datatracker.ietf.org/doc/html/draft-ietf-httpbis-
              semantics-17>.






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   [OVERVIEW] Vasiliev, V., "The WebTransport Protocol Framework", Work
              in Progress, Internet-Draft, draft-ietf-webtrans-overview-
              02, 28 July 2021, <https://datatracker.ietf.org/doc/html/
              draft-ietf-webtrans-overview-02>.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <https://www.rfc-editor.org/rfc/rfc2119>.

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              DOI 10.17487/RFC6454, December 2011,
              <https://www.rfc-editor.org/rfc/rfc6454>.

   [RFC6585]  Nottingham, M. and R. Fielding, "Additional HTTP Status
              Codes", RFC 6585, DOI 10.17487/RFC6585, April 2012,
              <https://www.rfc-editor.org/rfc/rfc6585>.

   [RFC7230]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Message Syntax and Routing",
              RFC 7230, DOI 10.17487/RFC7230, June 2014,
              <https://www.rfc-editor.org/rfc/rfc7230>.

   [RFC7231]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
              DOI 10.17487/RFC7231, June 2014,
              <https://www.rfc-editor.org/rfc/rfc7231>.

   [RFC7540]  Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
              Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
              DOI 10.17487/RFC7540, May 2015,
              <https://www.rfc-editor.org/rfc/rfc7540>.

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <https://www.rfc-editor.org/rfc/rfc8174>.

   [RFC8441]  McManus, P., "Bootstrapping WebSockets with HTTP/2",
              RFC 8441, DOI 10.17487/RFC8441, September 2018,
              <https://www.rfc-editor.org/rfc/rfc8441>.

   [WEBTRANSPORT-H3]
              Vasiliev, V., "WebTransport over HTTP/3", Work in
              Progress, Internet-Draft, draft-ietf-webtrans-http3-01, 19
              May 2021, <https://datatracker.ietf.org/doc/html/draft-
              ietf-webtrans-http3-01>.

9.2.  Informative References



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   [RFC6202]  Loreto, S., Saint-Andre, P., Salsano, S., and G. Wilkins,
              "Known Issues and Best Practices for the Use of Long
              Polling and Streaming in Bidirectional HTTP", RFC 6202,
              DOI 10.17487/RFC6202, April 2011,
              <https://www.rfc-editor.org/rfc/rfc6202>.

Acknowledgments

   Thanks to Anthony Chivetta, Joshua Otto, and Valentin Pistol for
   their contributions in the design and implementation of this work.

Authors' Addresses

   Alan Frindell
   Facebook Inc.

   Email: afrind@fb.com


   Eric Kinnear
   Apple Inc.
   One Apple Park Way
   Cupertino, California 95014,
   United States of America

   Email: ekinnear@apple.com


   Tommy Pauly
   Apple Inc.
   One Apple Park Way
   Cupertino, California 95014,
   United States of America

   Email: tpauly@apple.com


   Victor Vasiliev
   Google

   Email: vasilvv@google.com


   Guowu Xie
   Facebook Inc.

   Email: woo@fb.com




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