Network Working Group                                        A. Frindell
Internet-Draft                                                  Facebook
Intended status: Standards Track                              E. Kinnear
Expires: 4 September 2025                                     Apple Inc.
                                                             V. Vasiliev
                                                                  Google
                                                            3 March 2025


                        WebTransport over HTTP/3
                      draft-ietf-webtrans-http3-12

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/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
   (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-ietf-webtrans-http3/
   issues>.  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/.

   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 4 September 2025.



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Copyright Notice

   Copyright (c) 2025 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 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.  Overview  . . . . . . . . . . . . . . . . . . . . . . . . . .   3
     2.1.  QUIC, WebTransport, and HTTP/3  . . . . . . . . . . . . .   4
       2.1.1.  Minimizing Implementation Complexity  . . . . . . . .   5
     2.2.  Protocol Overview . . . . . . . . . . . . . . . . . . . .   5
   3.  Session Establishment . . . . . . . . . . . . . . . . . . . .   6
     3.1.  Establishing a WebTransport-Capable HTTP/3 Connection . .   6
     3.2.  Extended CONNECT in HTTP/3  . . . . . . . . . . . . . . .   7
     3.3.  Creating a New Session  . . . . . . . . . . . . . . . . .   7
     3.4.  Application Protocol Negotiation  . . . . . . . . . . . .   8
     3.5.  Prioritization  . . . . . . . . . . . . . . . . . . . . .   9
   4.  WebTransport Features . . . . . . . . . . . . . . . . . . . .   9
     4.1.  Unidirectional streams  . . . . . . . . . . . . . . . . .  10
     4.2.  Bidirectional Streams . . . . . . . . . . . . . . . . . .  10
     4.3.  Resetting Data Streams  . . . . . . . . . . . . . . . . .  11
     4.4.  Datagrams . . . . . . . . . . . . . . . . . . . . . . . .  12
     4.5.  Buffering Incoming Streams and Datagrams  . . . . . . . .  13
     4.6.  Interaction with HTTP/3 GOAWAY frame  . . . . . . . . . .  13
     4.7.  Use of Keying Material Exporters  . . . . . . . . . . . .  14
   5.  Flow Control  . . . . . . . . . . . . . . . . . . . . . . . .  14
     5.1.  Limiting the Number of Simultaneous Sessions  . . . . . .  15
     5.2.  Limiting the Number of Streams Within a Session . . . . .  15
     5.3.  Data Limits . . . . . . . . . . . . . . . . . . . . . . .  16
     5.4.  Flow Control and Intermediaries . . . . . . . . . . . . .  16
     5.5.  Flow Control SETTINGS . . . . . . . . . . . . . . . . . .  17
     5.6.  Flow Control Capsules . . . . . . . . . . . . . . . . . .  17
       5.6.1.  WT_MAX_STREAMS Capsule  . . . . . . . . . . . . . . .  17
     5.7.  WT_STREAMS_BLOCKED Capsule  . . . . . . . . . . . . . . .  18
     5.8.  WT_MAX_DATA Capsule . . . . . . . . . . . . . . . . . . .  19
     5.9.  WT_DATA_BLOCKED Capsule . . . . . . . . . . . . . . . . .  20
   6.  Session Termination . . . . . . . . . . . . . . . . . . . . .  20



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   7.  Considerations for Future Versions  . . . . . . . . . . . . .  22
     7.1.  Negotiating the Draft Version . . . . . . . . . . . . . .  22
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  22
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  23
     9.1.  Upgrade Token Registration  . . . . . . . . . . . . . . .  23
     9.2.  HTTP/3 SETTINGS Parameter Registration  . . . . . . . . .  23
     9.3.  Frame Type Registration . . . . . . . . . . . . . . . . .  25
     9.4.  Stream Type Registration  . . . . . . . . . . . . . . . .  25
     9.5.  HTTP/3 Error Code Registration  . . . . . . . . . . . . .  26
     9.6.  Capsule Types . . . . . . . . . . . . . . . . . . . . . .  26
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  28
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  28
     10.2.  Informative References . . . . . . . . . . . . . . . . .  30
   Appendix A.  Changelog  . . . . . . . . . . . . . . . . . . . . .  30
     A.1.  Changes between draft versions 02 and 07  . . . . . . . .  30
   Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .  31
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  32

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",
   "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/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.  Overview







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2.1.  QUIC, WebTransport, and HTTP/3

   QUIC version 1 [RFC9000] is a secure transport protocol with flow
   control and congestion control.  QUIC supports application data
   exchange via streams; reliable and ordered byte streams that can be
   multiplexed.  Stream independence can mitigate head-of-line blocking.
   While QUIC provides streams as a transport service, it is
   unopinionated about their usage.  The applicability of streams is
   described by section 4 of [RFC9308].

   HTTP is an application-layer protocol, defined by "HTTP Semantics"
   [RFC9110].  HTTP/3 is the application mapping for QUIC, defined in
   [RFC9114].  It describes how QUIC streams are used to carry control
   data or HTTP request and response message sequences in the form of
   frames and describes details of stream and connection lifecycle
   management.  HTTP/3 offers two features in addition to HTTP
   Semantics: QPACK header compression [RFC9208] and Server Push
   Section 4.6 of [RFC9114].

   WebTransport session establishment involves interacting at the HTTP
   layer with a resource.  For Web user agents, this interaction is
   important for security reasons, especially to ensure that the
   resource is willing to use WebTransport.

   Although WebTransport requires HTTP for its handshake, when HTTP/3 is
   in use, HTTP is not used for anything else related to an established
   session.  Instead, QUIC streams begin with a header sequence of bytes
   that links them to the established session.  The remainder of the
   stream is the body, which carries the payload supplied by the
   application using WebTransport.  This process is similar to
   WebSockets over HTTP/1.1 [ORIGIN], where access to the underlying
   byte stream is enabled after both sides have completed the handshake.

   The layering of QUIC, HTTP/3, and WebTransport is shown in Figure 1.
   Once a WebTransport session is established, applications have nearly
   direct access to QUIC.

   ,--------------------------------,
   |            WebTransport        |
   ,----------------,---------------,
   | HTTP Semantics |               |
   |      and       |               |
   | Session Setup  | Nearly direct |
   ,----------------,               |
   |     HTTP/3     |               |
   ,----------------`---------------,
   |               QUIC             |
   `--------------------------------'



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                      Figure 1: WebTransport Layering

2.1.1.  Minimizing Implementation Complexity

   WebTransport has minimal interaction with HTTP and HTTP/3.  Clients
   or servers can constrain their use of features to only those required
   to complete a WebTransport handshake:

   *  Generating/parsing the request method, host, path, protocol,
      optional Origin header, and perhaps some extra headers.

   *  Generating/parsing the response status code, and possibly some
      extra headers.

   The receiver can likely perform several of its HTTP-level
   requirements using bytestring comparisons.

   While HTTP/3 encodes HTTP messages using QPACK, the complexity can be
   minimized.  Receivers can disable dynamic decompression entirely but
   must always support static decompression and Huffman decoding.
   Senders can opt to never use dynamic compression, static compression,
   or Huffman encoding.

2.2.  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
   correspondingly).

   When an HTTP/3 connection is established, the server sends a
   SETTINGS_WEBTRANSPORT_MAX_SESSIONS setting in order to indicate
   support for WebTransport over HTTP/3.  This process also negotiates
   the use of additional HTTP/3 extensions.

   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, a 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 and transfer its
      ownership to WebTransport by providing a special signal in the
      first bytes.

   *  A server can create a bidirectional stream and transfer its
      ownership to WebTransport by providing a special signal in the
      first bytes.

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

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

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

3.  Session Establishment

3.1.  Establishing a WebTransport-Capable HTTP/3 Connection

   In order to indicate support for WebTransport, the server MUST send a
   SETTINGS_WEBTRANSPORT_MAX_SESSIONS value greater than "0" in its
   SETTINGS frame.  The default value for the
   SETTINGS_WEBTRANSPORT_MAX_SESSIONS parameter is "0", meaning that the
   endpoint is not willing to receive any WebTransport sessions.  Note
   that the client does not need to send any value to indicate support
   for WebTransport; clients indicate support for WebTransport by using
   the "webtransport" upgrade token in CONNECT requests establishing
   WebTransport sessions (see Section 9.1).

   The client MUST NOT send a WebTransport request until it has received
   the setting indicating WebTransport support from the server.

   [[RFC editor: please remove the following paragraph before
   publication.]]

   For draft verisons of WebTransport only, 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 the
   WebTransport extension that is different from the one used by the
   server.










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   Because WebTransport over HTTP/3 requires support for HTTP/3
   datagrams and the Capsule Protocol, both the client and the server
   MUST indicate support for HTTP/3 datagrams by sending a
   SETTINGS_H3_DATAGRAM value set to 1 in their SETTINGS frame (see
   Section 2.1.1 of [HTTP-DATAGRAM]).  Servers should also note that
   CONNECT requests to establish new WebTransport sessions, in addition
   to other messages, may arrive before this SETTING is received (see
   Section 4.5).

   WebTransport over HTTP/3 also requires support for QUIC datagrams.
   To indicate support, both the client and the server MUST send a
   max_datagram_frame_size transport parameter with a value greater than
   0 (see Section 3 of [QUIC-DATAGRAM]).

   Any WebTransport requests sent by the client without enabling QUIC
   and HTTP datagrams MUST be treated as malformed by the server, as
   described in Section 4.1.2 of [HTTP3].

   WebTransport over HTTP/3 relies on the RESET_STREAM_AT frame defined
   in [RESET-STREAM-AT].  To indicate support, both the client and the
   server MUST enable the extension as described in Section 3 of
   [RESET-STREAM-AT].

3.2.  Extended CONNECT in HTTP/3

   [RFC8441] defines an extended CONNECT method in Section 4, enabled by
   the SETTINGS_ENABLE_CONNECT_PROTOCOL setting.  That setting is
   defined for HTTP/3 by [RFC9220].  A server supporting WebTransport
   over HTTP/3 MUST send both the SETTINGS_WEBTRANSPORT_MAX_SESSIONS
   setting with a value greater than "0" and the
   SETTINGS_ENABLE_CONNECT_PROTOCOL setting with a value of "1".

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).

   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.  If the WebTransport session is
   coming from a browser client, an Origin header [RFC6454] MUST be
   provided within the request; otherwise, the header is OPTIONAL.

   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



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   it does not, it SHOULD reply with status code 404 (Section 15.5.5 of
   [HTTP]).  When the request contains the Origin header, the
   WebTransport server MUST verify the Origin header to ensure that the
   specified origin is allowed to access the server in question.  If the
   verification fails, the WebTransport server SHOULD reply with status
   code 403 (Section 15.5.4 of [HTTP]).  If all checks pass, the
   WebTransport server MAY accept the session by replying with a 2xx
   series status code, as defined in Section 15.3 of [HTTP].

   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.

   The server may reply with a 3xx response, indicating a redirection
   (Section 15.4 of [HTTP]).  The user agent MUST NOT automatically
   follow such redirects, as the client could potentially already have
   sent data for the WebTransport session in question; it MAY notify the
   client about the redirect.

   Clients cannot initiate WebTransport in 0-RTT packets, as the CONNECT
   method is not considered safe (see Section 10.9 of [HTTP3]).
   However, WebTransport-related SETTINGS parameters may be retained
   from the previous session as described in Section 7.2.4.2 of [HTTP3].
   If the server accepts 0-RTT, the server MUST NOT reduce the limit of
   maximum open WebTransport sessions from the one negotiated during the
   previous session; such change would be deemed incompatible, and MUST
   result in a H3_SETTINGS_ERROR connection error.

   The webtransport HTTP Upgrade Token uses the Capsule Protocol as
   defined in [HTTP-DATAGRAM].  The Capsule Protocol is negotiated when
   the server sends a 2xx response.  The capsule-protocol header field
   Section 3.4 of [HTTP-DATAGRAM] is not required by WebTransport and
   can safely be ignored by WebTransport endpoints.

3.4.  Application Protocol Negotiation

   WebTransport over HTTP/3 offers a protocol negotiation mechanism,
   similar to TLS Application-Layer Protocol Negotiation Extension
   (ALPN) [RFC7301]; the intent is to simplify porting pre-existing
   protocols that use QUIC and rely on this functionality.

   The user agent MAY include a WT-Available-Protocols header field in
   the CONNECT request.  The WT-Available-Protocols enumerates the
   possible protocols in preference order.  If the server receives such
   a header, it MAY include a WT-Protocol field in a successful (2xx)
   response.  If it does, the server SHALL include a single choice from
   the client's list in that field.  Servers MAY reject the request if
   the client did not include a suitable protocol.



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   Both WT-Available-Protocols and WT-Protocol are Structured Fields
   [RFC8941].  WT-Available-Protocols is a List of Tokens, and WT-
   Protocol is a Token.  The token in the WT-Protocol response header
   field MUST be one of the tokens listed in WT-Available-Protocols of
   the request.  The semantics of individual token values is determined
   by the WebTransport resource in question and are not registered in
   IANA's "ALPN Protocol IDs" registry.

3.5.  Prioritization

   WebTransport sessions are initiated using extended CONNECT.  While
   Section 11 of [RFC9218] describes how extensible priorities can be
   applied to data sent on a CONNECT stream, WebTransport extends the
   types of data that are exchanged in relation to the request and
   response, which requires additional considerations.

   WebTransport CONNECT requests and responses MAY contain the Priority
   header field (Section 5 of [RFC9218]); clients MAY reprioritize by
   sending PRIORITY_UPDATE frames (Section 7 of [RFC9218]).  In
   extension to [RFC9218], it is RECOMMENDED that clients and servers
   apply the scheduling guidance in both Section 9 of [RFC9218] and
   Section 10 of [RFC9218] for all data that they send in the enclosing
   WebTransport session, including Capsules, WebTransport streams and
   datagrams.  WebTransport does not provide any priority signaling
   mechanism for streams and datagrams within a WebTransport session;
   such mechanisms can be defined by application protocols using
   WebTransport.  It is RECOMMENDED that such mechanisms only affect
   scheduling within a session and not scheduling of other data on the
   same HTTP/3 connection.

   The client/server priority merging guidance given in Section 8 of
   [RFC9218] also applies to WebTransport session.  For example, a
   client that receives a response Priority header field could alter its
   view of a WebTransport session priority and alter the scheduling of
   outgoing data as a result.

   Endpoints that prioritize WebTransport sessions need to consider how
   they interact with other sessions or requests on the same HTTP/3
   connection.

4.  WebTransport Features

   WebTransport over HTTP/3 provides the following features described in
   [OVERVIEW]: unidirectional streams, bidirectional streams and
   datagrams, initiated by either endpoint.  Protocols designed for use
   with WebTransport over HTTP/3 are constrained to these features.  The
   Capsule Protocol is an implementation detail of WebTransport over
   HTTP/3 and is not a WebTransport feature.



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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
   [RFC9000].

   The client MAY optimistically open unidirectional and bidirectional
   streams, as well as send datagrams, for a session that it has sent
   the CONNECT request for, even if it has not yet received the server's
   response to the request.  On the server side, opening streams and
   sending datagrams is possible as soon as the CONNECT request has been
   received.

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

4.1.  Unidirectional streams

   WebTransport endpoints can initiate 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 2).

   Unidirectional Stream {
       Stream Type (i) = 0x54,
       Session ID (i),
       Stream Body (..)
   }

            Figure 2: Unidirectional WebTransport stream format

4.2.  Bidirectional Streams

   All client-initiated bidirectional streams are reserved by HTTP/3 as
   request streams, which are a sequence of HTTP/3 frames with a variety
   of rules (see Sections 4.1 and 6.1 of [HTTP3]).

   WebTransport extends HTTP/3 to allow clients to declare and use
   alternative request stream rules.  Once a client receives settings
   indicating WebTransport support (Section 3.1), it can send a special
   signal value, encoded as a variable-length integer, as the first
   bytes of the stream in order to indicate how the remaining bytes on
   the stream are used.

   WebTransport extends HTTP/3 by defining rules for all server-
   initiated bidirectional streams.  Once a server receives an incoming
   CONNECT request establishing a WebTransport session (Section 3.1), it



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   can open a bidirectional stream for use with that session and SHALL
   send a special signal value, encoded as a variable-length integer, as
   the first bytes of the stream in order to indicate how the remaining
   bytes on the stream are used.

   The signal value, 0x41, is used by clients and servers to open a
   bidirectional WebTransport stream.  Following this is the associated
   session ID, encoded as a variable-length integer; the rest of the
   stream is the application payload of the WebTransport stream
   (Figure 3).

   Bidirectional Stream {
       Signal Value (i) = 0x41,
       Session ID (i),
       Stream Body (..)
   }

             Figure 3: Bidirectional WebTransport stream format

   This document reserves the special signal value 0x41 as a
   WEBTRANSPORT_STREAM frame type.  While it is registered as an HTTP/3
   frame type to avoid collisions, WEBTRANSPORT_STREAM is not a proper
   HTTP/3 frame, as it lacks length; it is an extension of HTTP/3 frame
   syntax that MUST be supported by any peer negotiating WebTransport.
   Endpoints that implement this extension are also subject to
   additional frame handling requirements.  Endpoints MUST NOT send
   WEBTRANSPORT_STREAM as a frame type on HTTP/3 streams other than the
   very first bytes of a request stream.  Receiving this frame type in
   any other circumstances MUST be treated as a connection error of type
   H3_FRAME_ERROR.

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 32-bit integer, assuming values between 0x00000000 and
   0xffffffff.  WebTransport implementations SHALL remap those error
   codes into the error range reserved for
   WEBTRANSPORT_APPLICATION_ERROR, where 0x00000000 corresponds to
   0x52e4a40fa8db, and 0xffffffff corresponds to 0x52e5ac983162.  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
   skipped when mapping the error codes (i.e. the two HTTP/3 error



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   codepoints adjacent to a reserved codepoint would map to two adjacent
   WebTransport application error codepoints).  An example pseudocode
   can be seen in Figure 4.

       first = 0x52e4a40fa8db
       last = 0x52e5ac983162

       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 - floor(shifted / 0x1f)

          Figure 4: Pseudocode for converting between WebTransport
                 application errors and HTTP/3 error codes

   WebTransport data streams are associated with sessions through a
   header at the beginning of the stream; resetting a stream might
   result in that data being discarded when using a RESET_STREAM frame.
   To prevent this, WebTransport implementations MUST use the
   RESET_STREAM_AT frame [RESET-STREAM-AT] with a Reliable Size set to
   at least the size of the WebTransport header when resetting a
   WebTransport data stream.  This ensures that the ID field associating
   the data stream with a WebTransport session is always delivered.

   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 (Section 2.1 of [HTTP-DATAGRAM]).  Note
   that the payload field directly follows the Quarter Stream ID field,
   which is at the start of the QUIC DATAGRAM frame payload and refers
   to the CONNECT stream that established the WebTransport session.






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4.5.  Buffering Incoming Streams and Datagrams

   In WebTransport over HTTP/3, the client MUST wait for receipt of the
   server's SETTINGS frame before establishing any WebTransport sessions
   by sending CONNECT requests using the WebTransport upgrade token (see
   Section 3.1).  This ensures that the client will always know what
   versions of WebTransport can be used on a given HTTP/3 connection.

   Clients can, however, send a SETTINGS frame, 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/
   or STOP_SENDING with the WEBTRANSPORT_BUFFERED_STREAM_REJECTED 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.

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.

   A client receiving GOAWAY cannot initiate CONNECT requests for new
   WebTransport sessions on that HTTP/3 connection; it must open a new
   HTTP/3 connection to initiate new WebTransport sessions with the same
   peer.

   An HTTP/3 GOAWAY frame is also a signal to applications to initiate
   shutdown for all WebTransport sessions.  To shut down a single
   WebTransport session, either endpoint can send a
   DRAIN_WEBTRANSPORT_SESSION (0x78ae) capsule.

   DRAIN_WEBTRANSPORT_SESSION Capsule {
     Type (i) = DRAIN_WEBTRANSPORT_SESSION,
     Length (i) = 0
   }




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   After sending or receiving either a DRAIN_WEBTRANSPORT_SESSION
   capsule or a HTTP/3 GOAWAY frame, an endpoint MAY continue using the
   session and MAY open new streams.  The signal is intended for the
   application using WebTransport, which is expected to attempt to
   gracefully terminate the session as soon as possible.

4.7.  Use of Keying Material Exporters

   WebTransport over HTTP/3 supports the use of TLS keying material
   exporters [RFC8446], Section 7.5.  Since the underlying QUIC
   connection may be shared by multiple WebTransport sessions,
   WebTransport defines its own mechanism for deriving a TLS exporter
   that separates keying material for different sessions.  If the user
   requests an exporter for a given WebTransport session with a
   specified label and context, the resulting exporter SHALL be a TLS
   exporter as defined in[RFC8446], Section 7.5 with the label set to
   "EXPORTER-WebTransport" and the context set to the serialization of
   the "WebTransport Exporter Context" struct as defined below.

   WebTransport Exporter Context {
     WebTransport Session ID (64),
     WebTransport Application-Supplied Exporter Label Length (8),
     WebTransport Application-Supplied Exporter Label (8..),
     WebTransport Application-Supplied Exporter Context Length (8),
     WebTransport Application-Supplied Exporter Context (..)
   }

   A TLS exporter API might permit the context field to be omitted.  In
   this case, as with TLS 1.3, the WebTransport Application-Supplied
   Exporter Context becomes zero-length if omitted.

5.  Flow Control

   Flow control governs the amount of resources that can be consumed or
   data that can be sent.  When using WebTransport over HTTP/3,
   endpoints can limit the number of sessions that a peer can create on
   a single HTTP/3 connection and the number of streams that a peer can
   create within a session.  Endpoints can also limit the amount of data
   that can be consumed by each session and by each stream within a
   session.

   WebTransport over HTTP/3 provides connection-level limit that governs
   the number of sessions that can be created on an HTTP/3 connection;
   see Section 5.1.  It also provides the session-level limits that
   govern the number of streams that can be created in a session and
   limit the amount of data that can be exchanged across all streams in
   each session; see Section 5.2.




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   The underlying QUIC connection provides connection and stream level
   flow control.  The connection data limit defines the total amount of
   data that can be sent across all WebTransport sessions and other non-
   WebTransport streams.  A stream's data limit controls the amount of
   data that can be sent on that stream, WebTransport or otherwise; see
   Section 4 of [RFC9000].

5.1.  Limiting the Number of Simultaneous Sessions

   This document defines a SETTINGS_WEBTRANSPORT_MAX_SESSIONS parameter
   that allows the server to limit the maximum number of concurrent
   WebTransport sessions on a single HTTP/3 connection.  The client MUST
   NOT open more simultaneous sessions than indicated in the server
   SETTINGS parameter.  The server MUST NOT close the connection if the
   client opens sessions exceeding this limit, as the client and the
   server do not have a consistent view of how many sessions are open
   due to the asynchronous nature of the protocol; instead, it MUST
   reset all of the CONNECT streams it is not willing to process with
   the HTTP_REQUEST_REJECTED status defined in [HTTP3].

5.2.  Limiting the Number of Streams Within a Session

   The WT_MAX_STREAMS capsule (Section 5.6.1) establishes a limit on the
   number of streams within a WebTransport session.  Like the QUIC
   MAX_STREAMS frame (Section 19.11 of [RFC9000]), this capsule has two
   types that provide separate limits for unidirectional and
   bidirectional streams that are initiated by a peer.

   Note that the CONNECT stream for the session is not included in
   either the bidirectional or the unidirectional stream limits; the
   number of CONNECT streams a client can open is limited by the
   SETTINGS_WEBTRANSPORT_MAX_SESSIONS setting and QUIC flow control's
   stream limits.

   The session-level stream limit applies in addition to the QUIC
   MAX_STREAMS frame, which provides a connection-level stream limit.
   New streams can only be created within the session if both the
   stream- and the connection-level limit permit; see Section 4.6 of
   [RFC9000] for details on how QUIC stream limits are applied.

   Unlike the the QUIC MAX_STREAMS frame, there is no simple
   relationship between the value in this frame and stream IDs in QUIC
   STREAM frames.  This especially applies if there are other users of
   streams on the connection.

   The WT_STREAMS_BLOCKED capsule (Section 5.7) can be sent to indicate
   that an endpoint was unable to create a stream due to the session-
   level stream limit.



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   Note that enforcing this limit requires reliable resets for stream
   headers so that both endpoints can agree on the number of streams
   that are open.

5.3.  Data Limits

   The WT_MAX_DATA capsule (Section 5.8) establishes a limit on the
   amount of data that can be sent within a WebTransport session.  This
   limit counts all data that is sent on streams of the corresponding
   type, excluding the stream header (see Section 4.1 and Section 4.2).
   The stream header is excluded from this limit so that this limit does
   not prevent the sending of information that is essential in linking
   new streams to a specific WebTransport session.

   Implementing WT_MAX_DATA requires that the QUIC stack provide the
   WebTransport implementation with information about the final size of
   streams; see { {Section 4.5 of !RFC9000}}.  This allows both
   endpoints to agree on how much data was consumed by that stream,
   although the stream header exclusion above applies.

   The WT_DATA_BLOCKED capsule (Section 5.9) can be sent to indicate
   that an endpoint was unable to send data due to a limit set by the
   WT_MAX_DATA capsule.

   Because WebTransport over HTTP/3 uses a native QUIC stream for each
   WebTransport stream, per-stream data limits are provided by QUIC
   natively; see Section 4.1 of [RFC9000].  The WT_MAX_STREAM_DATA and
   WT_STREAM_DATA_BLOCKED capsules (Part XX of
   [I-D.ietf-webtrans-http2]) are not used and so are prohibited.
   Endpoints MUST treat receipt of a WT_MAX_STREAM_DATA or a
   WT_STREAM_DATA_BLOCKED capsule as a session error.

5.4.  Flow Control and Intermediaries

   WebTransport over HTTP/3 uses several capsules for flow control, and
   all of these capsules define special intermediary handling as
   described in Section 3.2 of [HTTP-DATAGRAM].  These capsules,
   referred to as the "flow control capsules" are WT_MAX_DATA,
   WT_MAX_STREAMS, WT_DATA_BLOCKED, and WT_STREAMS_BLOCKED.

   Because flow control in WebTransport is hop-by-hop and does not
   provide an end-to-end signal, intermediaries MUST consume flow
   control signals and express their own flow control limits to the next
   hop.  The intermediary can send these signals via HTTP/3 flow control
   messages, HTTP/2 flow control messages, or as WebTransport flow
   control capsules, where appropriate.  Intermediaries are responsible
   for storing any data for which they advertise flow control credit if
   that data cannot be immediately forwarded to the next hop.



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   In practice, an intermediary that translates flow control signals
   between similar WebTransport protocols, such as between two HTTP/3
   connections, can often simply reexpress the same limits received on
   one connection directly on the other connection.

   An intermediary that does not want to be responsible for storing data
   that cannot be immediately sent on its translated connection can
   ensure that it does not advertise a higher flow control limit on one
   connection than the corresponding limit on the translated connection.

5.5.  Flow Control SETTINGS

   Initial flow control limits can be exchanged via HTTP/3 SETTINGS
   (Section 9.2) by providing non-zero values for

   *  WT_MAX_STREAMS via SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_UNI
      and SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_BIDI

   *  WT_MAX_DATA via SETTINGS_WEBTRANSPORT_INITIAL_MAX_DATA

5.6.  Flow Control Capsules

5.6.1.  WT_MAX_STREAMS Capsule

   An HTTP capsule [HTTP-DATAGRAM] called WT_MAX_STREAMS is introduced
   to inform the peer of the cumulative number of streams of a given
   type it is permitted to open.  A WT_MAX_STREAMS capsule with a type
   of 0x190B4D3F applies to bidirectional streams, and a WT_MAX_STREAMS
   capsule with a type of 0x190B4D40 applies to unidirectional streams.

   Note that, because Maximum Streams is a cumulative value representing
   the total allowed number of streams, including previously closed
   streams, endpoints repeatedly send new WT_MAX_STREAMS capsules with
   increasing Maximum Streams values as streams are opened.

   WT_MAX_STREAMS Capsule {
     Type (i) = 0x190B4D3F..0x190B4D40,
     Length (i),
     Maximum Streams (i),
   }

                  Figure 5: WT_MAX_STREAMS Capsule Format

   WT_MAX_STREAMS capsules contain the following field:

   Maximum Streams:  A count of the cumulative number of streams of the





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      corresponding type that can be opened over the lifetime of the
      session.  This value cannot exceed 2^60, as it is not possible to
      encode stream IDs larger than 2^62-1.

   An endpoint MUST NOT open more streams than permitted by the current
   stream limit set by its peer.  For instance, a server that receives a
   unidirectional stream limit of 3 is permitted to open streams 3, 7,
   and 11, but not stream 15.

   Note that this limit includes streams that have been closed as well
   as those that are open.

   The WT_MAX_STREAMS capsule defines special intermediary handling, as
   described in Section 3.2 of [HTTP-DATAGRAM].  Intermedaries MUST
   consume WT_MAX_STREAMS capsules for flow control purposes and MUST
   generate and send appropriate flow control signals for their limits.

   Initial values for these limits MAY be communicated by sending non-
   zero values for SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_UNI and
   SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_BIDI.

5.7.  WT_STREAMS_BLOCKED Capsule

   A sender SHOULD send a WT_STREAMS_BLOCKED capsule (type=0x190B4D43 or
   0x190B4D44) when it wishes to open a stream but is unable to do so
   due to the maximum stream limit set by its peer.  A
   WT_STREAMS_BLOCKED capsule of type 0x190B4D43 is used to indicate
   reaching the bidirectional stream limit, and a STREAMS_BLOCKED
   capsule of type 0x190B4D44 is used to indicate reaching the
   unidirectional stream limit.

   A WT_STREAMS_BLOCKED capsule does not open the stream, but informs
   the peer that a new stream was needed and the stream limit prevented
   the creation of the stream.

   WT_STREAMS_BLOCKED Capsule {
     Type (i) = 0x190B4D43..0x190B4D44,
     Length (i),
     Maximum Streams (i),
   }

                Figure 6: WT_STREAMS_BLOCKED Capsule Format

   WT_STREAMS_BLOCKED capsules contain the following field:

   Maximum Streams:  A variable-length integer indicating the maximum





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      number of streams allowed at the time the capsule was sent.  This
      value cannot exceed 2^60, as it is not possible to encode stream
      IDs larger than 2^62-1.

   The WT_STREAMS_BLOCKED capsule defines special intermediary handling,
   as described in Section 3.2 of [HTTP-DATAGRAM].  Intermedaries MUST
   consume WT_STREAMS_BLOCKED capsules for flow control purposes and
   MUST generate and send appropriate flow control signals for their
   limits.

5.8.  WT_MAX_DATA Capsule

   An HTTP capsule [HTTP-DATAGRAM] called WT_MAX_DATA (type=0x190B4D3D)
   is introduced to inform the peer of the maximum amount of data that
   can be sent on the WebTransport session as a whole.

   This limit counts all data that is sent on streams of the
   corresponding type, excluding the stream header (see Section 4.1 and
   Section 4.2).  Implementing WT_MAX_DATA requires that the QUIC stack
   provide the WebTransport implementation with information about the
   final size of streams; see Section 4.5 of [RFC9000].

   WT_MAX_DATA Capsule {
     Type (i) = 0x190B4D3D,
     Length (i),
     Maximum Data (i),
   }

                    Figure 7: WT_MAX_DATA Capsule Format

   WT_MAX_DATA capsules contain the following field:

   Maximum Data:  A variable-length integer indicating the maximum
      amount of data that can be sent on the entire session, in units of
      bytes.

   All data sent in WT_STREAM capsules counts toward this limit.  The
   sum of the lengths of Stream Data fields in WT_STREAM capsules MUST
   NOT exceed the value advertised by a receiver.

   The WT_MAX_DATA capsule defines special intermediary handling, as
   described in Section 3.2 of [HTTP-DATAGRAM].  Intermedaries MUST
   consume WT_MAX_DATA capsules for flow control purposes and MUST
   generate and send appropriate flow control signals for their limits;
   see Section 5.4.

   The initial value for this limit MAY be communicated by sending a
   non-zero value for SETTINGS_WEBTRANSPORT_INITIAL_MAX_DATA.



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5.9.  WT_DATA_BLOCKED Capsule

   A sender SHOULD send a WT_DATA_BLOCKED capsule (type=0x190B4D41) when
   it wishes to send data but is unable to do so due to WebTransport
   session-level flow control.  WT_DATA_BLOCKED capsules can be used as
   input to tuning of flow control algorithms.

   WT_DATA_BLOCKED Capsule {
     Type (i) = 0x190B4D41,
     Length (i),
     Maximum Data (i),
   }

                  Figure 8: WT_DATA_BLOCKED Capsule Format

   WT_DATA_BLOCKED capsules contain the following field:

   Maximum Data:  A variable-length integer indicating the session-level
      limit at which blocking occurred.

   The WT_DATA_BLOCKED capsule defines special intermediary handling, as
   described in Section 3.2 of [HTTP-DATAGRAM].  Intermedaries MUST
   consume WT_DATA_BLOCKED capsules for flow control purposes and MUST
   generate and send appropriate flow control signals for their limits;
   see Section 5.4.

6.  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 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:





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   CLOSE_WEBTRANSPORT_SESSION Capsule {
     Type (i) = CLOSE_WEBTRANSPORT_SESSION,
     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 session.

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

   An endpoint that sends a CLOSE_WEBTRANSPORT_SESSION capsule MUST
   immediately send a FIN.  The endpoint MAY send a STOP_SENDING to
   indicate it is no longer reading from the CONNECT stream.  The
   recipient MUST either close or reset the stream in response.  If any
   additional stream data is received on the CONNECT stream after
   receiving a CLOSE_WEBTRANSPORT_SESSION capsule, the stream MUST be
   reset with code H3_MESSAGE_ERROR.

   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.

   In some scenarios, an endpoint might want to send a
   CLOSE_WEBTRANSPORT_SESSION with detailed close information and then
   immediately close the underlying QUIC connection.  If the endpoint
   were to do both of those simultaneously, the peer could potentially
   receive the CONNECTION_CLOSE before receiving the
   CLOSE_WEBTRANSPORT_SESSION, thus never receiving the application
   error data contained in the latter.  To avoid this, the endpoint
   SHOULD wait until all CONNECT streams have been closed by the peer
   before sending the CONNECTION_CLOSE; this gives
   CLOSE_WEBTRANSPORT_SESSION properties similar to that of the QUIC
   CONNECTION_CLOSE mechanism as a best-effort mechanism of delivering
   application close metadata.









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7.  Considerations for Future Versions

   Future versions of WebTransport that change the syntax of the CONNECT
   requests used to establish WebTransport sessions will need to modify
   the upgrade token used to identify WebTransport, allowing servers to
   offer multiple versions simultaneously (see Section 9.1).

   Servers that support future incompatible versions of WebTransport
   signal that support by changing the codepoint used for the
   SETTINGS_WEBTRANSPORT_MAX_SESSIONS parameter (see Section 9.2).
   Clients can select the associated upgrade token, if applicable, to
   use when establishing a new session, ensuring that servers will
   always know the syntax in use for every incoming request.

   Changes to future stream formats require changes to the
   Unidirectional Stream type (see Section 4.1) and Bidirectional Stream
   signal value (see Section 4.2) to allow recipients of incoming frames
   to determine the WebTransport version, and corresponding wire format,
   used for the session associated with that stream.

7.1.  Negotiating the Draft Version

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

   The wire format aspects of the protocol are negotiated by changing
   the codepoint used for the SETTINGS_WEBTRANSPORT_MAX_SESSIONS
   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.

8.  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.

   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



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   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.

   A client could attempt to exhaust resources by opening too many
   WebTransport sessions at once.  In cases when the client is
   untrusted, the user agent SHOULD limit the number of outgoing
   sessions the client can open.

9.  IANA Considerations

9.1.  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
   WebTransport:

   Value:  webtransport

   Description:  WebTransport over HTTP/3

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

9.2.  HTTP/3 SETTINGS Parameter Registration

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

   The SETTINGS_WEBTRANSPORT_MAX_SESSIONS parameter indicates that the
   specified HTTP/3 endpoint is WebTransport-capable and the number of
   concurrent sessions it is willing to receive.  The default value for
   the SETTINGS_WEBTRANSPORT_MAX_SESSIONS parameter is "0", meaning that
   the endpoint is not willing to receive any WebTransport sessions.

   Setting Name:  WEBTRANSPORT_MAX_SESSIONS

   Value:  0xc671706a

   Default:  0




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   Specification:  This document

   The SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_UNI parameter indicates
   the initial value for the unidirectional max stream limit, otherwise
   communicated by the WT_MAX_STREAMS capsule (see Section 5.6.1).  The
   default value for the SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_UNI
   parameter is "0", indicating that the endpoint needs to send
   WT_MAX_STREAMS capsules on each individual WebTransport session
   before its peer is allowed to create any unidirectional streams
   within that session.

   Note that this limit applies to all WebTransport sessions that use
   the HTTP/3 connection on which this SETTING is sent.

   Setting Name:  SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_UNI

   Value:  0x2b64

   Default:  0

   Specification:  This document

   The SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_BIDI parameter
   indicates the initial value for the bidirectional max stream limit,
   otherwise communicated by the WT_MAX_STREAMS capsule (see
   Section 5.6.1).  The default value for the
   SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_BIDI parameter is "0",
   indicating that the endpoint needs to send WT_MAX_STREAMS capsules on
   each individual WebTransport session before its peer is allowed to
   create any bidirectional streams within that session.

   Note that this limit applies to all WebTransport sessions that use
   the HTTP/3 connection on which this SETTING is sent.

   Setting Name:  SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_BIDI

   Value:  0x2b65

   Default:  0

   Specification:  This document










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   The SETTINGS_WEBTRANSPORT_INITIAL_MAX_DATA parameter indicates the
   initial value for the session data limit, otherwise communicated by
   the WT_MAX_DATA capsule (see Section 5.8).  The default value for the
   SETTINGS_WEBTRANSPORT_INITIAL_MAX_DATA parameter is "0", indicating
   that the endpoint needs to send a WT_MAX_DATA capsule within each
   session before its peer is allowed to send any stream data within
   that session.

   Note that this limit applies to all WebTransport sessions that use
   the HTTP/3 connection on which this SETTING is sent.

   Setting Name:  SETTINGS_WEBTRANSPORT_INITIAL_MAX_DATA

   Value:  0x2b61

   Default:  0

   Specification:  This document

9.3.  Frame Type Registration

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

   The WEBTRANSPORT_STREAM frame is reserved for the purpose of avoiding
   collision with WebTransport HTTP/3 extensions:

   Code:  0x41

   Frame Type:  WEBTRANSPORT_STREAM

   Specification:  This document

9.4.  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



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9.5.  HTTP/3 Error Code Registration

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

   Name:  WEBTRANSPORT_BUFFERED_STREAM_REJECTED

   Value:  0x3994bd84

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

   Specification:  This document.

   Name:  WEBTRANSPORT_SESSION_GONE

   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:

   Name:  WEBTRANSPORT_APPLICATION_ERROR

   Value:  0x52e4a40fa8db to 0x52e5ac983162 inclusive, with the
      exception of the codepoints of form 0x1f * N + 0x21.

   Description:  WebTransport application error codes.

   Specification:  This document.

9.6.  Capsule Types

   The following entries are added to the "HTTP Capsule Types" registry
   established by [HTTP-DATAGRAM]:

   The CLOSE_WEBTRANSPORT_SESSION capsule.

   Value:  0x2843
   Capsule Type:  CLOSE_WEBTRANSPORT_SESSION
   Status:  permanent
   Specification:  This document
   Change Controller:  IETF
   Contact:  WebTransport Working Group webtransport@ietf.org
      (mailto:webtransport@ietf.org)



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   Notes:  None

   The DRAIN_WEBTRANSPORT_SESSION capsule.

   Value:  0x78ae
   Capsule Type:  DRAIN_WEBTRANSPORT_SESSION
   Status:  provisional (when this document is approved this will become
      permanent)
   Specification:  This document
   Change Controller:  IETF
   Contact:  WebTransport Working Group webtransport@ietf.org
      (mailto:webtransport@ietf.org)
   Notes:  None

   The WT_MAX_STREAMS capsule.

   Value:  0x190B4D3F..0x190B4D40
   Capsule Type:  WT_MAX_STREAMS
   Status:  permanent
   Specification:  This document
   Change Controller:  IETF
   Contact:  WebTransport Working Group webtransport@ietf.org
      (mailto:webtransport@ietf.org)
   Notes:  None

   The WT_STREAMS_BLOCKED capsule.

   Value:  0x190B4D43..0x190B4D44
   Capsule Type:  WT_STREAMS_BLOCKED
   Status:  permanent
   Specification:  This document
   Change Controller:  IETF
   Contact:  WebTransport Working Group webtransport@ietf.org
      (mailto:webtransport@ietf.org)
   Notes:  None

   The WT_MAX_DATA capsule.

   Value:  0x190B4D3D
   Capsule Type:  WT_MAX_DATA
   Status:  permanent
   Specification:  This document
   Change Controller:  IETF
   Contact:  WebTransport Working Group webtransport@ietf.org
      (mailto:webtransport@ietf.org)
   Notes:  None

   The WT_DATA_BLOCKED capsule.



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   Value:  0x190B4D41
   Capsule Type:  WT_DATA_BLOCKED
   Status:  permanent
   Specification:  This document
   Change Controller:  IETF
   Contact:  WebTransport Working Group webtransport@ietf.org
      (mailto:webtransport@ietf.org)
   Notes:  None

10.  References

10.1.  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,
              <https://www.rfc-editor.org/rfc/rfc9110>.

   [HTTP-DATAGRAM]
              Schinazi, D. and L. Pardue, "HTTP Datagrams and the
              Capsule Protocol", RFC 9297, DOI 10.17487/RFC9297, August
              2022, <https://www.rfc-editor.org/rfc/rfc9297>.

   [HTTP3]    Bishop, M., Ed., "HTTP/3", RFC 9114, DOI 10.17487/RFC9114,
              June 2022, <https://www.rfc-editor.org/rfc/rfc9114>.

   [OVERVIEW] Vasiliev, V., "The WebTransport Protocol Framework", Work
              in Progress, Internet-Draft, draft-ietf-webtrans-overview-
              09, 25 February 2025,
              <https://datatracker.ietf.org/doc/html/draft-ietf-
              webtrans-overview-09>.

   [QUIC-DATAGRAM]
              Pauly, T., Kinnear, E., and D. Schinazi, "An Unreliable
              Datagram Extension to QUIC", RFC 9221,
              DOI 10.17487/RFC9221, March 2022,
              <https://www.rfc-editor.org/rfc/rfc9221>.

   [RESET-STREAM-AT]
              Seemann, M. and K. Oku, "QUIC Stream Resets with Partial
              Delivery", Work in Progress, Internet-Draft, draft-ietf-
              quic-reliable-stream-reset-06, 28 February 2024,
              <https://datatracker.ietf.org/doc/html/draft-ietf-quic-
              reliable-stream-reset-06>.







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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,
              <https://www.rfc-editor.org/rfc/rfc2119>.

   [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,
              <https://www.rfc-editor.org/rfc/rfc3986>.

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

   [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>.

   [RFC8446]  Rescorla, E., "The Transport Layer Security (TLS) Protocol
              Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
              <https://www.rfc-editor.org/rfc/rfc8446>.

   [RFC8941]  Nottingham, M. and P. Kamp, "Structured Field Values for
              HTTP", RFC 8941, DOI 10.17487/RFC8941, February 2021,
              <https://www.rfc-editor.org/rfc/rfc8941>.

   [RFC9000]  Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
              Multiplexed and Secure Transport", RFC 9000,
              DOI 10.17487/RFC9000, May 2021,
              <https://www.rfc-editor.org/rfc/rfc9000>.

   [RFC9110]  Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,
              Ed., "HTTP Semantics", STD 97, RFC 9110,
              DOI 10.17487/RFC9110, June 2022,
              <https://www.rfc-editor.org/rfc/rfc9110>.

   [RFC9114]  Bishop, M., Ed., "HTTP/3", RFC 9114, DOI 10.17487/RFC9114,
              June 2022, <https://www.rfc-editor.org/rfc/rfc9114>.

   [RFC9218]  Oku, K. and L. Pardue, "Extensible Prioritization Scheme
              for HTTP", RFC 9218, DOI 10.17487/RFC9218, June 2022,
              <https://www.rfc-editor.org/rfc/rfc9218>.





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   [RFC9220]  Hamilton, R., "Bootstrapping WebSockets with HTTP/3",
              RFC 9220, DOI 10.17487/RFC9220, June 2022,
              <https://www.rfc-editor.org/rfc/rfc9220>.

10.2.  Informative References

   [I-D.ietf-webtrans-http2]
              Frindell, A., Kinnear, E., Pauly, T., Thomson, M.,
              Vasiliev, V., and G. Xie, "WebTransport over HTTP/2", Work
              in Progress, Internet-Draft, draft-ietf-webtrans-http2-10,
              21 October 2024, <https://datatracker.ietf.org/doc/html/
              draft-ietf-webtrans-http2-10>.

   [ORIGIN]   Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, DOI 10.17487/RFC6455, December 2011,
              <https://www.rfc-editor.org/rfc/rfc6455>.

   [RFC7301]  Friedl, S., Popov, A., Langley, A., and E. Stephan,
              "Transport Layer Security (TLS) Application-Layer Protocol
              Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301,
              July 2014, <https://www.rfc-editor.org/rfc/rfc7301>.

   [RFC9208]  Melnikov, A., "IMAP QUOTA Extension", RFC 9208,
              DOI 10.17487/RFC9208, March 2022,
              <https://www.rfc-editor.org/rfc/rfc9208>.

   [RFC9308]  Kühlewind, M. and B. Trammell, "Applicability of the QUIC
              Transport Protocol", RFC 9308, DOI 10.17487/RFC9308,
              September 2022, <https://www.rfc-editor.org/rfc/rfc9308>.

Appendix A.  Changelog

A.1.  Changes between draft versions 02 and 07

   The following changes make the draft-02 and draft-07 versions of this
   protocol incompatible:

   *  draft-07 requires SETTINGS_WEBTRANSPORT_MAX_SESSIONS (#86) and
      uses it for version negotiation (#129)

   *  draft-07 explicitly requires SETTINGS_ENABLE_CONNECT_PROTOCOL to
      be enabled (#93)

   *  draft-07 explicitly requires SETTINGS_H3_DATAGRAM to be enabled
      (#106)

   *  draft-07 only allows WEBTRANSPORT_STREAM at the beginning of the
      stream



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   The following changes that are present in draft-07 can be also
   implemented by a draft-02 implementation safely:

   *  Expanding stream reset error code space from 8 to 32 bits (#115)

   *  WEBTRANSPORT_SESSION_GONE error code (#75)

   *  Handling for HTTP GOAWAY (#76)

   *  DRAIN_WEBTRANSPORT_SESSION capsule (#79)

   *  Disallowing following redirects automatically (#113)

Index

   S W

      S

         SETTINGS_WEBTRANSPORT_INITIAL_MAX_DATA  Section 5.5, Paragraph
            2.2.1; Section 5.8, Paragraph 8; Section 9.2, Paragraph 10;
            Section 9.2, Paragraph 12.2.1
         SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_BIDI  Section 5.5,
            Paragraph 2.1.1; Section 5.6.1, Paragraph 9; Section 9.2,
            Paragraph 7; Section 9.2, Paragraph 9.2.1
         SETTINGS_WEBTRANSPORT_INITIAL_MAX_STREAMS_UNI  Section 5.5,
            Paragraph 2.1.1; Section 5.6.1, Paragraph 9; Section 9.2,
            Paragraph 4; Section 9.2, Paragraph 6.2.1

      W

         WT_DATA_BLOCKED  Section 5.3, Paragraph 3; Section 5.4,
            Paragraph 1; Section 5.9, Paragraph 1; Section 5.9,
            Paragraph 3; Section 5.9, Paragraph 5; Section 9.6,
            Paragraph 13.4.1
         WT_MAX_DATA  Section 5.3, Paragraph 1; Section 5.3, Paragraph
            2; Section 5.3, Paragraph 3; Section 5.4, Paragraph 1;
            Section 5.5, Paragraph 2.2.1; Section 5.8, Paragraph 1;
            Section 5.8, Paragraph 2; Section 5.8, Paragraph 4;
            Section 5.8, Paragraph 7; Section 9.2, Paragraph 10;
            Section 9.6, Paragraph 11.4.1
         WT_MAX_STREAMS  Section 5.2, Paragraph 1; Section 5.4,
            Paragraph 1; Section 5.5, Paragraph 2.1.1; Section 5.6.1,
            Paragraph 1; Section 5.6.1, Paragraph 2; Section 5.6.1,
            Paragraph 4; Section 5.6.1, Paragraph 8; Section 9.2,
            Paragraph 4; Section 9.2, Paragraph 7; Section 9.6,
            Paragraph 7.4.1
         WT_STREAMS_BLOCKED  Section 5.2, Paragraph 5; Section 5.4,



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            Paragraph 1; Section 5.7, Paragraph 1; Section 5.7,
            Paragraph 2; Section 5.7, Paragraph 4; Section 5.7,
            Paragraph 6; Section 9.6, Paragraph 9.4.1

Authors' Addresses

   Alan Frindell
   Facebook
   Email: afrind@fb.com


   Eric Kinnear
   Apple Inc.
   Email: ekinnear@apple.com


   Victor Vasiliev
   Google
   Email: vasilvv@google.com
































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