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<rfc xmlns:xi="http://www.w3.org/2001/XInclude" ipr="trust200902" docName="draft-ietf-webtrans-http3-04" category="std" consensus="true" tocInclude="true" sortRefs="true" symRefs="true" version="3">
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  <front>
    <title abbrev="WebTransport-H3">WebTransport over HTTP/3</title>
    <seriesInfo name="Internet-Draft" value="draft-ietf-webtrans-http3-04"/>
    <author initials="A." surname="Frindell" fullname="Alan Frindell">
      <organization>Facebook</organization>
      <address>
        <email>afrind@fb.com</email>
      </address>
    </author>
    <author initials="E." surname="Kinnear" fullname="Eric Kinnear">
      <organization>Apple Inc.</organization>
      <address>
        <email>ekinnear@apple.com</email>
      </address>
    </author>
    <author initials="V." surname="Vasiliev" fullname="Victor Vasiliev">
      <organization>Google</organization>
      <address>
        <email>vasilvv@google.com</email>
      </address>
    </author>
    <date/>
    <area>Transport</area>
    <abstract>
      <t>WebTransport <xref target="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 <xref target="HTTP3"/> and provides
support for unidirectional streams, bidirectional streams and datagrams, all
multiplexed within the same HTTP/3 connection.</t>
    </abstract>
    <note>
      <name>Note to Readers</name>
      <t>Discussion of this draft takes place on the WebTransport mailing list
(webtransport@ietf.org), which is archived at
&lt;https://mailarchive.ietf.org/arch/search/?email_list=webtransport&gt;.</t>
      <t>The repository tracking the issues for this draft can be found at
&lt;https://github.com/ietf-wg-webtrans/draft-ietf-webtrans-http3/issues&gt;.  The
web API draft corresponding to this document can be found at
&lt;https://w3c.github.io/webtransport/&gt;.</t>
    </note>
  </front>
  <middle>
    <section anchor="introduction">
      <name>Introduction</name>
      <t>HTTP/3 <xref target="HTTP3"/> is a protocol defined on top of QUIC <xref target="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
<xref target="OVERVIEW"/>.  Using the mechanism described here, multiple WebTransport
instances can be multiplexed simultaneously with regular HTTP traffic on the
same HTTP/3 connection.</t>
      <section anchor="terminology">
        <name>Terminology</name>
        <t>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 <xref target="RFC2119"/> <xref target="RFC8174"/>
when, and only when, they appear in all capitals, as shown here.</t>
        <t>This document follows terminology defined in Section 1.2 of <xref target="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.</t>
      </section>
    </section>
    <section anchor="protocol-overview">
      <name>Protocol Overview</name>
      <t>WebTransport servers in general are identified by a pair of authority value and
path value (defined in <xref target="RFC3986"/> Sections 3.2 and 3.3 correspondingly).</t>
      <t>When an HTTP/3 connection is established, both the client and server have to
send a SETTINGS_ENABLE_WEBTRANSPORT setting in order to indicate that they
both support WebTransport over HTTP/3.</t>
      <t>WebTransport sessions are initiated inside a given HTTP/3 connection by the
client, who sends an extended CONNECT request <xref target="RFC8441"/>.  If the server
accepts the request, a WebTransport session is established.  The resulting
stream will be further referred to as a <em>CONNECT stream</em>, 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 <em>Session ID</em>.</t>
      <t>After the session is established, the peers can exchange data using the
following mechanisms:</t>
      <ul spacing="normal">
        <li>A client can create a bidirectional stream using a special indefinite-length
HTTP/3 frame that transfers ownership of the stream to WebTransport.</li>
        <li>A server can create a bidirectional stream, which is possible since HTTP/3
does not define any semantics for server-initiated bidirectional streams.</li>
        <li>Both client and server can create a unidirectional stream using a special
stream type.</li>
        <li>A datagram can be sent using HTTP Datagrams
<xref target="HTTP-DATAGRAM"/>.</li>
      </ul>
      <t>A WebTransport session is terminated when the CONNECT stream that created it
is closed.</t>
    </section>
    <section anchor="session-establishment">
      <name>Session Establishment</name>
      <section anchor="establishing-a-transport-capable-http3-connection">
        <name>Establishing a Transport-Capable HTTP/3 Connection</name>
        <t>In order to indicate support for WebTransport, both the client and the server
MUST send a SETTINGS_ENABLE_WEBTRANSPORT value set to "1" in their SETTINGS
frame.  The SETTINGS_ENABLE_WEBTRANSPORT parameter value SHALL be either "0" or
"1", with "0" being the default; an endpoint that receives a value other
than "0" or "1" MUST close the connection with the H3_SETTINGS_ERROR error
code.</t>
        <t>The client MUST NOT send a WebTransport request until it has received the
setting indicating WebTransport support from the server.  Similarly, the server
MUST NOT process any incoming WebTransport requests until the client settings
have been received, as the client may be using a version of WebTransport
extension that is different from the one used by the server.</t>
        <t>In addition to the setting above, the server MUST send a
SETTINGS_MAX_WEBTRANSPORT_SESSIONS parameter indicating the maximum number of
concurrent sessions it is willing to receive.  The default value for the
SETTINGS_MAX_WEBTRANSPORT_SESSIONS parameter is "0", meaning that the server is
not willing to receive any WebTransport sessions.</t>
      </section>
      <section anchor="extended-connect-in-http3">
        <name>Extended CONNECT in HTTP/3</name>
        <t><xref target="RFC8441"/> defines an extended CONNECT method in Section 4, enabled by the
SETTINGS_ENABLE_CONNECT_PROTOCOL parameter.  That parameter is only defined for
HTTP/2.  This document does not create a new multi-purpose parameter to
indicate support for extended CONNECT in HTTP/3; instead, the
SETTINGS_ENABLE_WEBTRANSPORT setting implies that an endpoint supports extended
CONNECT.</t>
      </section>
      <section anchor="creating-a-new-session">
        <name>Creating a New Session</name>
        <t>As WebTransport sessions are established over HTTP/3, they are identified
using the <tt>https</tt> URI scheme (<xref target="HTTP"/>, Section 4.2.2).</t>
        <t>In order to create a new WebTransport session, a client can send an HTTP
CONNECT request.  The <tt>:protocol</tt> pseudo-header field (<xref target="RFC8441"/>) MUST be
set to <tt>webtransport</tt>.  The <tt>:scheme</tt> field MUST be <tt>https</tt>.  Both the
<tt>:authority</tt> and the <tt>:path</tt> value MUST be set; those fields indicate the
desired WebTransport server.  An <tt>Origin</tt> header <xref target="RFC6454"/> MUST be provided
within the request.</t>
        <t>Upon receiving an extended CONNECT request with a <tt>:protocol</tt> field set to
<tt>webtransport</tt>, the HTTP/3 server can check if it has a WebTransport
server associated with the specified <tt>:authority</tt> and <tt>:path</tt> values.  If it
does not, it SHOULD reply with status code 404 (Section 15.5.4, <xref target="HTTP"/>).
If it does, it MAY accept the session by replying with a 2xx series status
code, as defined in Section 15.3 of <xref target="HTTP"/>.  The WebTransport
server MUST verify the <tt>Origin</tt> header to ensure that the specified origin is
allowed to access the server in question.</t>
        <t>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.</t>
        <t>Clients cannot initiate WebTransport in 0-RTT packets, as the CONNECT method is not considered safe; see <xref section="10.9" sectionFormat="of" target="HTTP3"/>. However, WebTransport-related SETTINGS
parameters may be retained from the previous session as described in Section
7.2.4.2 of <xref target="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.</t>
        <t>The <tt>webtransport</tt> HTTP Upgrade Token uses the Capsule Protocol as defined in
<xref target="HTTP-DATAGRAM"/>.</t>
      </section>
      <section anchor="limiting-the-number-of-simultaneous-sessions">
        <name>Limiting the Number of Simultaneous Sessions</name>
        <t>This document defines a SETTINGS_MAX_WEBTRANSPORT_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 sessions than
indicated in the server SETTINGS parameters.  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 <tt>HTTP_REQUEST_REJECTED</tt>
status defined in <xref target="HTTP3"/>.</t>
        <t>Just like other HTTP requests, WebTransport sessions, and data sent on those
sessions, are counted against flow control limits.  This document does not
introduce additional mechanisms for endpoints to limit the relative amount of
flow control credit consumed by different WebTransport sessions, however
servers that wish to limit the rate of incoming requests on any particular
session have alternative mechanisms:</t>
        <ul spacing="normal">
          <li>The <tt>HTTP_REQUEST_REJECTED</tt> error code defined in <xref target="HTTP3"/> indicates to the
receiving HTTP/3 stack that the request was not processed in any way.</li>
          <li>HTTP status code 429 indicates that the request was rejected due to rate
limiting <xref target="RFC6585"/>.  Unlike the previous method, this signal is directly
propagated to the application.</li>
        </ul>
      </section>
    </section>
    <section anchor="webtransport-features">
      <name>WebTransport Features</name>
      <t>WebTransport over HTTP/3 provides the following features described in <xref target="OVERVIEW"/>:
unidirectional streams, bidirectional streams and datagrams, initiated by
either endpoint.</t>
      <t>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 <xref target="RFC9000"/>.</t>
      <t>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.</t>
      <t>If at any point a session ID is received that cannot a valid ID for a
client-initiated bidirectional stream, the recipient MUST close the connection
with an H3_ID_ERROR error code.</t>
      <section anchor="unidirectional-streams">
        <name>Unidirectional streams</name>
        <t>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 (<xref target="fig-unidi"/>).</t>
        <figure anchor="fig-unidi">
          <name>Unidirectional WebTransport stream format</name>
          <sourcecode type="drawing"><![CDATA[
  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                           0x54 (i)                          ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                        Session ID (i)                       ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                         Stream Body                         ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
]]></sourcecode>
        </figure>
      </section>
      <section anchor="bidirectional-streams">
        <name>Bidirectional Streams</name>
        <t>WebTransport endpoints can initiate bidirectional streams by opening an HTTP/3
bidirectional stream and sending an HTTP/3 frame with type
<tt>WEBTRANSPORT_STREAM</tt> (type=0x41).  The format of the frame SHALL be the frame
type, followed by the session ID, encoded as a variable-length integer,
followed by the user-specified stream data (<xref target="fig-bidi-client"/>).  The frame
SHALL last until the end of the stream.</t>
        <figure anchor="fig-bidi-client">
          <name>WEBTRANSPORT_STREAM frame format</name>
          <sourcecode type="drawing"><![CDATA[
  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                           0x41 (i)                          ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                        Session ID (i)                       ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                         Stream Body                         ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
]]></sourcecode>
        </figure>
        <t>HTTP/3 does not by itself define any semantics for server-initiated
bidirectional streams.  If WebTransport setting is negotiated by both
endpoints, the syntax of the server-initiated bidirectional streams SHALL be
the same as the syntax of client-initiated bidirectional streams, that is, a
sequence of HTTP/3 frames.  The only frame defined by this document for use
within server-initiated bidirectional streams is WEBTRANSPORT_STREAM.</t>
        <t>TODO: move the paragraph above into a separate draft; define what happens with
already existing HTTP/3 frames on server-initiated bidirectional streams.</t>
      </section>
      <section anchor="resetting-data-streams">
        <name>Resetting Data Streams</name>
        <t>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.</t>
        <t>A WebTransport application SHALL provide an error code for those operations.
Since WebTransport shares the error code space with HTTP/3, WebTransport
application errors for streams are limited to an unsigned 8-bit integer,
assuming values between 0x00 and 0xff.  WebTransport implementations SHALL
remap those error codes into an error range where 0x00 corresponds to
0x52e4a40fa8db, and 0xff corresponds to 0x52e4a40fa9e2.  Note that there are
code points inside that range of form "0x1f * N + 0x21" that are reserved by
<xref section="8.1" sectionFormat="of" target="HTTP3"/>; those have to be accounted for when mapping the error
codes by skipping them (i.e. the two HTTP/3 error codepoints adjacent to a
GREASE codepoint would map to two adjacent WebTransport application error
codepoints).  An example pseudocode can be seen in
<xref target="fig-remap-errors"/>.</t>
        <figure anchor="fig-remap-errors">
          <name>Pseudocode for converting between WebTransport application errors and HTTP/3 error codes; here, `//` is integer division</name>
          <artwork><![CDATA[
    first = 0x52e4a40fa8db
    last = 0x52e4a40fa9e2

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

    def http_code_to_webtransport_code(h):
        assert(first <= h <= last)
        assert((h - 0x21) % 0x1f != 0)
        shifted = h - first
        return shifted - shifted // 0x1f
]]></artwork>
        </figure>
        <t>WebTransport data streams are associated with sessions through a header at the
beginning of the stream; resetting a stream may result in that data being
discarded.  Because of that, WebTransport application error codes are best
effort, as the WebTransport stack is not always capable of associating the
reset code with a session.  The only exception is the situation where there is
only one session on a given HTTP/3 connection, and no intermediaries between
the client and the server.</t>
        <t>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.</t>
      </section>
      <section anchor="datagrams">
        <name>Datagrams</name>
        <t>Datagrams can be sent using HTTP Datagrams. The WebTransport datagram payload is
sent unmodified in the "HTTP Datagram Payload" field of an HTTP Datagram.</t>
      </section>
      <section anchor="buffering-incoming-streams-and-datagrams">
        <name>Buffering Incoming Streams and Datagrams</name>
        <t>In WebTransport over HTTP/3, the client MAY send its SETTINGS frame, as well as
multiple WebTransport CONNECT requests, WebTransport data streams and
WebTransport datagrams, all within a single flight.  As those can arrive out of
order, a WebTransport server could be put into a situation where it receives a
stream or a datagram without a corresponding session.  Similarly, a client may
receive a server-initiated stream or a datagram before receiving the CONNECT
response headers from the server.</t>
        <t>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
<tt>H3_WEBTRANSPORT_BUFFERED_STREAM_REJECTED</tt> 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.</t>
      </section>
      <section anchor="interaction-with-http3-goaway-frame">
        <name>Interaction with HTTP/3 GOAWAY frame</name>
        <t>HTTP/3 defines a graceful shutdown mechanism (Section 5.2 of <xref target="HTTP3"/>) that
allows a peer to send a GOAWAY frame indicating that it will no longer accept
any new incoming requests or pushes.  This mechanism applies to the CONNECT
requests for new WebTransport sessions.  A GOAWAY frame does not affect data
streams for existing WebTransport sessions; those can continue to be opened
even after the GOAWAY frame has been sent or received.</t>
      </section>
    </section>
    <section anchor="session-termination">
      <name>Session Termination</name>
      <t>A WebTransport session over HTTP/3 is considered terminated when either of the
following conditions is met:</t>
      <ul spacing="normal">
        <li>the CONNECT stream is closed, either cleanly or abruptly, on either side; or</li>
        <li>a CLOSE_WEBTRANSPORT_SESSION capsule is either sent or received.</li>
      </ul>
      <t>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 <xref target="RFC9000"/>) using the
H3_WEBTRANSPORT_SESSION_GONE error code; it MUST NOT send any new datagrams or
open any new streams.</t>
      <t>To terminate a session with a detailed error message, an application MAY send
an HTTP capsule <xref target="HTTP-DATAGRAM"/> of type CLOSE_WEBTRANSPORT_SESSION (0x2843).
The format of the capsule SHALL be as follows:</t>
      <artwork><![CDATA[
CLOSE_WEBTRANSPORT_SESSION Capsule {
  Type (i) = CLOSE_WEBTRANSPORT_SESSION,
  Length (i),
  Application Error Code (32),
  Application Error Message (..8192),
}
]]></artwork>
      <t>CLOSE_WEBTRANSPORT_SESSION has the following fields:</t>
      <dl>
        <dt>Application Error Code:</dt>
        <dd>
          <t>A 32-bit error code provided by the application closing the connection.</t>
        </dd>
        <dt>Application Error Message:</dt>
        <dd>
          <t>A UTF-8 encoded error message string provided by the application closing the
connection.  The message takes up the remainder of the capsule, and its
length MUST NOT exceed 1024 bytes.</t>
        </dd>
      </dl>
      <t>A CLOSE_WEBTRANSPORT_SESSION capsule MUST be followed by a FIN on the sender
side.  If any additional stream data is received on the CONNECT stream after
CLOSE_WEBTRANSPORT_SESSION, the stream MUST be reset with code
H3_MESSAGE_ERROR.  The recipient MUST close the stream upon receiving a FIN.
If the sender of CLOSE_WEBTRANSPORT_SESSION does not receive a FIN after some
time, it SHOULD send STOP_SENDING on the CONNECT stream.</t>
      <t>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.</t>
    </section>
    <section anchor="negotiating-the-draft-version">
      <name>Negotiating the Draft Version</name>
      <t>[[RFC editor: please remove this section before publication.]]</t>
      <t>WebTransport over HTTP/3 uses two different mechanisms to negotiate versions
for the different parts of the draft.</t>
      <t>The hop-by-hop wire format aspects of the protocol are negotiated by changing
the codepoint used for the SETTINGS_ENABLE_WEBTRANSPORT parameter.  Because
of that, any WebTransport endpoint MUST wait for the peer's SETTINGS frame
before sending or processing any WebTransport traffic.  When multiple versions
are supported by both of the peers, the most recent version supported by both
is selected.</t>
      <t>The data exchanged over the CONNECT stream is transmitted across intermediaries,
and thus cannot be versioned using a SETTINGS parameter.  To indicate support
for different versions of the protocol defined in this draft, the clients SHALL
send a header for each version of the draft supported.  The header
corresponding to the version described in this draft is
<tt>Sec-Webtransport-Http3-Draft02</tt>; its value SHALL be <tt>1</tt>.  The server SHALL
reply with a <tt>Sec-Webtransport-Http3-Draft</tt> header indicating the selected
version; its value SHALL be <tt>draft02</tt> for the version described in this draft.</t>
    </section>
    <section anchor="security-considerations">
      <name>Security Considerations</name>
      <t>WebTransport over HTTP/3 satisfies all of the security requirements imposed by
<xref target="OVERVIEW"/> on WebTransport protocols, thus providing a secure framework for
client-server communication in cases when the client is potentially untrusted.</t>
      <t>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.</t>
      <t>Just like HTTP traffic going over HTTP/3, WebTransport pools traffic to
different origins within a single connection.  Different origins imply
different trust domains, meaning that the implementations have to treat each
transport as potentially hostile towards others on the same connection.  One
potential attack is a resource exhaustion attack: since all of the transports
share both congestion control and flow control context, a single client
aggressively using up those resources can cause other transports to stall.  The
user agent thus SHOULD implement a fairness scheme that ensures that each
transport within connection gets a reasonable share of controlled resources;
this applies both to sending data and to opening new streams.</t>
      <t>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.</t>
    </section>
    <section anchor="iana-considerations">
      <name>IANA Considerations</name>
      <section anchor="upgrade-token-registration">
        <name>Upgrade Token Registration</name>
        <t>The following entry is added to the "Hypertext Transfer Protocol (HTTP) Upgrade
Token Registry" registry established by Section 16.7 of <xref target="HTTP"/>.</t>
        <t>The "webtransport" label identifies HTTP/3 used as a protocol for WebTransport:</t>
        <dl>
          <dt>Value:</dt>
          <dd>
            <t>webtransport</t>
          </dd>
          <dt>Description:</dt>
          <dd>
            <t>WebTransport over HTTP/3</t>
          </dd>
          <dt>Reference:</dt>
          <dd>
            <t>This document and <xref target="I-D.ietf-webtrans-http2"/></t>
          </dd>
        </dl>
      </section>
      <section anchor="http3-settings-parameter-registration">
        <name>HTTP/3 SETTINGS Parameter Registration</name>
        <t>The following entries are added to the "HTTP/3 Settings" registry established by
<xref target="HTTP3"/>:</t>
        <t>The <tt>SETTINGS_ENABLE_WEBTRANSPORT</tt> parameter indicates that the specified
HTTP/3 connection is WebTransport-capable.</t>
        <dl>
          <dt>Setting Name:</dt>
          <dd>
            <t>ENABLE_WEBTRANSPORT</t>
          </dd>
          <dt>Value:</dt>
          <dd>
            <t>0x2b603742</t>
          </dd>
          <dt>Default:</dt>
          <dd>
            <t>0</t>
          </dd>
          <dt>Specification:</dt>
          <dd>
            <t>This document</t>
          </dd>
        </dl>
        <t>The <tt>SETTINGS_WEBTRANSPORT_MAX_SESSIONS</tt> parameter indicates that the specified
HTTP/3 server is WebTransport-capable and the number of concurrent sessions
it is willing to receive.</t>
        <dl>
          <dt>Setting Name:</dt>
          <dd>
            <t>WEBTRANSPORT_MAX_SESSIONS</t>
          </dd>
          <dt>Value:</dt>
          <dd>
            <t>0x2b603743</t>
          </dd>
          <dt>Default:</dt>
          <dd>
            <t>0</t>
          </dd>
          <dt>Specification:</dt>
          <dd>
            <t>This document</t>
          </dd>
        </dl>
      </section>
      <section anchor="frame-type-registration">
        <name>Frame Type Registration</name>
        <t>The following entry is added to the "HTTP/3 Frame Type" registry established by
<xref target="HTTP3"/>:</t>
        <t>The <tt>WEBTRANSPORT_STREAM</tt> frame allows HTTP/3 client-initiated bidirectional
streams to be used by WebTransport:</t>
        <dl>
          <dt>Code:</dt>
          <dd>
            <t>0x41</t>
          </dd>
          <dt>Frame Type:</dt>
          <dd>
            <t>WEBTRANSPORT_STREAM</t>
          </dd>
          <dt>Specification:</dt>
          <dd>
            <t>This document</t>
          </dd>
        </dl>
      </section>
      <section anchor="stream-type-registration">
        <name>Stream Type Registration</name>
        <t>The following entry is added to the "HTTP/3 Stream Type" registry established by
<xref target="HTTP3"/>:</t>
        <t>The "WebTransport stream" type allows unidirectional streams to be used by
WebTransport:</t>
        <dl>
          <dt>Code:</dt>
          <dd>
            <t>0x54</t>
          </dd>
          <dt>Stream Type:</dt>
          <dd>
            <t>WebTransport stream</t>
          </dd>
          <dt>Specification:</dt>
          <dd>
            <t>This document</t>
          </dd>
          <dt>Sender:</dt>
          <dd>
            <t>Both</t>
          </dd>
        </dl>
      </section>
      <section anchor="http3-error-code-registration">
        <name>HTTP/3 Error Code Registration</name>
        <t>The following entry is added to the "HTTP/3 Error Code" registry established by
<xref target="HTTP3"/>:</t>
        <dl>
          <dt>Name:</dt>
          <dd>
            <t>H3_WEBTRANSPORT_BUFFERED_STREAM_REJECTED</t>
          </dd>
          <dt>Value:</dt>
          <dd>
            <t>0x3994bd84</t>
          </dd>
          <dt>Description:</dt>
          <dd>
            <t>WebTransport data stream rejected due to lack of associated session.</t>
          </dd>
          <dt>Specification:</dt>
          <dd>
            <t>This document.</t>
          </dd>
          <dt>Name:</dt>
          <dd>
            <t>H3_WEBTRANSPORT_SESSION_GONE</t>
          </dd>
          <dt>Value:</dt>
          <dd>
            <t>0x170d7b68</t>
          </dd>
          <dt>Description:</dt>
          <dd>
            <t>WebTransport data stream aborted because the associated WebTransport session
has been closed.</t>
          </dd>
          <dt>Specification:</dt>
          <dd>
            <t>This document.</t>
          </dd>
        </dl>
        <t>In addition, the following range of entries is registered:</t>
        <dl>
          <dt>Name:</dt>
          <dd>
            <t>H3_WEBTRANSPORT_APPLICATION_00 ... H3_WEBTRANSPORT_APPLICATION_FF</t>
          </dd>
          <dt>Value:</dt>
          <dd>
            <t>0x52e4a40fa8db to 0x52e4a40fa9e2 inclusive, with the exception of
0x52e4a40fa8f9, 0x52e4a40fa918, 0x52e4a40fa937, 0x52e4a40fa956,
0x52e4a40fa975, 0x52e4a40fa994, 0x52e4a40fa9b3, and 0x52e4a40fa9d2.</t>
          </dd>
          <dt>Description:</dt>
          <dd>
            <t>WebTransport application error codes.</t>
          </dd>
          <dt>Specification:</dt>
          <dd>
            <t>This document.</t>
          </dd>
        </dl>
      </section>
      <section anchor="capsule-types">
        <name>Capsule Types</name>
        <t>The following entries are added to the "HTTP Capsule Types" registry established
by <xref target="HTTP-DATAGRAM"/>:</t>
        <t>The <tt>CLOSE_WEBTRANSPORT_SESSION</tt> capsule.</t>
        <dl spacing="compact">
          <dt>Value:</dt>
          <dd>
            <t>0x2843</t>
          </dd>
          <dt>Capsule Type:</dt>
          <dd>
            <t>CLOSE_WEBTRANSPORT_SESSION</t>
          </dd>
          <dt>Status:</dt>
          <dd>
            <t>permanent</t>
          </dd>
          <dt>Specification:</dt>
          <dd>
            <t>This document</t>
          </dd>
          <dt>Change Controller:</dt>
          <dd>
            <t>IETF</t>
          </dd>
          <dt>Contact:</dt>
          <dd>
            <t>WebTransport Working Group <eref target="mailto:webtransport@ietf.org">webtransport@ietf.org</eref></t>
          </dd>
          <dt>Notes:</dt>
          <dd>
            <t>None</t>
          </dd>
        </dl>
      </section>
    </section>
  </middle>
  <back>
    <references>
      <name>References</name>
      <references>
        <name>Normative References</name>
        <reference anchor="OVERVIEW">
          <front>
            <title>The WebTransport Protocol Framework</title>
            <author fullname="Victor Vasiliev" initials="V." surname="Vasiliev">
              <organization>Google</organization>
            </author>
            <date day="11" month="July" year="2022"/>
            <abstract>
              <t>   The WebTransport Protocol Framework enables clients constrained by
   the Web security model to communicate with a remote server using a
   secure multiplexed transport.  It consists of a set of individual
   protocols that are safe to expose to untrusted applications, combined
   with a model that allows them to be used interchangeably.

   This document defines the overall requirements on the protocols used
   in WebTransport, as well as the common features of the protocols,
   support for some of which may be optional.

              </t>
            </abstract>
          </front>
          <seriesInfo name="Internet-Draft" value="draft-ietf-webtrans-overview-04"/>
        </reference>
        <reference anchor="HTTP3">
          <front>
            <title>HTTP/3</title>
            <author fullname="M. Bishop" initials="M." role="editor" surname="Bishop">
              <organization/>
            </author>
            <date month="June" year="2022"/>
            <abstract>
              <t>The QUIC transport protocol has several features that are desirable in a transport for HTTP, such as stream multiplexing, per-stream flow control, and low-latency connection establishment.  This document describes a mapping of HTTP semantics over QUIC.  This document also identifies HTTP/2 features that are subsumed by QUIC and describes how HTTP/2 extensions can be ported to HTTP/3.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="9114"/>
          <seriesInfo name="DOI" value="10.17487/RFC9114"/>
        </reference>
        <reference anchor="RFC9000">
          <front>
            <title>QUIC: A UDP-Based Multiplexed and Secure Transport</title>
            <author fullname="J. Iyengar" initials="J." role="editor" surname="Iyengar">
              <organization/>
            </author>
            <author fullname="M. Thomson" initials="M." role="editor" surname="Thomson">
              <organization/>
            </author>
            <date month="May" year="2021"/>
            <abstract>
              <t>This document defines the core of the QUIC transport protocol.  QUIC provides applications with flow-controlled streams for structured communication, low-latency connection establishment, and network path migration. QUIC includes security measures that ensure confidentiality, integrity, and availability in a range of deployment circumstances.  Accompanying documents describe the integration of TLS for key negotiation, loss detection, and an exemplary congestion control algorithm.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="9000"/>
          <seriesInfo name="DOI" value="10.17487/RFC9000"/>
        </reference>
        <reference anchor="RFC2119">
          <front>
            <title>Key words for use in RFCs to Indicate Requirement Levels</title>
            <author fullname="S. Bradner" initials="S." surname="Bradner">
              <organization/>
            </author>
            <date month="March" year="1997"/>
            <abstract>
              <t>In many standards track documents several words are used to signify the requirements in the specification.  These words are often capitalized. This document defines these words as they should be interpreted in IETF documents.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t>
            </abstract>
          </front>
          <seriesInfo name="BCP" value="14"/>
          <seriesInfo name="RFC" value="2119"/>
          <seriesInfo name="DOI" value="10.17487/RFC2119"/>
        </reference>
        <reference anchor="RFC8174">
          <front>
            <title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</title>
            <author fullname="B. Leiba" initials="B." surname="Leiba">
              <organization/>
            </author>
            <date month="May" year="2017"/>
            <abstract>
              <t>RFC 2119 specifies common key words that may be used in protocol  specifications.  This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the  defined special meanings.</t>
            </abstract>
          </front>
          <seriesInfo name="BCP" value="14"/>
          <seriesInfo name="RFC" value="8174"/>
          <seriesInfo name="DOI" value="10.17487/RFC8174"/>
        </reference>
        <reference anchor="RFC3986">
          <front>
            <title>Uniform Resource Identifier (URI): Generic Syntax</title>
            <author fullname="T. Berners-Lee" initials="T." surname="Berners-Lee">
              <organization/>
            </author>
            <author fullname="R. Fielding" initials="R." surname="Fielding">
              <organization/>
            </author>
            <author fullname="L. Masinter" initials="L." surname="Masinter">
              <organization/>
            </author>
            <date month="January" year="2005"/>
            <abstract>
              <t>A Uniform Resource Identifier (URI) is a compact sequence of characters that identifies an abstract or physical resource.  This specification defines the generic URI syntax and a process for resolving URI references that might be in relative form, along with guidelines and security considerations for the use of URIs on the Internet.  The URI syntax defines a grammar that is a superset of all valid URIs, allowing an implementation to parse the common components of a URI reference without knowing the scheme-specific requirements of every possible identifier.  This specification does not define a generative grammar for URIs; that task is performed by the individual specifications of each URI scheme.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="STD" value="66"/>
          <seriesInfo name="RFC" value="3986"/>
          <seriesInfo name="DOI" value="10.17487/RFC3986"/>
        </reference>
        <reference anchor="RFC8441">
          <front>
            <title>Bootstrapping WebSockets with HTTP/2</title>
            <author fullname="P. McManus" initials="P." surname="McManus">
              <organization/>
            </author>
            <date month="September" year="2018"/>
            <abstract>
              <t>This document defines a mechanism for running the WebSocket Protocol (RFC 6455) over a single stream of an HTTP/2 connection.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8441"/>
          <seriesInfo name="DOI" value="10.17487/RFC8441"/>
        </reference>
        <reference anchor="HTTP-DATAGRAM">
          <front>
            <title>HTTP Datagrams and the Capsule Protocol</title>
            <author fullname="David Schinazi" initials="D." surname="Schinazi">
              <organization>Google LLC</organization>
            </author>
            <author fullname="Lucas Pardue" initials="L." surname="Pardue">
              <organization>Cloudflare</organization>
            </author>
            <date day="17" month="June" year="2022"/>
            <abstract>
              <t>This document describes HTTP Datagrams, a convention for conveying multiplexed, potentially unreliable datagrams inside an HTTP connection.

 In HTTP/3, HTTP Datagrams can be sent unreliably using the QUIC DATAGRAM extension. When the QUIC DATAGRAM frame is unavailable or undesirable, HTTP Datagrams can be sent using the Capsule Protocol, which is a more general convention for conveying data in HTTP connections.

 HTTP Datagrams and the Capsule Protocol are intended for use by HTTP extensions, not applications.
              </t>
            </abstract>
          </front>
          <seriesInfo name="Internet-Draft" value="draft-ietf-masque-h3-datagram-11"/>
        </reference>
        <reference anchor="RFC6454">
          <front>
            <title>The Web Origin Concept</title>
            <author fullname="A. Barth" initials="A." surname="Barth">
              <organization/>
            </author>
            <date month="December" year="2011"/>
            <abstract>
              <t>This document defines the concept of an "origin", which is often used as the scope of authority or privilege by user agents.  Typically, user agents isolate content retrieved from different origins to prevent malicious web site operators from interfering with the operation of benign web sites.  In addition to outlining the principles that underlie the concept of origin, this document details how to determine the origin of a URI and how to serialize an origin into a string.  It also defines an HTTP header field, named "Origin", that indicates which origins are associated with an HTTP request.   [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="6454"/>
          <seriesInfo name="DOI" value="10.17487/RFC6454"/>
        </reference>
        <reference anchor="HTTP">
          <front>
            <title>HTTP Semantics</title>
            <author fullname="R. Fielding" initials="R." role="editor" surname="Fielding">
              <organization/>
            </author>
            <author fullname="M. Nottingham" initials="M." role="editor" surname="Nottingham">
              <organization/>
            </author>
            <author fullname="J. Reschke" initials="J." role="editor" surname="Reschke">
              <organization/>
            </author>
            <date month="June" year="2022"/>
            <abstract>
              <t>The Hypertext Transfer Protocol (HTTP) is a stateless application-level protocol for distributed, collaborative, hypertext information systems. This document describes the overall architecture of HTTP, establishes common terminology, and defines aspects of the protocol that are shared by all versions. In this definition are core protocol elements, extensibility mechanisms, and the "http" and "https" Uniform Resource Identifier (URI) schemes. </t>
              <t>This document updates RFC 3864 and obsoletes RFCs 2818, 7231, 7232, 7233, 7235, 7538, 7615, 7694, and portions of 7230.</t>
            </abstract>
          </front>
          <seriesInfo name="STD" value="97"/>
          <seriesInfo name="RFC" value="9110"/>
          <seriesInfo name="DOI" value="10.17487/RFC9110"/>
        </reference>
        <reference anchor="RFC6585">
          <front>
            <title>Additional HTTP Status Codes</title>
            <author fullname="M. Nottingham" initials="M." surname="Nottingham">
              <organization/>
            </author>
            <author fullname="R. Fielding" initials="R." surname="Fielding">
              <organization/>
            </author>
            <date month="April" year="2012"/>
            <abstract>
              <t>This document specifies additional HyperText Transfer Protocol (HTTP) status codes for a variety of common situations.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="6585"/>
          <seriesInfo name="DOI" value="10.17487/RFC6585"/>
        </reference>
      </references>
      <references>
        <name>Informative References</name>
        <reference anchor="I-D.ietf-webtrans-http2">
          <front>
            <title>WebTransport using HTTP/2</title>
            <author fullname="Alan Frindell" initials="A." surname="Frindell">
              <organization>Facebook Inc.</organization>
            </author>
            <author fullname="Eric Kinnear" initials="E." surname="Kinnear">
              <organization>Apple Inc.</organization>
            </author>
            <author fullname="Tommy Pauly" initials="T." surname="Pauly">
              <organization>Apple Inc.</organization>
            </author>
            <author fullname="Martin Thomson" initials="M." surname="Thomson">
              <organization>Mozilla</organization>
            </author>
            <author fullname="Victor Vasiliev" initials="V." surname="Vasiliev">
              <organization>Google</organization>
            </author>
            <author fullname="Guowu Xie" initials="G." surname="Xie">
              <organization>Facebook Inc.</organization>
            </author>
            <date day="19" month="September" year="2022"/>
            <abstract>
              <t>   WebTransport defines a set of low-level communications features
   designed for client-server interactions that are initiated by Web
   clients.  This document describes a protocol that can provide many of
   the capabilities of WebTransport over HTTP/2.  This protocol enables
   the use of WebTransport when a UDP-based protocol is not available.

              </t>
            </abstract>
          </front>
          <seriesInfo name="Internet-Draft" value="draft-ietf-webtrans-http2-04"/>
        </reference>
      </references>
    </references>
  </back>
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