Network Working Group A. Frindell
Internet-Draft Facebook
Intended status: Standards Track E. Kinnear
Expires: 28 April 2022 Apple Inc.
V. Vasiliev
Google
25 October 2021
WebTransport over HTTP/3
draft-ietf-webtrans-http3-02
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 28 April 2022.
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Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (https://trustee.ietf.org/
license-info) in effect on the date of publication of this document.
Please review these documents carefully, as they describe your rights
and restrictions with respect to this document. Code Components
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as described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 3
2. Protocol Overview . . . . . . . . . . . . . . . . . . . . . . 3
3. Session Establishment . . . . . . . . . . . . . . . . . . . . 4
3.1. Establishing a Transport-Capable HTTP/3 Connection . . . 4
3.2. Extended CONNECT in HTTP/3 . . . . . . . . . . . . . . . 4
3.3. Creating a New Session . . . . . . . . . . . . . . . . . 4
3.4. Limiting the Number of Simultaneous Sessions . . . . . . 5
4. WebTransport Features . . . . . . . . . . . . . . . . . . . . 5
4.1. Unidirectional streams . . . . . . . . . . . . . . . . . 6
4.2. Bidirectional Streams . . . . . . . . . . . . . . . . . . 6
4.3. Resetting Data Streams . . . . . . . . . . . . . . . . . 7
4.4. Datagrams . . . . . . . . . . . . . . . . . . . . . . . . 8
4.5. Buffering Incoming Streams and Datagrams . . . . . . . . 8
5. Session Termination . . . . . . . . . . . . . . . . . . . . . 9
6. Negotiating the Draft Version . . . . . . . . . . . . . . . . 10
7. Security Considerations . . . . . . . . . . . . . . . . . . . 10
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11
8.1. Upgrade Token Registration . . . . . . . . . . . . . . . 11
8.2. HTTP/3 SETTINGS Parameter Registration . . . . . . . . . 11
8.3. Frame Type Registration . . . . . . . . . . . . . . . . . 12
8.4. Stream Type Registration . . . . . . . . . . . . . . . . 12
8.5. HTTP/3 Error Code Registration . . . . . . . . . . . . . 12
8.6. Datagram Format Type . . . . . . . . . . . . . . . . . . 13
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 13
9.1. Normative References . . . . . . . . . . . . . . . . . . 13
9.2. Informative References . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15
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1. Introduction
HTTP/3 [HTTP3] is a protocol defined on top of QUIC [RFC9000] that
can multiplex HTTP requests over a QUIC connection. This document
defines a mechanism for multiplexing non-HTTP data with HTTP/3 in a
manner that conforms with the WebTransport protocol requirements and
semantics [OVERVIEW]. Using the mechanism described here, multiple
WebTransport instances can be multiplexed simultaneously with regular
HTTP traffic on the same HTTP/3 connection.
1.1. Terminology
The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"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. 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, both the client and server
have to send a SETTINGS_ENABLE_WEBTRANSPORT setting in order to
indicate that they both support WebTransport over HTTP/3.
WebTransport sessions are initiated inside a given HTTP/3 connection
by the client, who sends an extended CONNECT request [RFC8441]. If
the server accepts the request, an WebTransport session is
established. The resulting stream will be further referred to as a
_CONNECT stream_, and its stream ID is used to uniquely identify a
given WebTransport session within the connection. The ID of the
CONNECT stream that established a given WebTransport session will be
further referred to as a _Session ID_.
After the session is established, the peers can exchange data using
the following mechanisms:
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* A client can create a bidirectional stream using a special
indefinite-length HTTP/3 frame that transfers ownership of the
stream to WebTransport.
* A server can create a bidirectional stream, which is possible
since HTTP/3 does not define any semantics for server-initiated
bidirectional streams.
* Both client and server can create a unidirectional stream using a
special stream type.
* A datagram can be sent using HTTP Datagrams [HTTP-DATAGRAM].
An WebTransport session is terminated when the CONNECT stream that
created it is closed.
3. Session Establishment
3.1. Establishing a Transport-Capable HTTP/3 Connection
In order to indicate support for WebTransport, both the client and
the server MUST send a SETTINGS_ENABLE_WEBTRANSPORT value set to "1"
in their SETTINGS frame. The SETTINGS_ENABLE_WEBTRANSPORT parameter
value SHALL be either "0" or "1", with "0" being the default; an
endpoint that receives a value other than "0" or "1" MUST close the
connection with the H3_SETTINGS_ERROR error code.
The client MUST NOT send a WebTransport request until it has received
the setting indicating WebTransport support from the server.
Similarly, the server MUST NOT process any incoming WebTransport
requests until the client settings have been received, as the client
may be using a version of WebTransport extension that is different
from the one used by the server.
3.2. Extended CONNECT in HTTP/3
[RFC8441] defines an extended CONNECT method in Section 4, enabled by
the SETTINGS_ENABLE_CONNECT_PROTOCOL parameter. That parameter is
only defined for HTTP/2. This document does not create a new multi-
purpose parameter to indicate support for extended CONNECT in HTTP/3;
instead, the SETTINGS_ENABLE_WEBTRANSPORT setting implies that an
endpoint supports extended CONNECT.
3.3. Creating a New Session
As WebTransport sessions are established over HTTP/3, they are
identified using the https URI scheme [RFC7230].
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In order to create a new WebTransport session, a client can send an
HTTP CONNECT request. The :protocol pseudo-header field ([RFC8441])
MUST be set to webtransport. The :scheme field MUST be https. Both
the :authority and the :path value MUST be set; those fields indicate
the desired WebTransport server. An Origin header [RFC6454] MUST be
provided within the request.
Upon receiving an extended CONNECT request with a :protocol field set
to webtransport, the HTTP/3 server can check if it has a WebTransport
server associated with the specified :authority and :path values. If
it does not, it SHOULD reply with status code 404 (Section 6.5.4,
[RFC7231]). If it does, it MAY accept the session by replying with a
2xx series status code, as defined in Section 15.3 of [SEMANTICS].
The WebTransport server MUST verify the Origin header to ensure that
the specified origin is allowed to access the server in question.
From the client's perspective, a WebTransport session is established
when the client receives a 2xx response. From the server's
perspective, a session is established once it sends a 2xx response.
WebTransport over HTTP/3 does not support 0-RTT.
The webtransport HTTP Upgrade Token uses the Capsule Protocol as
defined in [HTTP-DATAGRAM].
3.4. Limiting the Number of Simultaneous Sessions
From the flow control perspective, WebTransport sessions count
against the stream flow control just like regular HTTP requests,
since they are established via an HTTP CONNECT request. This
document does not make any effort to introduce a separate flow
control mechanism for sessions, nor to separate HTTP requests from
WebTransport data streams. If the server needs to limit the rate of
incoming requests, it has alternative mechanisms at its disposal:
* HTTP_REQUEST_REJECTED error code defined in [HTTP3] indicates to
the receiving HTTP/3 stack that the request was not processed in
any way.
* HTTP status code 429 indicates that the request was rejected due
to rate limiting [RFC6585]. Unlike the previous method, this
signal is directly propagated to the application.
4. WebTransport Features
WebTransport over HTTP/3 provides the following features described in
[OVERVIEW]: unidirectional streams, bidirectional streams and
datagrams, initiated by either endpoint.
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Session IDs are used to demultiplex streams and datagrams belonging
to different WebTransport sessions. On the wire, session IDs are
encoded using the QUIC variable length integer scheme described in
[RFC9000].
If at any point a session ID is received that cannot a valid ID for a
client-initiated bidirectional stream, the recepient MUST close the
connection with an H3_ID_ERROR error code.
4.1. Unidirectional streams
Once established, both endpoints can open unidirectional streams.
The HTTP/3 unidirectional stream type SHALL be 0x54. The body of the
stream SHALL be the stream type, followed by the session ID, encoded
as a variable-length integer, followed by the user-specified stream
data (Figure 1).
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 0x54 (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Session ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream Body ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: Unidirectional WebTransport stream format
4.2. Bidirectional Streams
WebTransport endpoints can initiate bidirectional streams by opening
an HTTP/3 bidirectional stream and sending an HTTP/3 frame with type
WEBTRANSPORT_STREAM (type=0x41). The format of the frame SHALL be
the frame type, followed by the session ID, encoded as a variable-
length integer, followed by the user-specified stream data
(Figure 2). The frame SHALL last until the end of the stream.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 0x41 (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Session ID (i) ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Stream Body ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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Figure 2: WEBTRANSPORT_STREAM frame format
HTTP/3 does not by itself define any semantics for server-initiated
bidirectional streams. If WebTransport setting is negotiated by both
endpoints, the syntax of the server-initiated bidirectional streams
SHALL be the same as the syntax of client-initated bidirectional
streams, that is, a sequence of HTTP/3 frames. The only frame
defined by this document for use within server-initiated
bidirectional streams is WEBTRANSPORT_STREAM.
TODO: move the paragraph above into a separate draft; define what
happens with already existing HTTP/3 frames on server-initiated
bidirectional streams.
4.3. Resetting Data Streams
A WebTransport endpoint may send a RESET_STREAM or a STOP_SENDING
frame for a WebTransport data stream. Those signals are propagated
by the WebTransport implementation to the application.
A WebTransport application SHALL provide an error code for those
operations. Since WebTransport shares the error code space with
HTTP/3, WebTransport application errors for streams are limited to an
unsigned 8-bit integer, assuming values between 0x00 and 0xff.
WebTransport implementations SHALL remap those error codes into an
error range where 0x00 corresponds to 0x52e4a40fa8db, and 0xff
corresponds to 0x52e4a40fa9e2. Note that there are code points
inside that range of form "0x1f * N + 0x21" that are reserved by
Section 8.1 of [HTTP3]; those have to be accounted for when mapping
the error codes by skipping them (i.e. the two HTTP/3 error
codepoints adjacent to a GREASE codepoint would map to two adjacent
WebTransport application error codepoints). An example pseudocode
can be seen in Figure 3.
first = 0x52e4a40fa8db
last = 0x52e4a40fa9e2
def webtransport_code_to_http_code(n):
return first + n + floor(n / 0x1e)
def http_code_to_webtransport_code(h):
assert(first <= h <= last)
assert((h - 0x21) % 0x1f != 0)
shifted = h - first
return shifted - shifted // 0x1f
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Figure 3: Pseudocode for converting between WebTransport
application errors and HTTP/3 error codes; here, `//` is integer
division
WebTransport data streams are associated with sessions through a
header at the beginning of the stream; resetting a stream may result
in that data being discarded. Because of that, WebTransport
application error codes are best effort, as the WebTransport stack is
not always capable of associating the reset code with a session. The
only exception is the situation where there is only one session on a
given HTTP/3 connection, and no intermediaries between the client and
the server.
WebTransport implementations SHALL forward the error code for a
stream associated with a known session to the application that owns
that session; similarly, the intermediaries SHALL reset the streams
with corresponding error code when receiving a reset from the peer.
If a WebTransport implementation intentionally allows only one
session over a given HTTP/3 connection, it SHALL forward the error
codes within WebTransport application error code range to the
application that owns the only session on that connection.
4.4. Datagrams
Datagrams can be sent using HTTP Datagrams, using the WEB_TRANSPORT
HTTP Datagram Format Type (see value in Section 8.6). When using the
WEB_TRANSPORT HTTP Datagram Format Type, the WebTransport datagram
payload is sent unmodified in the "HTTP Datagram Payload" field of an
HTTP Datagram. When sending a registration capsule using the
"Datagram Format Type" set to WEB_TRANSPORT, the "Datagram Format
Additional Data" field SHALL be empty.
In QUIC, a datagram frame can span at most one packet. Because of
that, the applications have to know the maximum size of the datagram
they can send. However, when proxying the datagrams, the hop-by-hop
MTUs can vary. TODO: Describe how the path MTU can be computed,
specifically propagation across HTTP proxies.
4.5. Buffering Incoming Streams and Datagrams
In WebTransport over HTTP/3, the client MAY send its SETTINGS frame,
as well as multiple WebTransport CONNECT requests, WebTransport data
streams and WebTransport datagrams, all within a single flight. As
those can arrive out of order, a WebTransport server could be put
into a situation where it receives a stream or a datagram without a
corresponding session. Similarly, a client may receive a server-
initiated stream or a datagram before receiving the CONNECT response
headers from the server.
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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 H3_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.
5. Session Termination
A WebTransport session over HTTP/3 is considered terminated when
either of the following conditions is met:
* the CONNECT stream is closed, either cleanly or abruptly, on
either side; or
* a CLOSE_WEBTRANSPORT_SESSION capsule is either sent or received.
Upon learning that the session has been terminated, the endpoint MUST
reset all of the streams associated with the session; 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:
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 connection.
Application Error Message: A UTF-8 encoded error message string
provided by the application closing the connection. The message
takes up the remainer of the capsule, and its length MUST NOT
exceed 1024 bytes.
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A CLOSE_WEBTRANSPORT_SESSION capsule MUST be followed by a FIN on the
sender side. If any additional stream data is received on the
CONNECT stream after CLOSE_WEBTRANSPORT_SESSION, the stream MUST be
reset with code H3_MESSAGE_ERROR. The recipient MUST close the
stream upon receiving a FIN. If the sender of
CLOSE_WEBTRANSPORT_SESSION does not receive a FIN after some time, it
SHOULD send STOP_SENDING on the CONNECT stream.
Cleanly terminating a CONNECT stream without a
CLOSE_WEBTRANSPORT_SESSION capsule SHALL be semantically equivalent
to terminating it with a CLOSE_WEBTRANSPORT_SESSION capsule that has
an error code of 0 and an empty error string.
6. Negotiating the Draft Version
[[RFC editor: please remove this section before publication.]]
WebTransport over HTTP/3 uses two different mechanisms to negotiate
versions for the different parts of the draft.
The hop-by-hop wire format aspects of the protocol are negotiated by
changing the codepoint used for the SETTINGS_ENABLE_WEBTRANSPORT
parameter. Because of that, any WebTransport endpoint MUST wait for
the peer's SETTINGS frame before sending or processing any
WebTransport traffic. When multiple versions are supported by both
of the peers, the most recent version supported by both is selected.
The data exchanged over the CONNECT stream is transmitted across
intermediaries, and thus cannot be versioned using a SETTINGS
parameter. To indicate support for different versions of the
protocol defined in this draft, the clients SHALL send a header for
each version of the draft supported. The header corresponding to the
version described in this draft is Sec-Webtransport-Http3-Draft02;
its value SHALL be 1. The server SHALL reply with a Sec-
Webtransport-Http3-Draft header indicating the selected version; its
value SHALL be draft02 for the version described in this draft.
7. Security Considerations
WebTransport over HTTP/3 satisfies all of the security requirements
imposed by [OVERVIEW] on WebTransport protocols, thus providing a
secure framework for client-server communication in cases when the
client is potentially untrusted.
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WebTransport over HTTP/3 requires explicit opt-in through the use of
a QUIC transport parameter; this avoids potential protocol confusion
attacks by ensuring the HTTP/3 server explicitly supports it. It
also requires the use of the Origin header, providing the server with
the ability to deny access to Web-based clients that do not originate
from a trusted origin.
Just like HTTP traffic going over HTTP/3, WebTransport pools traffic
to different origins within a single connection. Different origins
imply different trust domains, meaning that the implementations have
to treat each transport as potentially hostile towards others on the
same connection. One potential attack is a resource exhaustion
attack: since all of the transports share both congestion control and
flow control context, a single client aggressively using up those
resources can cause other transports to stall. The user agent thus
SHOULD implement a fairness scheme that ensures that each transport
within connection gets a reasonable share of controlled resources;
this applies both to sending data and to opening new streams.
8. IANA Considerations
8.1. Upgrade Token Registration
The following entry is added to the "Hypertext Transfer Protocol
(HTTP) Upgrade Token Registry" registry established by [RFC7230]:
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]
8.2. HTTP/3 SETTINGS Parameter Registration
The following entry is added to the "HTTP/3 Settings" registry
established by [HTTP3]:
The SETTINGS_ENABLE_WEBTRANSPORT parameter indicates that the
specified HTTP/3 connection is WebTransport-capable.
Setting Name: ENABLE_WEBTRANSPORT
Value: 0x2b603742
Default: 0
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Specification: This document
8.3. Frame Type Registration
The following entry is added to the "HTTP/3 Frame Type" registry
established by [HTTP3]:
The WEBTRANSPORT_STREAM frame allows HTTP/3 client-initiated
bidirectional streams to be used by WebTransport:
Code: 0x41
Frame Type: WEBTRANSPORT_STREAM
Specification: This document
8.4. Stream Type Registration
The following entry is added to the "HTTP/3 Stream Type" registry
established by [HTTP3]:
The "WebTransport stream" type allows unidirectional streams to be
used by WebTransport:
Code: 0x54
Stream Type: WebTransport stream
Specification: This document
Sender: Both
8.5. HTTP/3 Error Code Registration
The following entry is added to the "HTTP/3 Error Code" registry
established by [HTTP3]:
Name: H3_WEBTRANSPORT_BUFFERED_STREAM_REJECTED
Value: 0x3994bd84
Description: WebTransport data stream rejected due to lack of
associated session.
Specification: This document.
In addition, the following range of entries is registered:
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Name: H3_WEBTRANSPORT_APPLICATION_00 ...
H3_WEBTRANSPORT_APPLICATION_FF
Value: 0x52e4a40fa8db to 0x52e4a40fa9e2 inclusive, with the
exception of 0x52e4a40fa8f9, 0x52e4a40fa918, 0x52e4a40fa937,
0x52e4a40fa956, 0x52e4a40fa975, 0x52e4a40fa994, 0x52e4a40fa9b3,
and 0x52e4a40fa9d2.
Description: WebTransport application error codes.
Specification: This document.
8.6. Datagram Format Type
This document will request IANA to register WEB_TRANSPORT in the
"HTTP Datagram Format Types" registry established by [HTTP-DATAGRAM].
+===============+==========+===============+
| Type | Value | Specification |
+===============+==========+===============+
| WEB_TRANSPORT | 0xff7c00 | This Document |
+---------------+----------+---------------+
Table 1: Registered Datagram Format Type
9. References
9.1. Normative References
[HTTP-DATAGRAM]
Schinazi, D. and L. Pardue, "Using Datagrams with HTTP",
Work in Progress, Internet-Draft, draft-ietf-masque-h3-
datagram-05, 25 October 2021,
<https://datatracker.ietf.org/doc/html/draft-ietf-masque-
h3-datagram-05>.
[HTTP3] Bishop, M., Ed., "Hypertext Transfer Protocol Version 3
(HTTP/3)", Work in Progress, Internet-Draft, draft-ietf-
quic-http, <https://datatracker.ietf.org/doc/html/draft-
ietf-quic-http>.
[OVERVIEW] Vasiliev, V., "The WebTransport Protocol Framework", Work
in Progress, Internet-Draft, draft-ietf-webtrans-overview-
latest, <https://datatracker.ietf.org/doc/html/draft-ietf-
webtrans-overview-latest>.
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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://doi.org/10.17487/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://doi.org/10.17487/RFC3986>.
[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
DOI 10.17487/RFC6454, December 2011,
<https://doi.org/10.17487/RFC6454>.
[RFC6585] Nottingham, M. and R. Fielding, "Additional HTTP Status
Codes", RFC 6585, DOI 10.17487/RFC6585, April 2012,
<https://doi.org/10.17487/RFC6585>.
[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Message Syntax and Routing",
RFC 7230, DOI 10.17487/RFC7230, June 2014,
<https://doi.org/10.17487/RFC7230>.
[RFC7231] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
DOI 10.17487/RFC7231, June 2014,
<https://doi.org/10.17487/RFC7231>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://doi.org/10.17487/RFC8174>.
[RFC8441] McManus, P., "Bootstrapping WebSockets with HTTP/2",
RFC 8441, DOI 10.17487/RFC8441, September 2018,
<https://doi.org/10.17487/RFC8441>.
[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://doi.org/10.17487/RFC9000>.
[SEMANTICS]
Fielding, R. T., Nottingham, M., and J. Reschke, "HTTP
Semantics", Work in Progress, Internet-Draft, draft-ietf-
httpbis-semantics-19, 12 September 2021,
<https://datatracker.ietf.org/doc/html/draft-ietf-httpbis-
semantics-19>.
Frindell, et al. Expires 28 April 2022 [Page 14]
Internet-Draft WebTransport-H3 October 2021
9.2. Informative References
[I-D.ietf-webtrans-http2]
Frindell, A., Kinnear, E., Pauly, T., Vasiliev, V., and G.
Xie, "WebTransport using HTTP/2", Work in Progress,
Internet-Draft, draft-ietf-webtrans-http2-01, 30 July
2021, <https://datatracker.ietf.org/doc/html/draft-ietf-
webtrans-http2-01>.
Authors' Addresses
Alan Frindell
Facebook
Email: afrind@fb.com
Eric Kinnear
Apple Inc.
Email: ekinnear@apple.com
Victor Vasiliev
Google
Email: vasilvv@google.com
Frindell, et al. Expires 28 April 2022 [Page 15]