Internet DRAFT - draft-ietf-webtrans-http3
draft-ietf-webtrans-http3
Network Working Group A. Frindell
Internet-Draft Facebook
Intended status: Standards Track E. Kinnear
Expires: 5 September 2024 Apple Inc.
V. Vasiliev
Google
4 March 2024
WebTransport over HTTP/3
draft-ietf-webtrans-http3-09
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 5 September 2024.
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Copyright Notice
Copyright (c) 2024 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
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provided without warranty as described in the Revised BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 3
2. Protocol Overview . . . . . . . . . . . . . . . . . . . . . . 3
3. Session Establishment . . . . . . . . . . . . . . . . . . . . 4
3.1. Establishing a Transport-Capable HTTP/3 Connection . . . 4
3.2. Extended CONNECT in HTTP/3 . . . . . . . . . . . . . . . 5
3.3. Creating a New Session . . . . . . . . . . . . . . . . . 5
3.4. Subprotocol Negotiation . . . . . . . . . . . . . . . . . 6
3.5. Limiting the Number of Simultaneous Sessions . . . . . . 7
3.6. Prioritization . . . . . . . . . . . . . . . . . . . . . 7
4. WebTransport Features . . . . . . . . . . . . . . . . . . . . 8
4.1. Unidirectional streams . . . . . . . . . . . . . . . . . 9
4.2. Bidirectional Streams . . . . . . . . . . . . . . . . . . 9
4.3. Resetting Data Streams . . . . . . . . . . . . . . . . . 10
4.4. Datagrams . . . . . . . . . . . . . . . . . . . . . . . . 11
4.5. Buffering Incoming Streams and Datagrams . . . . . . . . 11
4.6. Interaction with HTTP/3 GOAWAY frame . . . . . . . . . . 12
5. Session Termination . . . . . . . . . . . . . . . . . . . . . 12
6. Considerations for Future Versions . . . . . . . . . . . . . 14
6.1. Negotiating the Draft Version . . . . . . . . . . . . . . 14
7. Security Considerations . . . . . . . . . . . . . . . . . . . 14
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15
8.1. Upgrade Token Registration . . . . . . . . . . . . . . . 15
8.2. HTTP/3 SETTINGS Parameter Registration . . . . . . . . . 15
8.3. Frame Type Registration . . . . . . . . . . . . . . . . . 16
8.4. Stream Type Registration . . . . . . . . . . . . . . . . 16
8.5. HTTP/3 Error Code Registration . . . . . . . . . . . . . 16
8.6. Capsule Types . . . . . . . . . . . . . . . . . . . . . . 17
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 18
9.1. Normative References . . . . . . . . . . . . . . . . . . 18
9.2. Informative References . . . . . . . . . . . . . . . . . 19
Appendix A. Changelog . . . . . . . . . . . . . . . . . . . . . 20
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A.1. Changes between draft versions 02 and 07 . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 20
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, 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_.
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After the session is established, the peers can exchange data using
the following mechanisms:
* 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 Transport-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 8.1).
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.
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
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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]).
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". To use
WebTransport over HTTP/3, clients MUST send 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
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].
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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. Subprotocol Negotiation
WebTransport over HTTP/3 offers a subprotocol 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 WebTransport-Subprotocols-Available
header field in the CONNECT request, enumerating the possible
subprotocols. If the server receives such a header, it MAY include a
WebTransport-Subprotocol field in a successful (2xx) response. If it
does, the server SHALL include a single subprotocol from the client's
list in that field. Servers MAY reject the request if the client did
not include a suitable subprotocol.
Both WebTransport-Subprotocols-Available and WebTransport-Subprotocol
are Structured Fields [RFC8941]. WebTransport-Subprotocols-Available
is a List of Tokens, and WebTransport-Subprotocol is a Token. The
token in the WebTransport-Subprotocol response header field MUST be
one of the tokens listed in WebTransport-Subprotocols-Available 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.
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3.5. 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 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
HTTP_REQUEST_REJECTED status defined in [HTTP3].
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:
* The 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.
3.6. 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.
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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.
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 a valid ID for a
client-initiated bidirectional stream, the recipient MUST close the
connection with an H3_ID_ERROR error code.
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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 1).
Unidirectional Stream {
Stream Type (i) = 0x54,
Session ID (i),
Stream Body (..)
}
Figure 1: 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
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 2).
Bidirectional Stream {
Signal Value (i) = 0x41,
Session ID (i),
Stream Body (..)
}
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Figure 2: 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
codepoints adjacent to a reserved codepoint would map to two adjacent
WebTransport application error codepoints). An example pseudocode
can be seen in Figure 3.
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)
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Figure 3: 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 may result
in that data being discarded. Because of that, WebTransport
application error codes are best effort, as the WebTransport stack is
not always capable of associating the reset code with a session. The
only exception is the situation where there is only one session on a
given HTTP/3 connection, and no intermediaries between the client and
the server.
WebTransport implementations SHALL forward the error code for a
stream associated with a known session to the application that owns
that session; similarly, the intermediaries SHALL reset the streams
with corresponding error code when receiving a reset from the peer.
If a WebTransport implementation intentionally allows only one
session over a given HTTP/3 connection, it SHALL forward the error
codes within WebTransport application error code range to the
application that owns the only session on that connection.
4.4. Datagrams
Datagrams can be sent using HTTP Datagrams. The WebTransport
datagram payload is sent unmodified in the "HTTP Datagram Payload"
field of an HTTP Datagram (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.
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.
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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 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 if the stream identifier is equal to or greater
than the indicated stream ID.
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
}
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.
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.
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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:
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 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 close the stream upon receiving a FIN. 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
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SHOULD wait until all of the data on the CONNECT stream is
acknowledged 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.
6. 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 8.1).
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 8.2).
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.
6.1. 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.
7. Security Considerations
WebTransport over HTTP/3 satisfies all of the security requirements
imposed by [OVERVIEW] on WebTransport protocols, thus providing a
secure framework for client-server communication in cases when the
client is potentially untrusted.
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WebTransport over HTTP/3 requires explicit opt-in through the use of
an HTTP/3 setting; this avoids potential protocol confusion attacks
by ensuring the HTTP/3 server explicitly supports it. It also
requires the use of the Origin header, providing the server with the
ability to deny access to Web-based clients that do not originate
from a trusted origin.
Just like HTTP traffic going over HTTP/3, WebTransport pools traffic
to different origins within a single connection. Different origins
imply different trust domains, meaning that the implementations have
to treat each transport as potentially hostile towards others on the
same connection. One potential attack is a resource exhaustion
attack: since all of the transports share both congestion control and
flow control context, a single client aggressively using up those
resources can cause other transports to stall. The user agent thus
SHOULD implement a fairness scheme that ensures that each transport
within connection gets a reasonable share of controlled resources;
this applies both to sending data and to opening new streams.
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.
8. IANA Considerations
8.1. Upgrade Token Registration
The following entry is added to the "Hypertext Transfer Protocol
(HTTP) Upgrade Token Registry" registry established by Section 16.7
of [HTTP].
The "webtransport" label identifies HTTP/3 used as a protocol for
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]:
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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
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 is reserved for the purpose of avoiding
collision with WebTransport HTTP/3 extensions:
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]:
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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.
8.6. 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)
Notes: None
The DRAIN_WEBTRANSPORT_SESSION capsule.
Value: 0x78ae
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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
9. References
9.1. Normative References
[HTTP] Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,
Ed., "HTTP Semantics", STD 97, RFC 9110,
DOI 10.17487/RFC9110, June 2022,
<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-
07, 4 March 2024, <https://datatracker.ietf.org/doc/html/
draft-ietf-webtrans-overview-07>.
[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>.
[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>.
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Internet-Draft WebTransport-H3 March 2024
[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
DOI 10.17487/RFC6454, December 2011,
<https://www.rfc-editor.org/rfc/rfc6454>.
[RFC6585] Nottingham, M. and R. Fielding, "Additional HTTP Status
Codes", RFC 6585, DOI 10.17487/RFC6585, April 2012,
<https://www.rfc-editor.org/rfc/rfc6585>.
[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>.
[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>.
[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>.
[RFC9220] Hamilton, R., "Bootstrapping WebSockets with HTTP/3",
RFC 9220, DOI 10.17487/RFC9220, June 2022,
<https://www.rfc-editor.org/rfc/rfc9220>.
9.2. 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-http2-08,
4 March 2024, <https://datatracker.ietf.org/doc/html/
draft-ietf-webtrans-http2-08>.
[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>.
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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
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)
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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