WebTransport over HTTP/3
draft-ietf-webtrans-http3-15
| Document | Type | Active Internet-Draft (webtrans WG) | |
|---|---|---|---|
| Authors | Alan Frindell , Eric Kinnear , Victor Vasiliev | ||
| Last updated | 2026-03-08 (Latest revision 2026-03-02) | ||
| Replaces | draft-vvv-webtransport-http3 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Intended RFC status | Proposed Standard | ||
| Formats | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | In WG Last Call | |
| Document shepherd | (None) | ||
| IESG | IESG state | I-D Exists | |
| Consensus boilerplate | Yes | ||
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | (None) |
draft-ietf-webtrans-http3-15
WEBTRANS A. Frindell
Internet-Draft Facebook
Intended status: Standards Track E. Kinnear
Expires: 3 September 2026 Apple Inc.
V. Vasiliev
Google
2 March 2026
WebTransport over HTTP/3
draft-ietf-webtrans-http3-15
Abstract
WebTransport [OVERVIEW] is a protocol framework that enables
application clients constrained by the Web security model to
communicate with a remote application 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.
About This Document
This note is to be removed before publishing as an RFC.
The latest revision of this draft can be found at https://ietf-wg-
webtrans.github.io/draft-ietf-webtrans-http3/#go.draft-ietf-webtrans-
http3.html. Status information for this document may be found at
https://datatracker.ietf.org/doc/draft-ietf-webtrans-http3/.
Discussion of this document takes place on the WebTransport Working
Group mailing list (mailto:webtransport@ietf.org), which is archived
at https://mailarchive.ietf.org/arch/browse/webtransport/. Subscribe
at https://www.ietf.org/mailman/listinfo/webtransport/.
Source for this draft and an issue tracker can be found at
https://github.com/ietf-wg-webtrans/draft-ietf-webtrans-http3.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on 3 September 2026.
Copyright Notice
Copyright (c) 2026 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/
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Please review these documents carefully, as they describe your rights
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 3
2. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. QUIC, WebTransport, and HTTP/3 . . . . . . . . . . . . . 4
2.1.1. Minimizing Implementation Complexity . . . . . . . . 5
2.1.2. Capsule-Based WebTransport over HTTP/3 . . . . . . . 5
2.2. Protocol Overview . . . . . . . . . . . . . . . . . . . . 6
3. Session Establishment . . . . . . . . . . . . . . . . . . . . 7
3.1. Establishing a WebTransport-Capable HTTP/3 Connection . . 7
3.2. Creating a New Session . . . . . . . . . . . . . . . . . 9
3.3. Application Protocol Negotiation . . . . . . . . . . . . 10
3.4. Prioritization . . . . . . . . . . . . . . . . . . . . . 11
4. WebTransport Features . . . . . . . . . . . . . . . . . . . . 12
4.1. Transport Properties . . . . . . . . . . . . . . . . . . 13
4.2. Unidirectional streams . . . . . . . . . . . . . . . . . 13
4.3. Bidirectional Streams . . . . . . . . . . . . . . . . . . 13
4.4. Resetting Data Streams . . . . . . . . . . . . . . . . . 14
4.5. Datagrams . . . . . . . . . . . . . . . . . . . . . . . . 15
4.6. Buffering Incoming Streams and Datagrams . . . . . . . . 16
4.7. Interaction with the HTTP/3 GOAWAY frame . . . . . . . . 16
4.8. Use of Keying Material Exporters . . . . . . . . . . . . 17
5. Flow Control . . . . . . . . . . . . . . . . . . . . . . . . 18
5.1. Negotiating the Use of Flow Control . . . . . . . . . . . 18
5.2. Limiting the Number of Simultaneous Sessions . . . . . . 19
5.3. Limiting the Number of Streams Within a Session . . . . . 19
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5.4. Data Limits . . . . . . . . . . . . . . . . . . . . . . . 20
5.5. Flow Control SETTINGS . . . . . . . . . . . . . . . . . . 21
5.5.1. SETTINGS_WT_INITIAL_MAX_STREAMS_UNI . . . . . . . . . 21
5.5.2. SETTINGS_WT_INITIAL_MAX_STREAMS_BIDI . . . . . . . . 21
5.5.3. SETTINGS_WT_INITIAL_MAX_DATA . . . . . . . . . . . . 22
5.6. Flow Control Capsules . . . . . . . . . . . . . . . . . . 22
5.6.1. Flow Control and Intermediaries . . . . . . . . . . . 22
5.6.2. WT_MAX_STREAMS Capsule . . . . . . . . . . . . . . . 23
5.6.3. WT_STREAMS_BLOCKED Capsule . . . . . . . . . . . . . 24
5.6.4. WT_MAX_DATA Capsule . . . . . . . . . . . . . . . . . 25
5.6.5. WT_DATA_BLOCKED Capsule . . . . . . . . . . . . . . . 26
6. Session Termination . . . . . . . . . . . . . . . . . . . . . 26
7. Considerations for Future Versions . . . . . . . . . . . . . 28
7.1. Negotiating the Draft Version . . . . . . . . . . . . . . 28
8. Security Considerations . . . . . . . . . . . . . . . . . . . 29
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 30
9.1. Upgrade Token Registration . . . . . . . . . . . . . . . 30
9.2. HTTP/3 SETTINGS Parameter Registration . . . . . . . . . 30
9.3. Frame Type Registration . . . . . . . . . . . . . . . . . 31
9.4. Stream Type Registration . . . . . . . . . . . . . . . . 32
9.5. HTTP/3 Error Code Registration . . . . . . . . . . . . . 32
9.6. Capsule Types . . . . . . . . . . . . . . . . . . . . . . 34
9.7. Protocol Negotiation HTTP Header Fields . . . . . . . . . 36
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 36
10.1. Normative References . . . . . . . . . . . . . . . . . . 36
10.2. Informative References . . . . . . . . . . . . . . . . . 38
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 39
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, or sessions, 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.
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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. An application client is user or
developer-provided code, often untrusted, that utilizes the interface
offered by the WebTransport client to communicate with an application
server. The application server uses the interface offered by the
WebTransport server to accept incoming WebTransport sessions.
2. Overview
2.1. QUIC, WebTransport, and HTTP/3
QUIC version 1 [RFC9000] is a secure transport protocol with flow
control and congestion control. QUIC supports application data
exchange via streams; reliable and ordered byte streams that can be
multiplexed. Stream independence can mitigate head-of-line blocking.
While QUIC provides streams as a transport service, it is
unopinionated about their usage. The applicability of streams is
described by section 4 of [RFC9308].
HTTP is an application-layer protocol, defined by "HTTP Semantics"
[HTTP]. HTTP/3 is the application mapping for QUIC, defined in
[RFC9114]. It describes how QUIC streams are used to carry control
data or HTTP request and response message sequences in the form of
frames and describes details of stream and connection lifecycle
management. HTTP/3 offers two features in addition to HTTP
Semantics: QPACK header compression [RFC9208] and Server Push
Section 4.6 of [RFC9114].
WebTransport session establishment involves interacting at the HTTP
layer with a resource. For Web user agents and other WebTransport
clients, this interaction is important for security reasons,
especially to ensure that the resource is willing to use
WebTransport.
Although WebTransport requires HTTP for its handshake, when HTTP/3 is
in use, HTTP is not used for anything else related to an established
session. Instead, QUIC streams begin with a sequence of header bytes
that links them to the established session. The remainder of the
stream is the body, which carries the payload supplied by the
application using WebTransport. This process is similar to
WebSockets over HTTP/1.1 [ORIGIN], where access to the underlying
byte stream is enabled after both sides have completed an initial
handshake.
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The layering of QUIC, HTTP/3, and WebTransport is shown in Figure 1.
Once a WebTransport session is established, applications have nearly
direct access to QUIC.
,--------------------------------,
| WebTransport |
,----------------,---------------,
| HTTP Semantics | |
| and | |
| Session Setup | Nearly direct |
,----------------, |
| HTTP/3 | |
,----------------`---------------,
| QUIC |
`--------------------------------'
Figure 1: WebTransport Layering
2.1.1. Minimizing Implementation Complexity
WebTransport has minimal interaction with HTTP and HTTP/3. Clients
and servers can constrain their use of features to only those
required to complete a WebTransport handshake:
* Generating/parsing the request method, host, path, protocol,
optional Origin header field, and perhaps some extra header
fields.
* Generating/parsing the response status code and possibly some
extra header fields.
A WebTransport endpoint, whether a client or a server, can likely
perform several of its HTTP-level requirements using bytestring
comparisons.
While HTTP/3 encodes HTTP messages using QPACK, this complexity can
be minimized. When receiving, a WebTransport endpoint can disable
dynamic decompression entirely but must always support static
decompression and Huffman decoding. When sending, endpoints can opt
to never use dynamic compression, static compression, or Huffman
encoding.
2.1.2. Capsule-Based WebTransport over HTTP/3
WebTransport over HTTP/3 as defined in this document provides the
best performance by using native QUIC streams and datagrams.
Endpoints SHOULD always use this protocol when using WebTransport
over an HTTP/3 connection.
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However, it is also possible to use WebTransport over a single HTTP/3
stream using the capsule-based protocol defined in [WEBTRANS-H2].
The two protocols are distinguished by their upgrade tokens: this
document uses the "webtransport-h3" token (Section 9.1), while
[WEBTRANS-H2] uses the "webtransport" token.
The capsule-based protocol can be useful for intermediaries that
proxy WebTransport sessions between HTTP/2 and HTTP/3 connections, as
it avoids the need to translate between the two wire formats. It can
also be useful in deployment environments such as data centers where
existing routing infrastructure supports forwarding streams but does
not support the HTTP/3 extensions required by this document.
Endpoints that use the capsule-based protocol over HTTP/3 lose the
benefits of stream independence, as all WebTransport streams within a
session share a single QUIC stream and are subject to head-of-line
blocking. Datagrams sent using the capsule-based protocol are also
retransmitted by QUIC, and therefore do not provide unreliable
delivery.
2.2. Protocol Overview
WebTransport servers in general are identified by a pair of authority
value and path value (defined in [RFC3986] Sections 3.2 and 3.3
respectively).
When an HTTP/3 connection is established, the server sends a
SETTINGS_WT_ENABLED setting to indicate support for WebTransport over
HTTP/3. This process also negotiates the use of additional HTTP/3
extensions to enable both endpoints to open WebTransport streams.
WebTransport sessions are initiated inside a given HTTP/3 connection
by the client, who sends an extended CONNECT request [RFC9220]. If
the server accepts the request, a WebTransport session is
established. The resulting stream will be further referred to as a
_CONNECT stream_, and its stream ID is used to uniquely identify a
given WebTransport session within the connection. The ID of the
CONNECT stream that established a given WebTransport session will be
further referred to as a _Session ID_.
After the session is established, the endpoints 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.
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* 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.
* Both client and server can send datagrams using HTTP Datagrams
[HTTP-DATAGRAM].
A WebTransport session is terminated when the CONNECT stream that
created it is closed.
3. Session Establishment
3.1. Establishing a WebTransport-Capable HTTP/3 Connection
A WebTransport-Capable HTTP/3 connection requires the server to
signal support for WebTransport over HTTP/3 using a setting. Clients
also signal support by using the "webtransport-h3" upgrade token in
extended CONNECT requests when establishing sessions (see
Section 9.1).
This document defines a SETTINGS_WT_ENABLED setting that WebTransport
servers use to indicate their support for WebTransport. The default
value for the SETTINGS_WT_ENABLED setting is "0", meaning that the
server does not support WebTransport. Clients MUST NOT attempt to
establish WebTransport sessions until they have received the setting
indicating WebTransport support from the server.
WebTransport over HTTP/3 uses extended CONNECT in HTTP/3 as described
in [RFC9220], which defines the SETTINGS_ENABLE_CONNECT_PROTOCOL
setting.
WebTransport over HTTP/3 requires support for HTTP/3 datagrams and
the Capsule Protocol, and both the client and the server indicate
support for HTTP/3 datagrams by sending a SETTINGS_H3_DATAGRAM
setting value set to 1 in their SETTINGS frame (see Section 2.1.1 of
[HTTP-DATAGRAM]).
WebTransport over HTTP/3 also requires support for QUIC datagrams.
To indicate support, both the client and the server send a
max_datagram_frame_size transport parameter with a value greater than
0 (see Section 3 of [QUIC-DATAGRAM]).
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WebTransport over HTTP/3 relies on the RESET_STREAM_AT frame defined
in [RESET-STREAM-AT]. To indicate support, both the client and the
server enable the extension by sending an empty reset_stream_at
transport parameter as described in Section 3 of [RESET-STREAM-AT].
In summary, servers supporting WebTransport over HTTP/3 send:
* A SETTINGS_WT_ENABLED setting with a value greater than "0"
* A SETTINGS_ENABLE_CONNECT_PROTOCOL setting with a value of "1"
* A SETTINGS_H3_DATAGRAM setting with a value of 1
* A max_datagram_frame_size transport parameter with a value greater
than 0
* An empty reset_stream_at transport parameter
Clients supporting WebTransport over HTTP/3 send:
* A SETTINGS_H3_DATAGRAM setting with a value of 1
* A max_datagram_frame_size transport parameter with a value greater
than 0
* An empty reset_stream_at transport parameter
[[RFC editor: please remove the following paragraph before
publication.]]
For draft versions of WebTransport only, clients MUST also send
SETTINGS_WT_ENABLED with the draft-specific codepoint to allow the
server to identify the client's supported version; see Section 7.1.
Servers should note that CONNECT requests to establish new
WebTransport sessions, in addition to other messages, can arrive
before the client's SETTINGS are received (see Section 4.6). If the
server receives SETTINGS that do not have correct values for every
required setting, or transport parameters that do not have correct
values for every required transport parameter, the server MUST treat
all established and newly incoming WebTransport sessions as
malformed, as described in Section 4.1.2 of [HTTP3].
A client MUST NOT establish WebTransport sessions if the server's
SETTINGS do not have correct values for every required setting or if
the server's transport parameters do not have correct values for
every required transport parameter. If a client does not wish to use
the connection for purposes other than WebTransport when the
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requirements for WebTransport are not met, the client MAY close the
HTTP/3 connection with a WT_REQUIREMENTS_NOT_MET error code to aid in
debugging.
[[RFC editor: please remove the following paragraph before
publication.]]
For draft versions of WebTransport only, the server MUST NOT process
any incoming WebTransport requests until the client's SETTINGS have
been received; see Section 7.1.
3.2. Creating a New Session
As WebTransport sessions are established over HTTP/3, they are
identified using the https URI scheme (Section 4.2.2 of [HTTP]).
In order to create a new WebTransport session, a WebTransport client
sends an HTTP extended CONNECT request. In this request:
* The :protocol pseudo-header field ([RFC8441]) MUST be set to
webtransport-h3.
* The :scheme field MUST be https.
* Both the :authority and the :path value MUST be set; these fields
identify the desired WebTransport server resource.
* 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-h3, 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].
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.
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The server may reply with a 3xx response, indicating a redirection
(Section 15.4 of [HTTP]). The WebTransport client MUST NOT
automatically follow such redirects, as it potentially could have
already sent data for the WebTransport session in question; it MAY
notify the application 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, or other initial flow control
values, from the values negotiated during the previous session; such
change would be deemed incompatible, and MUST result in a
H3_SETTINGS_ERROR connection error.
The "webtransport-h3" 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.
3.3. Application Protocol Negotiation
WebTransport over HTTP/3 offers a protocol negotiation mechanism,
similar to the TLS Application-Layer Protocol Negotiation (ALPN)
extension [RFC7301]; the intent is to simplify porting existing
protocols that use QUIC and rely on this functionality.
The client MAY include a WT-Available-Protocols header field in the
CONNECT request. The WT-Available-Protocols field enumerates the
possible protocols in preference order, with the most preferred
protocol listed first. If the server receives such a header, it MAY
include a WT-Protocol field in a successful (2xx) response. If it
does, the server MUST include a single choice from the client's list
in that field. Servers MAY reject the request if the client did not
include a suitable protocol.
Both WT-Available-Protocols and WT-Protocol are Structured Fields
[FIELDS]. WT-Available-Protocols is a List. WT-Protocol is defined
as an Item. In both cases, the only valid value type is a String.
Any value type other than String MUST be treated as an error that
causes the entire field to be ignored as recommended in [FIELDS],
allowing application protocol negotiation to remain optional. No
semantics are defined for parameters on either field; parameters MUST
be ignored.
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A server that requires application protocol negotiation MAY reject
the session if the WT-Available-Protocols header field is absent or
if it is malformed and therefore ignored.
A client that requires application protocol negotiation MUST close
the WebTransport session with a WT_ALPN_ERROR error code if the
server does not include a WT-Protocol header field, or if it is
malformed and therefore ignored, in a successful response.
If the client sends a WT-Available-Protocols header field and the
server responds with a WT-Protocol header field, the value in the WT-
Protocol response header field MUST be one of the values listed in
WT-Available-Protocols of the request. If the client receives a WT-
Protocol value that was not included in its WT-Available-Protocols
list, the client MUST close the WebTransport session with a
WT_ALPN_ERROR error code.
The semantics of individual values used in WT-Available-Protocols and
WT-Protocol are determined by the WebTransport resource in question
and are not required to be registered in IANA's "ALPN Protocol IDs"
registry.
3.4. Prioritization
WebTransport sessions are initiated using extended CONNECT. While
Section 11 of [RFC9218] describes how extensible priorities can be
applied to data sent on a CONNECT stream, WebTransport extends the
types of data that are exchanged in relation to the request and
response, which requires additional considerations.
WebTransport CONNECT requests and responses MAY contain the Priority
header field (Section 5 of [RFC9218]); clients MAY reprioritize by
sending PRIORITY_UPDATE frames (Section 7 of [RFC9218]). In
extension to [RFC9218], it is RECOMMENDED that clients and servers
apply the scheduling guidance in both Section 9 of [RFC9218] and
Section 10 of [RFC9218] for all data that they send in the enclosing
WebTransport session, including Capsules, WebTransport streams and
datagrams. WebTransport does not provide any priority signaling
mechanism for streams and datagrams within a WebTransport session;
such mechanisms can be defined by application protocols using
WebTransport. It is RECOMMENDED that such mechanisms only affect
scheduling within a session and not scheduling of other data on the
same HTTP/3 connection.
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The client/server priority merging guidance in Section 8 of [RFC9218]
also applies to WebTransport sessions. 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, all of which can be 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, on a session for which it has
sent the CONNECT request, 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.
Session IDs are derived from the stream ID of the CONNECT stream that
established the session and therefore MUST always correspond to a
client-initiated bidirectional stream, as defined in Section 2.1 of
[RFC9000]. If an endpoint receives a session ID on a unidirectional
stream, bidirectional stream, or datagram that does not correspond to
a client-initiated bidirectional stream ID, the endpoint MUST close
the connection with an H3_ID_ERROR error code. Session IDs that
correspond to closed sessions are not considered invalid for the
purposes of this check; endpoints handle data for closed sessions as
described in Section 6.
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4.1. Transport Properties
The WebTransport framework [OVERVIEW] defines a set of optional
transport properties that clients can use to determine the presence
of features which might allow additional optimizations beyond the
common set of properties available via all WebTransport protocols.
Below are details about support in WebTransport over HTTP/3 for the
properties defined by the WebTransport framework.
Unreliable Delivery: WebTransport over HTTP/3 supports unreliable
delivery. Resetting a stream results in lost stream data no
longer being retransmitted. WebTransport over HTTP/3 also
supports datagrams, which are not retransmitted.
Pooling: WebTransport over HTTP/3 provides optional support for
pooling. Endpoints that do not support pooling can reply to
CONNECT requests with a header indicating a rate limit policy with
a quota of "1" ([I-D.ietf-httpapi-ratelimit-headers]).
4.2. Unidirectional streams
WebTransport endpoints can initiate unidirectional streams. The
HTTP/3 unidirectional stream type SHALL be 0x54. The body of the
stream SHALL be the stream type, followed by the session ID, encoded
as a variable-length integer, followed by the user-specified stream
data (Figure 2).
Unidirectional Stream {
Stream Type (i) = 0x54,
Session ID (i),
User-Specified Stream Data (..)
}
Figure 2: Unidirectional WebTransport stream format
4.3. 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 to use
alternative request stream rules. Once a client receives settings
indicating WebTransport support (Section 3.1), it MUST send a special
signal value, encoded as a variable-length integer, as the first
bytes of each bidirectional WebTransport stream it initiates to
indicate how the remaining bytes on the stream are used.
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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 MUST
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.
Clients and servers use the signal value 0x41 to open a bidirectional
WebTransport stream. Following this is the associated session ID,
encoded as a variable-length integer; the rest of the stream is the
application payload of the WebTransport stream (Figure 3).
Bidirectional Stream {
Signal Value (i) = 0x41,
Session ID (i),
Stream Body (..)
}
Figure 3: Bidirectional WebTransport stream format
This document reserves the special signal value 0x41 as a WT_STREAM
frame type. While it is registered as an HTTP/3 frame type to avoid
collisions, WT_STREAM lacks length and is not a proper HTTP/3 frame;
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 WT_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.
4.4. 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 MUST 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 MUST remap those error
codes into the error range reserved for WT_APPLICATION_ERROR, where
0x00000000 corresponds to 0x52e4a40fa8db, and 0xffffffff corresponds
to 0x52e5ac983162. Note that there are codepoints 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.,
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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 4.
first = 0x52e4a40fa8db
last = 0x52e5ac983162
def webtransport_code_to_http_code(n):
return first + n + floor(n / 0x1e)
def http_code_to_webtransport_code(h):
assert(first <= h <= last)
assert((h - 0x21) % 0x1f != 0)
shifted = h - first
return shifted - floor(shifted / 0x1f)
Figure 4: Pseudocode for converting between WebTransport
application errors and HTTP/3 error codes
WebTransport data streams are associated with sessions through a
header at the beginning of the stream; resetting a stream might
result in that data being discarded when using a RESET_STREAM frame.
To prevent this, WebTransport implementations MUST use the
RESET_STREAM_AT frame [RESET-STREAM-AT] with a Reliable Size set to
at least the size of the WebTransport header when resetting a
WebTransport data stream. This ensures reliable delivery of the ID
field associating the data stream with a WebTransport session.
WebTransport endpoints MUST forward the error code for a stream
associated with a known session to the application that owns that
session; similarly, intermediaries MUST reset such streams with a
corresponding error code when receiving a reset from their peer. If
a RESET_STREAM or STOP_SENDING frame is received with an error code
outside the range reserved for WT_APPLICATION_ERROR, the stream is
still considered reset, but the error code is not mapped to a
WebTransport application error code. The WebTransport implementation
SHOULD deliver this to the application as a stream reset with no
application error code.
4.5. Datagrams
Datagrams can be sent using HTTP Datagrams. The WebTransport
datagram payload is sent unmodified in the "HTTP Datagram Payload"
field of an HTTP Datagram (Section 2.1 of [HTTP-DATAGRAM]). Note
that the payload field directly follows the Quarter Stream ID field,
which is at the start of the QUIC DATAGRAM frame payload and refers
to the CONNECT stream that established the WebTransport session.
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4.6. Buffering Incoming Streams and Datagrams
In WebTransport over HTTP/3, the client MUST wait for receipt of the
server's SETTINGS frame before establishing any WebTransport sessions
by sending CONNECT requests using the WebTransport upgrade token (see
Section 3.1). This ensures that the client will always know what
versions of WebTransport can be used on a given HTTP/3 connection.
Clients can, however, send a SETTINGS frame, multiple WebTransport
CONNECT requests, WebTransport data streams, and WebTransport
datagrams all within a single flight. As those can arrive out of
order, a WebTransport server could be put into a situation where it
receives a stream or a datagram without a corresponding session.
Similarly, a client may receive a server-initiated stream or a
datagram before receiving the CONNECT response headers from the
server.
To handle this case, WebTransport endpoints SHOULD buffer streams and
datagrams until they can be associated with an established session.
To avoid resource exhaustion, 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 WT_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.
4.7. Interaction with the HTTP/3 GOAWAY frame
HTTP/3 defines a graceful shutdown mechanism (Section 5.2 of [HTTP3])
that allows a peer to send a GOAWAY frame indicating that it will no
longer accept any new incoming requests or pushes.
A client receiving GOAWAY cannot initiate CONNECT requests for new
WebTransport sessions on that HTTP/3 connection; it must open a new
HTTP/3 connection to initiate new WebTransport sessions with the same
peer.
An HTTP/3 GOAWAY frame is also a signal to applications to initiate
shutdown for all WebTransport sessions. To shut down a single
WebTransport session, either endpoint can send a WT_DRAIN_SESSION
(0x78ae) capsule.
WT_DRAIN_SESSION Capsule {
Type (i) = WT_DRAIN_SESSION,
Length (i) = 0
}
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Figure 5: WT_DRAIN_SESSION Capsule Format
After sending or receiving either a WT_DRAIN_SESSION capsule or a
HTTP/3 GOAWAY frame, an endpoint MAY continue using the session and
MAY open new WebTransport streams. The signal is intended for the
application using WebTransport, which is expected to attempt to
gracefully terminate the session as soon as possible.
The WT_DRAIN_SESSION capsule is useful when an end-to-end
WebTransport session passes through an intermediary. For example,
when the backend shuts down, it sends a GOAWAY to the intermediary.
The intermediary can convert this signal to a WT_DRAIN_SESSION
capsule on the client-facing session, without impacting other
requests or sessions carried on that connection.
4.8. Use of Keying Material Exporters
WebTransport over HTTP/3 supports the use of TLS keying material
exporters Section 7.5 of [RFC8446]. Since the underlying QUIC
connection may be shared by multiple WebTransport sessions,
WebTransport defines a mechanism for deriving a TLS exporter that
separates keying material for different sessions. If the application
requests an exporter for a given WebTransport session with a
specified label and context, the resulting exporter SHALL be a TLS
exporter as defined in Section 7.5 of [RFC8446] with the label set to
"EXPORTER-WebTransport" and the context set to the serialization of
the "WebTransport Exporter Context" struct as defined below.
WebTransport Exporter Context {
WebTransport Session ID (64),
WebTransport Application-Supplied Exporter Label Length (8),
WebTransport Application-Supplied Exporter Label (8..),
WebTransport Application-Supplied Exporter Context Length (8),
WebTransport Application-Supplied Exporter Context (..)
}
Figure 6: WebTransport Exporter Context struct
A TLS exporter API might permit the context field to be omitted. In
this case, as with TLS 1.3, the WebTransport Application-Supplied
Exporter Context becomes zero-length if omitted.
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5. Flow Control
Flow control governs the amount of resources that can be consumed or
data that can be sent. When using WebTransport over HTTP/3,
endpoints can limit the number of sessions that a peer can create on
a single HTTP/3 connection and the number of streams that a peer can
create within a session. Endpoints can also limit the amount of data
that can be consumed by each session and by each stream within a
session.
WebTransport over HTTP/3 provides a connection-level limit that
governs the number of sessions that can be created on an HTTP/3
connection (see Section 5.2). It also provides session-level limits
that govern the number of streams that can be created in a session
and limit the amount of data that can be exchanged across all streams
in each session (see Section 5.3).
The underlying QUIC connection provides connection and stream level
flow control. The QUIC connection data limit defines the total
amount of data that can be sent across all WebTransport sessions and
other non-WebTransport streams. A QUIC stream's data limit controls
the amount of data that can be sent on that stream, WebTransport or
otherwise (see Section 4 of [RFC9000]).
5.1. Negotiating the Use of Flow Control
A WebTransport endpoint that allows a WebTransport session to share
an underlying transport connection with other WebTransport sessions
MUST enable flow control. This prevents an application from
consuming excessive resources on a single session and starving
traffic for other sessions (see Section 8).
Flow control is enabled when both endpoints declare their intent to
use flow control by taking any of the following actions:
* Sending SETTINGS_WT_INITIAL_MAX_STREAMS_UNI with any value other
than "0".
* Sending SETTINGS_WT_INITIAL_MAX_STREAMS_BIDI with any value other
than "0".
* Sending SETTINGS_WT_INITIAL_MAX_DATA with any value other than
"0".
If both endpoints take at least one of these actions, flow control is
enabled, and the limits described in the entirety of Section 5 apply.
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Flow control can be enabled regardless of the number of WebTransport
sessions a server supports.
If flow control is not enabled, clients MUST NOT attempt to establish
more than one simultaneous WebTransport session. A server that
receives more than one session on an underlying transport connection
when flow control is not enabled MUST reset the excessive CONNECT
streams with a H3_REQUEST_REJECTED status (see Section 5.2).
Also, if flow control is not enabled, an endpoint MUST ignore receipt
of any flow control capsules (see Section 5.6), since the peer might
not have received SETTINGS at the time they were sent or packets
might have been reordered.
5.2. Limiting the Number of Simultaneous Sessions
Servers SHOULD limit the rate of incoming WebTransport sessions on
HTTP/3 connections to prevent excessive consumption of resources. To
do so, they have multiple mechanisms available:
* The H3_REQUEST_REJECTED error code defined in Section 8.1 of
[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.
WebTransport servers can use rate limit header fields in responses to
CONNECT requests to signal quota policies and service limits to
WebTransport clients (see [I-D.ietf-httpapi-ratelimit-headers]).
This provides hints to clients about how many sessions they can
reasonably expect to be able to open. An endpoint that does not
support pooling and flow control MUST NOT accept more than one
incoming WebTransport session at a time.
5.3. Limiting the Number of Streams Within a Session
The WT_MAX_STREAMS capsule (Section 5.6.2) establishes a limit on the
number of streams within a WebTransport session. Like the QUIC
MAX_STREAMS frame (Section 19.11 of [RFC9000]), this capsule has two
types that provide separate limits for unidirectional and
bidirectional streams that a peer initiates.
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Note that the CONNECT stream for the session is not included in
either the bidirectional or the unidirectional stream limits; the
number of CONNECT streams a client can open is limited by QUIC flow
control's stream limits and any rate limit that a WebTransport server
enforces.
The session-level stream limit applies in addition to the QUIC
MAX_STREAMS frame, which provides a connection-level stream limit.
New streams can only be created within the session if both the
stream- and the connection-level limit permit, see Section 4.6 of
[RFC9000] for details on how QUIC stream limits are applied.
Unlike the QUIC MAX_STREAMS frame, there is no simple relationship
between the value in this frame and stream IDs in QUIC STREAM frames.
This especially applies if there are other users of streams on the
connection.
The WT_STREAMS_BLOCKED capsule (Section 5.6.3) can be sent to
indicate that an endpoint was unable to create a stream due to the
session-level stream limit.
Note that enforcing this limit requires reliable resets for stream
headers so that both endpoints can agree on the number of streams
that are open.
5.4. Data Limits
The WT_MAX_DATA capsule (Section 5.6.4) establishes a limit on the
amount of data that can be sent within a WebTransport session. This
limit counts all data that is sent on streams of the corresponding
type, excluding the stream header (see Section 4.2 and Section 4.3).
The stream header is excluded from this limit so that this limit does
not prevent the sending of information that is essential in linking
new streams to a specific WebTransport session.
For streams that were reset, implementing WT_MAX_DATA requires that
the QUIC stack provide the WebTransport implementation with
information about the final size of streams; see Section 4.5 of
[RFC9000]. This guarantees that both endpoints agree on how much
WebTransport session flow control credit was consumed by the sender
on that stream.
The WT_DATA_BLOCKED capsule (Section 5.6.5) can be sent to indicate
that an endpoint was unable to send data due to a limit set by the
WT_MAX_DATA capsule.
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Because WebTransport over HTTP/3 uses a native QUIC stream for each
WebTransport stream, per-stream data limits are provided by QUIC
natively (see Section 4.1 of [RFC9000]). The WT_MAX_STREAM_DATA and
WT_STREAM_DATA_BLOCKED capsules (Sections 6.6 and 6.9 of
[I-D.ietf-webtrans-http2]) are not used and so are prohibited.
Endpoints MUST treat receipt of a WT_MAX_STREAM_DATA or a
WT_STREAM_DATA_BLOCKED capsule as a session error.
5.5. Flow Control SETTINGS
Initial flow control limits can be exchanged via HTTP/3 SETTINGS
(Section 9.2) by providing non-zero values for
* WT_MAX_STREAMS via SETTINGS_WT_INITIAL_MAX_STREAMS_UNI and
SETTINGS_WT_INITIAL_MAX_STREAMS_BIDI
* WT_MAX_DATA via SETTINGS_WT_INITIAL_MAX_DATA
5.5.1. SETTINGS_WT_INITIAL_MAX_STREAMS_UNI
The SETTINGS_WT_INITIAL_MAX_STREAMS_UNI setting indicates the initial
value for the unidirectional max stream limit, otherwise communicated
by the WT_MAX_STREAMS capsule (see Section 5.6.2). The default value
for the SETTINGS_WT_INITIAL_MAX_STREAMS_UNI setting is "0",
indicating that the endpoint needs to send WT_MAX_STREAMS capsules on
each individual WebTransport session before its peer is allowed to
create any unidirectional streams within that session.
Note that this limit applies to all WebTransport sessions that use
the HTTP/3 connection on which this SETTING is sent.
5.5.2. SETTINGS_WT_INITIAL_MAX_STREAMS_BIDI
The SETTINGS_WT_INITIAL_MAX_STREAMS_BIDI setting indicates the
initial value for the bidirectional max stream limit, otherwise
communicated by the WT_MAX_STREAMS capsule (see Section 5.6.2). The
default value for the SETTINGS_WT_INITIAL_MAX_STREAMS_BIDI setting is
"0", indicating that the endpoint needs to send WT_MAX_STREAMS
capsules on each individual WebTransport session before its peer is
allowed to create any bidirectional streams within that session.
Note that this limit applies to all WebTransport sessions that use
the HTTP/3 connection on which this SETTING is sent.
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5.5.3. SETTINGS_WT_INITIAL_MAX_DATA
The SETTINGS_WT_INITIAL_MAX_DATA setting indicates the initial value
for the session data limit, otherwise communicated by the WT_MAX_DATA
capsule (see Section 5.6.4). The default value for the
SETTINGS_WT_INITIAL_MAX_DATA setting is "0", indicating that the
endpoint needs to send a WT_MAX_DATA capsule within each session
before its peer is allowed to send any stream data within that
session.
Note that this limit applies to all WebTransport sessions that use
the HTTP/3 connection on which this SETTING is sent.
5.6. Flow Control Capsules
WebTransport over HTTP/3 uses several capsules for flow control, and
all of these capsules define special intermediary handling as
described in Section 3.2 of [HTTP-DATAGRAM]. These capsules,
referred to as the "flow control capsules", are WT_MAX_DATA,
WT_MAX_STREAMS, WT_DATA_BLOCKED, and WT_STREAMS_BLOCKED.
An endpoint MUST NOT wait for a WT_DATA_BLOCKED or WT_STREAMS_BLOCKED
capsule before sending a WT_MAX_DATA or WT_MAX_STREAMS capsule; doing
so could result in the sender being blocked for at least an entire
round trip. Endpoints SHOULD send WT_MAX_DATA and WT_MAX_STREAMS
capsules as they consume data or close streams (similar to the
mechanism used in QUIC, see Section 4.2 of [RFC9000]).
5.6.1. Flow Control and Intermediaries
Because flow control in WebTransport is hop-by-hop and does not
provide an end-to-end signal, intermediaries MUST consume flow
control signals and express their own flow control limits to the next
hop. The intermediary can send these signals via HTTP/3 flow control
messages, HTTP/2 flow control messages, or as WebTransport flow
control capsules, where appropriate. Intermediaries are responsible
for storing any data for which they advertise flow control credit if
that data cannot be immediately forwarded to the next hop.
In practice, an intermediary that translates flow control signals
between similar WebTransport protocols, such as between two HTTP/3
connections, can often simply reexpress the same limits received on
one connection directly on the other connection.
An intermediary that does not want to be responsible for storing data
that cannot be immediately sent on its translated connection can
ensure that it does not advertise a higher flow control limit on one
connection than the corresponding limit on the translated connection.
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5.6.2. WT_MAX_STREAMS Capsule
An HTTP capsule [HTTP-DATAGRAM] called WT_MAX_STREAMS is introduced
to inform the peer of the cumulative number of streams of a given
type it is permitted to open. A WT_MAX_STREAMS capsule with a type
of 0x190B4D3F applies to bidirectional streams, and a WT_MAX_STREAMS
capsule with a type of 0x190B4D40 applies to unidirectional streams.
Note that, because Maximum Streams is a cumulative value representing
the total allowed number of streams, including previously closed
streams, endpoints repeatedly send new WT_MAX_STREAMS capsules with
increasing Maximum Streams values as streams are opened.
WT_MAX_STREAMS Capsule {
Type (i) = 0x190B4D3F..0x190B4D40,
Length (i),
Maximum Streams (i),
}
Figure 7: WT_MAX_STREAMS Capsule Format
WT_MAX_STREAMS capsules contain the following field:
Maximum Streams: A count of the cumulative number of streams of the
corresponding type that can be opened over the lifetime of the
session. This value cannot exceed 2^60, as it is not possible to
encode stream IDs larger than 2^62-1. Receipt of a capsule with a
Maximum Streams value larger than this limit MUST be treated as an
HTTP/3 error of type H3_DATAGRAM_ERROR.
An endpoint MUST NOT open more streams than permitted by the current
stream limit set by its peer. For instance, a server that receives a
unidirectional stream limit of 3 is permitted to open streams 3, 7,
and 11, but not stream 15.
Note that this limit includes streams that have been closed as well
as those that are open.
If an endpoint receives an incoming stream for a session that would
exceed the advertised Maximum Streams value, it MUST close the
WebTransport session with a WT_FLOW_CONTROL_ERROR error code.
Unlike in QUIC, where MAX_STREAMS frames can be delivered in any
order, WT_MAX_STREAMS capsules are sent on the WebTransport session's
connect stream and are delivered in order. If an endpoint receives a
WT_MAX_STREAMS capsule with a Maximum Streams value less than a
previously received value, it MUST close the WebTransport session
with a WT_FLOW_CONTROL_ERROR error code.
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The WT_MAX_STREAMS capsule defines special intermediary handling, as
described in Section 3.2 of [HTTP-DATAGRAM]. Intermediaries MUST
consume WT_MAX_STREAMS capsules for flow control purposes and MUST
generate and send appropriate flow control signals for their limits.
Initial values for these limits MAY be communicated by sending non-
zero values for SETTINGS_WT_INITIAL_MAX_STREAMS_UNI and
SETTINGS_WT_INITIAL_MAX_STREAMS_BIDI.
5.6.3. WT_STREAMS_BLOCKED Capsule
A sender SHOULD send a WT_STREAMS_BLOCKED capsule (type=0x190B4D43 or
0x190B4D44) when it wishes to open a stream but is unable to do so
due to the maximum stream limit set by its peer. A
WT_STREAMS_BLOCKED capsule of type 0x190B4D43 is used to indicate
reaching the bidirectional stream limit, and a WT_STREAMS_BLOCKED
capsule of type 0x190B4D44 is used to indicate reaching the
unidirectional stream limit.
A WT_STREAMS_BLOCKED capsule does not open the stream, but informs
the peer that a new stream was needed and the stream limit prevented
the creation of the stream. A sender is not required to send
WT_STREAMS_BLOCKED capsules, however WT_STREAMS_BLOCKED capsules can
be used as input to tuning of flow control algorithms and for
debugging purposes.
WT_STREAMS_BLOCKED Capsule {
Type (i) = 0x190B4D43..0x190B4D44,
Length (i),
Maximum Streams (i),
}
Figure 8: WT_STREAMS_BLOCKED Capsule Format
WT_STREAMS_BLOCKED capsules contain the following field:
Maximum Streams: A variable-length integer indicating the maximum
number of streams allowed at the time the capsule was sent. This
value cannot exceed 2^60, as it is not possible to encode stream
IDs larger than 2^62-1.
The WT_STREAMS_BLOCKED capsule defines special intermediary handling,
as described in Section 3.2 of [HTTP-DATAGRAM]. Intermediaries MUST
consume WT_STREAMS_BLOCKED capsules for flow control purposes and
MUST generate and send appropriate flow control signals for their
limits.
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5.6.4. WT_MAX_DATA Capsule
An HTTP capsule [HTTP-DATAGRAM] called WT_MAX_DATA (type=0x190B4D3D)
is introduced to inform the peer of the maximum amount of data that
can be sent on the WebTransport session as a whole.
This limit counts all data that is sent on streams of the
corresponding type, excluding the stream header (see Section 4.2 and
Section 4.3). For streams that were reset, implementing WT_MAX_DATA
requires that the QUIC stack provide the WebTransport implementation
with information about the final size of streams (see Section 4.5 of
[RFC9000]).
WT_MAX_DATA Capsule {
Type (i) = 0x190B4D3D,
Length (i),
Maximum Data (i),
}
Figure 9: WT_MAX_DATA Capsule Format
WT_MAX_DATA capsules contain the following field:
Maximum Data: A variable-length integer indicating the maximum
amount of data that can be sent on the entire session, in units of
bytes.
The sum of the lengths of Stream Body data sent on all streams
associated with this session MUST NOT exceed the Maximum Data value
advertised by a receiver. Note that capsules sent on the CONNECT
stream, and the Signal Value, Stream Type, and Session ID fields, are
not included in this limit. If an endpoint receives Stream Body data
in excess of this limit, it MUST close the WebTransport session with
a WT_FLOW_CONTROL_ERROR error code.
Unlike in QUIC, where MAX_DATA frames can be delivered in any order,
WT_MAX_DATA capsules are sent on the WebTransport session's connect
stream and are delivered in order. If an endpoint receives a
WT_MAX_DATA capsule with a Maximum Data value less than a previously
received value, it MUST close the WebTransport session with a
WT_FLOW_CONTROL_ERROR error code.
The WT_MAX_DATA capsule defines special intermediary handling, as
described in Section 3.2 of [HTTP-DATAGRAM]. Intermediaries MUST
consume WT_MAX_DATA capsules for flow control purposes and MUST
generate and send appropriate flow control signals for their limits
(see Section 5.6.1).
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The initial value for this limit MAY be communicated by sending a
non-zero value for SETTINGS_WT_INITIAL_MAX_DATA.
5.6.5. WT_DATA_BLOCKED Capsule
A sender SHOULD send a WT_DATA_BLOCKED capsule (type=0x190B4D41) when
it wishes to send data but is unable to do so due to WebTransport
session-level flow control. A sender is not required to send
WT_DATA_BLOCKED capsules, however WT_DATA_BLOCKED capsules can be
used as input to tuning of flow control algorithms and for debugging
purposes.
WT_DATA_BLOCKED Capsule {
Type (i) = 0x190B4D41,
Length (i),
Maximum Data (i),
}
Figure 10: WT_DATA_BLOCKED Capsule Format
WT_DATA_BLOCKED capsules contain the following field:
Maximum Data: A variable-length integer indicating the session-level
limit at which blocking occurred.
The WT_DATA_BLOCKED capsule defines special intermediary handling, as
described in Section 3.2 of [HTTP-DATAGRAM]. Intermediaries MUST
consume WT_DATA_BLOCKED capsules for flow control purposes and MUST
generate and send appropriate flow control signals for their limits
(see Section 5.6.1).
6. Session Termination
A WebTransport session over HTTP/3 is considered terminated when
either of the following conditions is met:
* the CONNECT stream is closed, either cleanly or abruptly, on
either side; or
* a WT_CLOSE_SESSION capsule is either sent or received.
Upon learning that the session has been terminated, the endpoint MUST
reset the send side and abort reading on the receive side of all
unidirectional and bidirectional streams associated with the session
(see Section 2.4 of [RFC9000]) using the WT_SESSION_GONE error code;
it MUST NOT send any new datagrams or open any new streams.
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To terminate a session with a detailed error message, an application
MAY provide such a message for the WebTransport endpoint to send in
an HTTP capsule [HTTP-DATAGRAM] of type WT_CLOSE_SESSION (0x2843).
The format of the capsule SHALL be as follows:
WT_CLOSE_SESSION Capsule {
Type (i) = WT_CLOSE_SESSION,
Length (i),
Application Error Code (32),
Application Error Message (..8192),
}
Figure 11: WT_CLOSE_SESSION Capsule Format
WT_CLOSE_SESSION has the following fields:
Application Error Code: A 32-bit error code provided by the
application closing the session.
Application Error Message: A UTF-8 encoded error message string
provided by the application closing the session. The message
takes up the remainder of the capsule, and its length MUST NOT
exceed 1024 bytes.
Note that the Application Error Code field does not mirror the Error
Code field in QUIC's CONNECTION_CLOSE frame (Section 19.19 of
[RFC9000]) because WebTransport application errors use a subset of
the HTTP/3 Error Code space and need to fit within those bounds, see
Section 4.4.
An endpoint that sends a WT_CLOSE_SESSION capsule MUST immediately
send a FIN on the CONNECT Stream. The endpoint MAY also send a
STOP_SENDING with error code WT_SESSION_GONE to indicate it is no
longer reading from the CONNECT stream. The recipient MUST either
close or reset the stream in response. If any additional stream data
is received on the CONNECT stream after receiving a WT_CLOSE_SESSION
capsule, the stream MUST be reset with code H3_MESSAGE_ERROR.
Cleanly terminating a CONNECT stream without a WT_CLOSE_SESSION
capsule SHALL be semantically equivalent to terminating it with a
WT_CLOSE_SESSION capsule that has an error code of 0 and an empty
error string.
In some scenarios, an endpoint might want to send a WT_CLOSE_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 WT_CLOSE_SESSION, thus never
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receiving the application error data contained in the latter. To
avoid this, the endpoint SHOULD wait until all CONNECT streams have
been closed by the peer before sending the CONNECTION_CLOSE; this
gives WT_CLOSE_SESSION properties similar to that of the QUIC
CONNECTION_CLOSE mechanism as a best-effort mechanism of delivering
application close metadata.
7. Considerations for Future Versions
Future versions of WebTransport that change the syntax of the CONNECT
requests used to establish WebTransport sessions will need to modify
the upgrade token used to identify WebTransport, allowing servers to
offer multiple versions simultaneously (see Section 9.1).
Servers that support future incompatible versions of WebTransport
signal that support by changing the codepoint used for the
SETTINGS_WT_ENABLED setting (see Section 9.2). Clients can select
the associated upgrade token, if applicable, to use when establishing
a new session, ensuring that servers will always know the syntax in
use for every incoming request.
Changes to future stream formats require changes to the
Unidirectional Stream type (see Section 4.2) and Bidirectional Stream
signal value (see Section 4.3) to allow recipients of incoming frames
to determine the WebTransport version, and corresponding wire format,
used for the session associated with that stream.
7.1. Negotiating the Draft Version
[[RFC editor: please remove this section before publication.]]
The wire format aspects of the protocol are negotiated by changing
the codepoint used for the SETTINGS_WT_ENABLED setting. Each draft
version defines a distinct codepoint for SETTINGS_WT_ENABLED. Both
the client and the server MUST send SETTINGS_WT_ENABLED with the
codepoint corresponding to their supported draft version. An
endpoint that supports multiple draft versions sends a
SETTINGS_WT_ENABLED value for each supported version, as each version
uses a different setting identifier. The highest version supported
by both endpoints is selected.
Because data streams can arrive at the server before the CONNECT
request that establishes the associated session, and the wire format
of the stream header depends on the negotiated version, the server
needs to know the client's version before processing any incoming
WebTransport streams. For this reason, the server MUST NOT process
any incoming WebTransport requests until the client's SETTINGS have
been received.
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8. Security Considerations
WebTransport over HTTP/3 satisfies all of the security requirements
imposed by [OVERVIEW] on WebTransport protocols, thus providing a
secure framework for client-server communication in cases when the
application is potentially untrusted.
WebTransport over HTTP/3 requires explicit opt-in through the use of
an HTTP/3 setting; this avoids potential protocol confusion attacks
by ensuring the HTTP/3 server explicitly supports it. It also
requires the use of the Origin header for browser traffic, providing
the server with the ability to deny access to Web-based applications
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 WebTransport sessions share both congestion
control and flow control context, a single application aggressively
using up those resources can cause other sessions to stall. A
WebTransport endpoint MUST implement flow control mechanisms if it
allows a WebTransport session to share the transport connection with
other WebTransport sessions. WebTransport endpoints SHOULD implement
a fairness scheme that ensures that each session that shares a
transport connection gets a reasonable share of controlled resources;
this applies both to sending data and to opening new streams.
An application could attempt to exhaust resources by opening too many
WebTransport sessions at once. In cases when the application is
untrusted, a WebTransport client SHOULD limit the number of outgoing
sessions it will open.
Note that the security considerations of HTTP/3 [HTTP3] apply to
WebTransport over HTTP/3. In particular, the denial-of-service
considerations in Section 10.5 of [HTTP3] are relevant. WebTransport
extends HTTP/3 with additional features that have legitimate uses but
can become a burden when they are used unnecessarily or to excess.
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WebTransport introduces new interaction modes that permit either
endpoint to send streams and datagrams to its peer. This is
particularly novel for clients, which previously had limited exposure
to unsolicited server-initiated traffic beyond server push (see
Section 4.6 of [HTTP3]). An endpoint that does not monitor use of
these features exposes itself to a risk of denial-of-service attack.
Implementations SHOULD track the use of WebTransport features, such
as the number of incoming streams and datagrams, and set limits on
their use. An endpoint MAY treat activity that is suspicious as a
connection error of type H3_EXCESSIVE_LOAD.
9. IANA Considerations
This document registers an upgrade token (Section 9.1), HTTP/3
settings (Section 9.2), an HTTP/3 stream type (Section 9.4), an
HTTP/3 error code (Section 9.5), and an HTTP header field
(Section 9.7).
9.1. Upgrade Token Registration
The following entry is added to the "Hypertext Transfer Protocol
(HTTP) Upgrade Token Registry" registry established by Section 16.7
of [HTTP].
The "webtransport-h3" label identifies HTTP/3 used as a protocol for
WebTransport:
Value: webtransport-h3
Description: WebTransport over HTTP/3
Reference: This document
9.2. HTTP/3 SETTINGS Parameter Registration
The following entry is added to the "HTTP/3 Settings" registry
established by [HTTP3]:
Setting Name: SETTINGS_WT_ENABLED
Value: 0x2c7cf000
Default: 0
Reference: This document
Change Controller: IETF
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Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Setting Name: SETTINGS_WT_INITIAL_MAX_STREAMS_UNI
Value: 0x2b64
Default: 0
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Setting Name: SETTINGS_WT_INITIAL_MAX_STREAMS_BIDI
Value: 0x2b65
Default: 0
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Setting Name: SETTINGS_WT_INITIAL_MAX_DATA
Value: 0x2b61
Default: 0
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
9.3. Frame Type Registration
The following entry is added to the "HTTP/3 Frame Type" registry
established by [HTTP3]:
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The WT_STREAM frame is reserved for the purpose of avoiding collision
with WebTransport HTTP/3 extensions:
Value: 0x41
Frame Type: WT_STREAM
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
9.4. Stream Type Registration
The following entry is added to the "HTTP/3 Stream Type" registry
established by [HTTP3]:
The "WebTransport stream" type allows unidirectional streams to be
used by WebTransport:
Value: 0x54
Stream Type: WebTransport stream
Reference: This document
Sender: Both
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
9.5. HTTP/3 Error Code Registration
The following entries are added to the "HTTP/3 Error Code" registry
established by [HTTP3]:
Name: WT_BUFFERED_STREAM_REJECTED
Value: 0x3994bd84
Description: WebTransport data stream rejected due to lack of
associated session.
Reference: This document.
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Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Name: WT_SESSION_GONE
Value: 0x170d7b68
Description: WebTransport data stream aborted because the associated
WebTransport session has been closed. Also used to indicate that
the endpoint is no longer reading from the CONNECT stream.
Specification: This document.
Name: WT_FLOW_CONTROL_ERROR
Value: 0x045d4487
Description: WebTransport session aborted because a flow control
error was encountered.
Reference: This document.
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Name: WT_ALPN_ERROR
Value: 0x0817b3dd
Description: WebTransport session aborted because application
protocol negotiation failed.
Reference: This document.
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Name: WT_REQUIREMENTS_NOT_MET
Value: 0x212c0d48
Description: HTTP/3 connection closed because the features required
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for WebTransport are not supported. Either the client or server
is missing required SETTINGS or transport parameters needed for
WebTransport.
Reference: This document.
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
In addition, the following range of entries is registered:
Name: WT_APPLICATION_ERROR
Value: 0x52e4a40fa8db to 0x52e5ac983162 inclusive, with the
exception of the codepoints of form 0x1f * N + 0x21.
Description: WebTransport application error codes.
Reference: This document.
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
9.6. Capsule Types
The following entries are added to the "HTTP Capsule Types" registry
established by [HTTP-DATAGRAM]:
The WT_CLOSE_SESSION capsule.
Value: 0x2843
Capsule Type: WT_CLOSE_SESSION
Status: permanent
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Notes: None
The WT_DRAIN_SESSION capsule.
Value: 0x78ae
Capsule Type: WT_DRAIN_SESSION
Status: provisional (when this document is approved this will become
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permanent)
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Notes: None
The WT_MAX_STREAMS capsule:
Value: 0x190B4D3F..0x190B4D40
Capsule Type: WT_MAX_STREAMS
Status: permanent
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Notes: None
The WT_STREAMS_BLOCKED capsule:
Value: 0x190B4D43..0x190B4D44
Capsule Type: WT_STREAMS_BLOCKED
Status: permanent
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Notes: None
The WT_MAX_DATA capsule:
Value: 0x190B4D3D
Capsule Type: WT_MAX_DATA
Status: permanent
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
(mailto:webtransport@ietf.org)
Notes: None
The WT_DATA_BLOCKED capsule:
Value: 0x190B4D41
Capsule Type: WT_DATA_BLOCKED
Status: permanent
Reference: This document
Change Controller: IETF
Contact: WebTransport Working Group webtransport@ietf.org
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(mailto:webtransport@ietf.org)
Notes: None
9.7. Protocol Negotiation HTTP Header Fields
The following HTTP header fields are used for negotiating a protocol
(Section 3.3). These are added to the "HTTP Field Name" registry
established in Section 18.4 of [HTTP]:
The WT-Available-Protocols field:
Field Name: WT-Available-Protocols
Status: permanent
Structured Type: List
Reference: Section 3.3
Comments: None
The WT-Protocol field:
Field Name: WT-Protocol
Status: permanent
Structured Type: Item
Reference: Section 3.3
Comments: None
10. References
10.1. Normative References
[FIELDS] Nottingham, M. and P. Kamp, "Structured Field Values for
HTTP", RFC 9651, DOI 10.17487/RFC9651, September 2024,
<https://www.rfc-editor.org/rfc/rfc9651>.
[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>.
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[OVERVIEW] Kinnear, E. and V. Vasiliev, "The WebTransport Protocol
Framework", Work in Progress, Internet-Draft, draft-ietf-
webtrans-overview-11, 20 October 2025,
<https://datatracker.ietf.org/doc/html/draft-ietf-
webtrans-overview-11>.
[QUIC-DATAGRAM]
Pauly, T., Kinnear, E., and D. Schinazi, "An Unreliable
Datagram Extension to QUIC", RFC 9221,
DOI 10.17487/RFC9221, March 2022,
<https://www.rfc-editor.org/rfc/rfc9221>.
[RESET-STREAM-AT]
Seemann, M. and K. Oku, "QUIC Stream Resets with Partial
Delivery", Work in Progress, Internet-Draft, draft-ietf-
quic-reliable-stream-reset-07, 14 June 2025,
<https://datatracker.ietf.org/doc/html/draft-ietf-quic-
reliable-stream-reset-07>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/rfc/rfc2119>.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, DOI 10.17487/RFC3986, January 2005,
<https://www.rfc-editor.org/rfc/rfc3986>.
[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
DOI 10.17487/RFC6454, December 2011,
<https://www.rfc-editor.org/rfc/rfc6454>.
[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>.
[RFC8446] Rescorla, E., "The Transport Layer Security (TLS) Protocol
Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
<https://www.rfc-editor.org/rfc/rfc8446>.
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[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>.
[RFC9114] Bishop, M., Ed., "HTTP/3", RFC 9114, DOI 10.17487/RFC9114,
June 2022, <https://www.rfc-editor.org/rfc/rfc9114>.
[RFC9218] Oku, K. and L. Pardue, "Extensible Prioritization Scheme
for HTTP", RFC 9218, DOI 10.17487/RFC9218, June 2022,
<https://www.rfc-editor.org/rfc/rfc9218>.
[RFC9220] Hamilton, R., "Bootstrapping WebSockets with HTTP/3",
RFC 9220, DOI 10.17487/RFC9220, June 2022,
<https://www.rfc-editor.org/rfc/rfc9220>.
10.2. Informative References
[I-D.ietf-httpapi-ratelimit-headers]
Polli, R., Ruiz, A. M., and D. Miller, "RateLimit header
fields for HTTP", Work in Progress, Internet-Draft, draft-
ietf-httpapi-ratelimit-headers-10, 27 September 2025,
<https://datatracker.ietf.org/doc/html/draft-ietf-httpapi-
ratelimit-headers-10>.
[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-http2-13,
20 October 2025, <https://datatracker.ietf.org/doc/html/
draft-ietf-webtrans-http2-13>.
[ORIGIN] Fette, I. and A. Melnikov, "The WebSocket Protocol",
RFC 6455, DOI 10.17487/RFC6455, December 2011,
<https://www.rfc-editor.org/rfc/rfc6455>.
[RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan,
"Transport Layer Security (TLS) Application-Layer Protocol
Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301,
July 2014, <https://www.rfc-editor.org/rfc/rfc7301>.
[RFC9208] Melnikov, A., "IMAP QUOTA Extension", RFC 9208,
DOI 10.17487/RFC9208, March 2022,
<https://www.rfc-editor.org/rfc/rfc9208>.
[RFC9308] Kühlewind, M. and B. Trammell, "Applicability of the QUIC
Transport Protocol", RFC 9308, DOI 10.17487/RFC9308,
September 2022, <https://www.rfc-editor.org/rfc/rfc9308>.
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[WEBTRANS-H2]
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-http2-13,
20 October 2025, <https://datatracker.ietf.org/doc/html/
draft-ietf-webtrans-http2-13>.
Index
W
W
WT_DATA_BLOCKED Section 5.4, Paragraph 3; Section 5.6,
Paragraph 1; Section 5.6, Paragraph 2; Section 5.6.5,
Paragraph 1; Section 5.6.5, Paragraph 3; Section 5.6.5,
Paragraph 5; Section 9.6, Paragraph 13.4.1
WT_MAX_DATA Section 5.4, Paragraph 1; Section 5.4, Paragraph
2; Section 5.4, Paragraph 3; Section 5.5, Paragraph 2.2.1;
Section 5.5.3, Paragraph 1; Section 5.6, Paragraph 1;
Section 5.6, Paragraph 2; Section 5.6.4, Paragraph 1;
Section 5.6.4, Paragraph 2; Section 5.6.4, Paragraph 4;
Section 5.6.4, Paragraph 7; Section 5.6.4, Paragraph 8;
Section 9.6, Paragraph 11.4.1
WT_MAX_STREAMS Section 5.3, Paragraph 1; Section 5.5,
Paragraph 2.1.1; Section 5.5.1, Paragraph 1; Section 5.5.2,
Paragraph 1; Section 5.6, Paragraph 1; Section 5.6,
Paragraph 2; Section 5.6.2, Paragraph 1; Section 5.6.2,
Paragraph 2; Section 5.6.2, Paragraph 4; Section 5.6.2,
Paragraph 9; Section 5.6.2, Paragraph 10; Section 9.6,
Paragraph 7.4.1
WT_STREAMS_BLOCKED Section 5.3, Paragraph 5; Section 5.6,
Paragraph 1; Section 5.6, Paragraph 2; Section 5.6.3,
Paragraph 1; Section 5.6.3, Paragraph 2; Section 5.6.3,
Paragraph 4; Section 5.6.3, Paragraph 6; Section 9.6,
Paragraph 9.4.1
Authors' Addresses
Alan Frindell
Facebook
Email: afrind@fb.com
Eric Kinnear
Apple Inc.
Email: ekinnear@apple.com
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Victor Vasiliev
Google
Email: vasilvv@google.com
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