Internet-Draft WebTransport-H3 July 2024
Frindell, et al. Expires 10 January 2025 [Page]
Network Working Group
Intended Status:
Standards Track
A. Frindell
E. Kinnear
Apple Inc.
V. Vasiliev

WebTransport over HTTP/3


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 (, which is archived at <>.

The repository tracking the issues for this draft can be found at <>. The web API draft corresponding to this document can be found at <>.

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

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 10 January 2025.

Table of Contents

1. Introduction

HTTP/3 [HTTP3] is a protocol defined on top of QUIC [RFC9000] that can multiplex HTTP requests over a QUIC connection. This document defines a mechanism for multiplexing non-HTTP data with HTTP/3 in a manner that conforms with the WebTransport protocol requirements and semantics[OVERVIEW]. Using the mechanism described here, multiple WebTransport instances can be multiplexed simultaneously with regular HTTP traffic on the same HTTP/3 connection.

1.1. Terminology

The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.

This document follows terminology defined in Section 1.2 of [OVERVIEW]. Note that this document distinguishes between a WebTransport server and an HTTP/3 server. An HTTP/3 server is the server that terminates HTTP/3 connections; a WebTransport server is an application that accepts WebTransport sessions, which can be accessed via an HTTP/3 server.

2. Overview

2.1. QUIC, WebTransport, and HTTP/3

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

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

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

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

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

WebTransport HTTP Semantics and Session Setup Nearly direct HTTP/3 QUIC
Figure 1: WebTransport Layering

2.1.1. Minimizing Implementation Complexity

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

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

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

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

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

2.2. Protocol Overview

WebTransport servers in general are identified by a pair of authority value and path value (defined in [RFC3986] Sections 3.2 and 3.3 correspondingly).

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

WebTransport sessions are initiated inside a given HTTP/3 connection by the client, who sends an extended CONNECT request [RFC8441]. If the server accepts the request, a WebTransport session is established. The resulting stream will be further referred to as a CONNECT stream, and its stream ID is used to uniquely identify a given WebTransport session within the connection. The ID of the CONNECT stream that established a given WebTransport session will be further referred to as a Session ID.

After the session is established, the peers can exchange data using the following mechanisms:

  • 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 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]).

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

3.2. Extended CONNECT in HTTP/3

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

3.3. Creating a New Session

As WebTransport sessions are established over HTTP/3, they are identified using the https URI scheme ([HTTP], Section 4.2.2).

In order to create a new WebTransport session, a client can send an HTTP CONNECT request. The :protocol pseudo-header field ([RFC8441]) MUST be set to webtransport. The :scheme field MUST be https. Both the :authority and the :path value MUST be set; those fields indicate the desired WebTransport server. If the WebTransport session is coming from a browser client, an Origin header [RFC6454] MUST be provided within the request; otherwise, the header is OPTIONAL.

Upon receiving an extended CONNECT request with a :protocol field set to webtransport, the HTTP/3 server can check if it has a WebTransport server associated with the specified :authority and :path values. If it does not, it SHOULD reply with status code 404 (Section 15.5.5 of [HTTP]). When the request contains the Origin header, the WebTransport server MUST verify the Origin header to ensure that the specified origin is allowed to access the server in question. If the verification fails, the WebTransport server SHOULD reply with status code 403 (Section 15.5.4 of [HTTP]). If all checks pass, the WebTransport server MAY accept the session by replying with a 2xx series status code, as defined in Section 15.3 of [HTTP].

From the client's perspective, a WebTransport session is established when the client receives a 2xx response. From the server's perspective, a session is established once it sends a 2xx response.

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

Clients cannot initiate WebTransport in 0-RTT packets, as the CONNECT method is not considered safe (see Section 10.9 of [HTTP3]). However, WebTransport-related SETTINGS parameters may be retained from the previous session as described in Section 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.

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.

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.

4.1. Unidirectional streams

WebTransport endpoints can initiate unidirectional streams. The HTTP/3 unidirectional stream type SHALL be 0x54. The body of the stream SHALL be the stream type, followed by the session ID, encoded as a variable-length integer, followed by the user-specified stream data (Figure 2).

Unidirectional Stream {
    Stream Type (i) = 0x54,
    Session ID (i),
    Stream Body (..)
Figure 2: Unidirectional WebTransport stream format

4.2. Bidirectional Streams

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

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

WebTransport extends HTTP/3 by defining rules for all server-initiated bidirectional streams. Once a server receives an incoming CONNECT request establishing a WebTransport session (Section 3.1), it can open a bidirectional stream for use with that session and SHALL send a special signal value, encoded as a variable-length integer, as the first bytes of the stream in order to indicate how the remaining bytes on the stream are used.

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

Bidirectional Stream {
    Signal Value (i) = 0x41,
    Session ID (i),
    Stream Body (..)
Figure 3: Bidirectional WebTransport stream format

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

4.3. Resetting Data Streams

A WebTransport endpoint may send a RESET_STREAM or a STOP_SENDING frame for a WebTransport data stream. Those signals are propagated by the WebTransport implementation to the application.

A WebTransport application SHALL provide an error code for those operations. Since WebTransport shares the error code space with HTTP/3, WebTransport application errors for streams are limited to an unsigned 32-bit integer, assuming values between 0x00000000 and 0xffffffff. WebTransport implementations SHALL remap those error codes into the error range reserved for WEBTRANSPORT_APPLICATION_ERROR, where 0x00000000 corresponds to 0x52e4a40fa8db, and 0xffffffff corresponds to 0x52e5ac983162. Note that there are code points inside that range of form "0x1f * N + 0x21" that are reserved by Section 8.1 of [HTTP3]; those have to be skipped when mapping the error codes (i.e. the two HTTP/3 error codepoints adjacent to a reserved codepoint would map to two adjacent WebTransport application error codepoints). An example pseudocode can be seen in Figure 4.

    first = 0x52e4a40fa8db
    last = 0x52e5ac983162

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

    def http_code_to_webtransport_code(h):
        assert(first <= h <= last)
        assert((h - 0x21) % 0x1f != 0)
        shifted = h - first
        return shifted - floor(shifted / 0x1f)
Figure 4: Pseudocode for converting between WebTransport application errors and HTTP/3 error codes

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

WebTransport implementations SHALL forward the error code for a stream associated with a known session to the application that owns that session; similarly, the intermediaries SHALL reset the streams with corresponding error code when receiving a reset from the peer. If a WebTransport implementation intentionally allows only one session over a given HTTP/3 connection, it SHALL forward the error codes within WebTransport application error code range to the application that owns the only session on that connection.

4.4. Datagrams

Datagrams can be sent using HTTP Datagrams. The WebTransport datagram payload is sent unmodified in the "HTTP Datagram Payload" field of an HTTP Datagram (Section 2.1 of [HTTP-DATAGRAM]). Note that the payload field directly follows the Quarter Stream ID field, which is at the start of the QUIC DATAGRAM frame payload and refers to the CONNECT stream that established the WebTransport session.

4.5. Buffering Incoming Streams and Datagrams

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

Clients can, however, send a SETTINGS frame, multiple WebTransport CONNECT requests, WebTransport data streams, and WebTransport datagrams all within a single flight. As those can arrive out of order, a WebTransport server could be put into a situation where it receives a stream or a datagram without a corresponding session. Similarly, a client may receive a server-initiated stream or a datagram before receiving the CONNECT response headers from the server.

To handle this case, WebTransport endpoints SHOULD buffer streams and datagrams until those can be associated with an established session. To avoid resource exhaustion, the endpoints MUST limit the number of buffered streams and datagrams. When the number of buffered streams is exceeded, a stream SHALL be closed by sending a RESET_STREAM and/or STOP_SENDING with the WEBTRANSPORT_BUFFERED_STREAM_REJECTED error code. When the number of buffered datagrams is exceeded, a datagram SHALL be dropped. It is up to an implementation to choose what stream or datagram to discard.

4.6. Interaction with HTTP/3 GOAWAY frame

HTTP/3 defines a graceful shutdown mechanism (Section 5.2 of [HTTP3]) that allows a peer to send a GOAWAY frame indicating that it will no longer accept any new incoming requests or pushes.

A client receiving GOAWAY cannot initiate CONNECT requests for new WebTransport sessions 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.

  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.

4.7. Use of Keying Material Exporters

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

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

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

5. Session Termination

A WebTransport session over HTTP/3 is considered terminated when either of the following conditions is met:

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:

  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 either close or reset the stream in response. If any additional stream data is received on the CONNECT stream after receiving a CLOSE_WEBTRANSPORT_SESSION capsule, the stream MUST be reset with code H3_MESSAGE_ERROR.

Cleanly terminating a CONNECT stream without a CLOSE_WEBTRANSPORT_SESSION capsule SHALL be semantically equivalent to terminating it with a CLOSE_WEBTRANSPORT_SESSION capsule that has an error code of 0 and an empty error string.

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

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.

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:




WebTransport over HTTP/3


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

8.2. HTTP/3 SETTINGS Parameter Registration

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

The SETTINGS_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:







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:



Frame Type:



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:



Stream Type:

WebTransport stream


This document



8.5. HTTP/3 Error Code Registration

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






WebTransport data stream rejected due to lack of associated session.


This document.






WebTransport data stream aborted because the associated WebTransport session has been closed.


This document.

In addition, the following range of entries is registered:




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


WebTransport application error codes.


This document.

8.6. Capsule Types

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




Capsule Type:





This document

Change Controller:



WebTransport Working Group






Capsule Type:



provisional (when this document is approved this will become permanent)


This document

Change Controller:



WebTransport Working Group



9. References

9.1. Normative References

Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke, Ed., "HTTP Semantics", STD 97, RFC 9110, DOI 10.17487/RFC9110, , <>.
Schinazi, D. and L. Pardue, "HTTP Datagrams and the Capsule Protocol", RFC 9297, DOI 10.17487/RFC9297, , <>.
Bishop, M., Ed., "HTTP/3", RFC 9114, DOI 10.17487/RFC9114, , <>.
Vasiliev, V., "The WebTransport Protocol Framework", Work in Progress, Internet-Draft, draft-ietf-webtrans-overview-07, , <>.
Pauly, T., Kinnear, E., and D. Schinazi, "An Unreliable Datagram Extension to QUIC", RFC 9221, DOI 10.17487/RFC9221, , <>.
Seemann, M. and K. Oku, "QUIC Stream Resets with Partial Delivery", Work in Progress, Internet-Draft, draft-ietf-quic-reliable-stream-reset-06, , <>.
Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, , <>.
Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform Resource Identifier (URI): Generic Syntax", STD 66, RFC 3986, DOI 10.17487/RFC3986, , <>.
Barth, A., "The Web Origin Concept", RFC 6454, DOI 10.17487/RFC6454, , <>.
Nottingham, M. and R. Fielding, "Additional HTTP Status Codes", RFC 6585, DOI 10.17487/RFC6585, , <>.
Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, , <>.
McManus, P., "Bootstrapping WebSockets with HTTP/2", RFC 8441, DOI 10.17487/RFC8441, , <>.
Rescorla, E., "The Transport Layer Security (TLS) Protocol Version 1.3", RFC 8446, DOI 10.17487/RFC8446, , <>.
Nottingham, M. and P. Kamp, "Structured Field Values for HTTP", RFC 8941, DOI 10.17487/RFC8941, , <>.
Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based Multiplexed and Secure Transport", RFC 9000, DOI 10.17487/RFC9000, , <>.
Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke, Ed., "HTTP Semantics", STD 97, RFC 9110, DOI 10.17487/RFC9110, , <>.
Bishop, M., Ed., "HTTP/3", RFC 9114, DOI 10.17487/RFC9114, , <>.
Oku, K. and L. Pardue, "Extensible Prioritization Scheme for HTTP", RFC 9218, DOI 10.17487/RFC9218, , <>.
Hamilton, R., "Bootstrapping WebSockets with HTTP/3", RFC 9220, DOI 10.17487/RFC9220, , <>.

9.2. Informative References

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, , <>.
Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC 6455, DOI 10.17487/RFC6455, , <>.
Friedl, S., Popov, A., Langley, A., and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, , <>.
Melnikov, A., "IMAP QUOTA Extension", RFC 9208, DOI 10.17487/RFC9208, , <>.
Kühlewind, M. and B. Trammell, "Applicability of the QUIC Transport Protocol", RFC 9308, DOI 10.17487/RFC9308, , <>.

Appendix A. Changelog

A.1. Changes between draft versions 02 and 07

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

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

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

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

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

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)


  • Disallowing following redirects automatically (#113)

Authors' Addresses

Alan Frindell
Eric Kinnear
Apple Inc.
Victor Vasiliev