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Transport WebTransport 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 and provides support for unidirectional streams, bidirectional streams and datagrams, all multiplexed within the same HTTP/3 connection. Note to Readers Discussion of this draft takes place on the WebTransport mailing list (webtransport@ietf.org), which is archived at <https://mailarchive.ietf.org/arch/search/?email_list=webtransport>. The repository tracking the issues for this draft can be found at <https://github.com/ietf-wg-webtrans/draft-ietf-webtrans-http3/issues>. The web API draft corresponding to this document can be found at <https://w3c.github.io/webtransport/>.

Introduction HTTP/3 is a protocol defined on top of QUIC 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. Using the mechanism described here, multiple WebTransport instances can be multiplexed simultaneously with regular HTTP traffic on the same HTTP/3 connection.

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 when, and only when, they appear in all capitals, as shown here. This document follows terminology defined in Section 1.2 of . 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.

Protocol Overview WebTransport servers in general are identified by a pair of authority value and path value (defined in 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 . 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 .

A WebTransport session is terminated when the CONNECT stream that created it is closed.

Session Establishment

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 ). 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 ). 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 ). 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 ).

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

Creating a New Session As WebTransport sessions are established over HTTP/3, they are identified using the https URI scheme (, 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 () 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 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 (). 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 (). If all checks pass, the WebTransport server MAY accept the session by replying with a 2xx series status code, as defined in . 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 (). 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 ). However, WebTransport-related SETTINGS parameters may be retained from the previous session as described in Section 7.2.4.2 of . 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 . The Capsule Protocol is negotiated when the server sends a 2xx response. The capsule-protocol header field is not required by WebTransport and can safely be ignored by WebTransport endpoints.

Subprotocol Negotiation WebTransport over HTTP/3 offers a subprotocol negotiation mechanism, similar to TLS Application-Layer Protocol Negotiation Extension (ALPN) ; 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 . 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.

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 . 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 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 . Unlike the previous method, this signal is directly propagated to the application.

Prioritization WebTransport sessions are initiated using extended CONNECT. While 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 (); clients MAY reprioritize by sending PRIORITY_UPDATE frames (). In extension to , it is RECOMMENDED that clients and servers apply the scheduling guidance in both and 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 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.

WebTransport Features WebTransport over HTTP/3 provides the following features described in : 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 . 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.

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 ().

Unidirectional WebTransport stream format

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 and of ). WebTransport extends HTTP/3 to allow clients to declare and use alternative request stream rules. Once a client receives settings indicating WebTransport support (), 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 (), 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 ().

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.

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 ; 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 .

Pseudocode for converting between WebTransport application errors and HTTP/3 error codes

WebTransport data streams are associated with sessions through a header at the beginning of the stream; resetting a stream may result in that data being discarded. Because of that, WebTransport application error codes are best effort, as the WebTransport stack is not always capable of associating the reset code with a session. The only exception is the situation where there is only one session on a given HTTP/3 connection, and no intermediaries between the client and the server. WebTransport implementations SHALL forward the error code for a stream associated with a known session to the application that owns that session; similarly, the intermediaries SHALL reset the streams with corresponding error code when receiving a reset from the peer. If a WebTransport implementation intentionally allows only one session over a given HTTP/3 connection, it SHALL forward the error codes within WebTransport application error code range to the application that owns the only session on that connection.

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 ). 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.

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 ). 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.

Interaction with HTTP/3 GOAWAY frame HTTP/3 defines a graceful shutdown mechanism () 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. 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.

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

• the CONNECT stream is closed, either cleanly or abruptly, on either side; or
• a CLOSE_WEBTRANSPORT_SESSION capsule is either sent or received.

Upon learning that the session has been terminated, the endpoint MUST reset the send side and abort reading on the receive side of all of the streams associated with the session (see Section 2.4 of ) 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 of type CLOSE_WEBTRANSPORT_SESSION (0x2843). The format of the capsule SHALL be as follows: CLOSE_WEBTRANSPORT_SESSION has the following fields:

Application Error Code:

A 32-bit error code provided by the application closing the connection.

Application Error Message:

A UTF-8 encoded error message string provided by the application closing the connection. The message takes up the remainder of the capsule, and its length MUST NOT exceed 1024 bytes.

An endpoint that sends a CLOSE_WEBTRANSPORT_SESSION capsule MUST immediately send a FIN. The endpoint MAY send a STOP_SENDING to indicate it is no longer reading from the CONNECT stream. The recipient MUST close the stream upon receiving a FIN. If any additional stream data is received on the CONNECT stream after receiving a CLOSE_WEBTRANSPORT_SESSION capsule, the stream MUST be reset with code H3_MESSAGE_ERROR. Cleanly terminating a CONNECT stream without a CLOSE_WEBTRANSPORT_SESSION capsule SHALL be semantically equivalent to terminating it with a CLOSE_WEBTRANSPORT_SESSION capsule that has an error code of 0 and an empty error string. In some scenarios, an endpoint might want to send a CLOSE_WEBTRANSPORT_SESSION with detailed close information and then immediately close the underlying QUIC connection. If the endpoint were to do both of those simultaneously, the peer could potentially receive the CONNECTION_CLOSE before receiving the CLOSE_WEBTRANSPORT_SESSION, thus never receiving the application error data contained in the latter. To avoid this, the endpoint SHOULD wait until all of the data on the CONNECT stream is acknowledged before sending the CONNECTION_CLOSE; this gives CLOSE_WEBTRANSPORT_SESSION properties similar to that of the QUIC CONNECTION_CLOSE mechanism as a best-effort mechanism of delivering application close metadata.

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 ). Servers that support future incompatible versions of WebTransport signal that support by changing the codepoint used for the SETTINGS_WEBTRANSPORT_MAX_SESSIONS parameter (see ). 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 ) and Bidirectional Stream signal value (see ) to allow recipients of incoming frames to determine the WebTransport version, and corresponding wire format, used for the session associated with that stream.

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.

Security Considerations WebTransport over HTTP/3 satisfies all of the security requirements imposed by 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.

IANA Considerations

Upgrade Token Registration The following entry is added to the "Hypertext Transfer Protocol (HTTP) Upgrade Token Registry" registry established by Section 16.7 of . The "webtransport" label identifies HTTP/3 used as a protocol for WebTransport:

Value:

webtransport

Description:

WebTransport over HTTP/3

Reference:

This document and

HTTP/3 SETTINGS Parameter Registration The following entry is added to the "HTTP/3 Settings" registry established by : The SETTINGS_WEBTRANSPORT_MAX_SESSIONS parameter indicates that the specified HTTP/3 endpoint is WebTransport-capable and the number of concurrent sessions it is willing to receive. The default value for the SETTINGS_WEBTRANSPORT_MAX_SESSIONS parameter is "0", meaning that the endpoint is not willing to receive any WebTransport sessions.

Setting Name:

WEBTRANSPORT_MAX_SESSIONS

Value:

0xc671706a

Default:

0

Specification:

This document

Frame Type Registration The following entry is added to the "HTTP/3 Frame Type" registry established by : The WEBTRANSPORT_STREAM frame is reserved for the purpose of avoiding collision with WebTransport HTTP/3 extensions:

Code:

0x41

Frame Type:

WEBTRANSPORT_STREAM

Specification:

This document

Stream Type Registration The following entry is added to the "HTTP/3 Stream Type" registry established by : The "WebTransport stream" type allows unidirectional streams to be used by WebTransport:

Code:

0x54

Stream Type:

WebTransport stream

Specification:

This document

Sender:

Both

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

Name:

WEBTRANSPORT_BUFFERED_STREAM_REJECTED

Value:

0x3994bd84

Description:

WebTransport data stream rejected due to lack of associated session.

Specification:

This document.

Name:

WEBTRANSPORT_SESSION_GONE

Value:

0x170d7b68

Description:

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

Specification:

This document.

In addition, the following range of entries is registered:

Name:

WEBTRANSPORT_APPLICATION_ERROR

Value:

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

Description:

WebTransport application error codes.

Specification:

This document.

Capsule Types The following entries are added to the "HTTP Capsule Types" registry established by : The CLOSE_WEBTRANSPORT_SESSION capsule.

Value:

0x2843

Capsule Type:

CLOSE_WEBTRANSPORT_SESSION

Status:

permanent

Specification:

This document

Change Controller:

IETF

Contact:

WebTransport Working Group webtransport@ietf.org

Notes:

None

The DRAIN_WEBTRANSPORT_SESSION capsule.

Value:

0x78ae

Capsule Type:

DRAIN_WEBTRANSPORT_SESSION

Status:

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

Specification:

This document

Change Controller:

IETF

Contact:

WebTransport Working Group webtransport@ietf.org

Notes:

None

References Normative References Google The WebTransport Protocol Framework enables clients constrained by the Web security model to communicate with a remote server using a secure multiplexed transport. It consists of a set of individual protocols that are safe to expose to untrusted applications, combined with an abstract model that allows them to be used interchangeably. This document defines the overall requirements on the protocols used in WebTransport, as well as the common features of the protocols, support for some of which may be optional. The QUIC transport protocol has several features that are desirable in a transport for HTTP, such as stream multiplexing, per-stream flow control, and low-latency connection establishment. This document describes a mapping of HTTP semantics over QUIC. This document also identifies HTTP/2 features that are subsumed by QUIC and describes how HTTP/2 extensions can be ported to HTTP/3. This document defines the core of the QUIC transport protocol. QUIC provides applications with flow-controlled streams for structured communication, low-latency connection establishment, and network path migration. QUIC includes security measures that ensure confidentiality, integrity, and availability in a range of deployment circumstances. Accompanying documents describe the integration of TLS for key negotiation, loss detection, and an exemplary congestion control algorithm. In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings. A Uniform Resource Identifier (URI) is a compact sequence of characters that identifies an abstract or physical resource. This specification defines the generic URI syntax and a process for resolving URI references that might be in relative form, along with guidelines and security considerations for the use of URIs on the Internet. The URI syntax defines a grammar that is a superset of all valid URIs, allowing an implementation to parse the common components of a URI reference without knowing the scheme-specific requirements of every possible identifier. This specification does not define a generative grammar for URIs; that task is performed by the individual specifications of each URI scheme. [STANDARDS-TRACK] This document defines a mechanism for running the WebSocket Protocol (RFC 6455) over a single stream of an HTTP/2 connection. This document describes HTTP Datagrams, a convention for conveying multiplexed, potentially unreliable datagrams inside an HTTP connection. In HTTP/3, HTTP Datagrams can be sent unreliably using the QUIC DATAGRAM extension. When the QUIC DATAGRAM frame is unavailable or undesirable, HTTP Datagrams can be sent using the Capsule Protocol, which is a more general convention for conveying data in HTTP connections. HTTP Datagrams and the Capsule Protocol are intended for use by HTTP extensions, not applications. This document defines an extension to the QUIC transport protocol to add support for sending and receiving unreliable datagrams over a QUIC connection. The mechanism for running the WebSocket Protocol over a single stream of an HTTP/2 connection is equally applicable to HTTP/3, but the HTTP-version-specific details need to be specified. This document describes how the mechanism is adapted for HTTP/3. This document defines the concept of an "origin", which is often used as the scope of authority or privilege by user agents. Typically, user agents isolate content retrieved from different origins to prevent malicious web site operators from interfering with the operation of benign web sites. In addition to outlining the principles that underlie the concept of origin, this document details how to determine the origin of a URI and how to serialize an origin into a string. It also defines an HTTP header field, named "Origin", that indicates which origins are associated with an HTTP request. [STANDARDS-TRACK] The Hypertext Transfer Protocol (HTTP) is a stateless application-level protocol for distributed, collaborative, hypertext information systems. This document describes the overall architecture of HTTP, establishes common terminology, and defines aspects of the protocol that are shared by all versions. In this definition are core protocol elements, extensibility mechanisms, and the "http" and "https" Uniform Resource Identifier (URI) schemes. This document updates RFC 3864 and obsoletes RFCs 2818, 7231, 7232, 7233, 7235, 7538, 7615, 7694, and portions of 7230. This document describes a set of data types and associated algorithms that are intended to make it easier and safer to define and handle HTTP header and trailer fields, known as "Structured Fields", "Structured Headers", or "Structured Trailers". It is intended for use by specifications of new HTTP fields that wish to use a common syntax that is more restrictive than traditional HTTP field values. This document specifies additional HyperText Transfer Protocol (HTTP) status codes for a variety of common situations. [STANDARDS-TRACK] This document describes a scheme that allows an HTTP client to communicate its preferences for how the upstream server prioritizes responses to its requests, and also allows a server to hint to a downstream intermediary how its responses should be prioritized when they are forwarded. This document defines the Priority header field for communicating the initial priority in an HTTP version-independent manner, as well as HTTP/2 and HTTP/3 frames for reprioritizing responses. These share a common format structure that is designed to provide future extensibility. Informative References This document describes a Transport Layer Security (TLS) extension for application-layer protocol negotiation within the TLS handshake. For instances in which multiple application protocols are supported on the same TCP or UDP port, this extension allows the application layer to negotiate which protocol will be used within the TLS connection. Facebook Inc. Apple Inc. Apple Inc. Mozilla Google Facebook Inc. WebTransport defines a set of low-level communications features designed for client-server interactions that are initiated by Web clients. This document describes a protocol that can provide many of the capabilities of WebTransport over HTTP/2. This protocol enables the use of WebTransport when a UDP-based protocol is not available.

Changelog

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

• draft-07 requires SETTINGS_WEBTRANSPORT_MAX_SESSIONS (#86) and uses it for version negotiation (#129)
• draft-07 explicitly requires SETTINGS_ENABLE_CONNECT_PROTOCOL to be enabled (#93)
• draft-07 explicitly requires SETTINGS_H3_DATAGRAM to be enabled (#106)
• draft-07 only allows WEBTRANSPORT_STREAM at the beginning of the stream

The following changes that are present in draft-07 can be also implemented by a draft-02 implementation safely:

• Expanding stream reset error code space from 8 to 32 bits (#115)
• WEBTRANSPORT_SESSION_GONE error code (#75)
• Handling for HTTP GOAWAY (#76)
• DRAIN_WEBTRANSPORT_SESSION capsule (#79)
• Disallowing following redirects automatically (#113)