WebTransport

Editor’s Draft,

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Editors:
Bernard Aboba (Microsoft Corporation)
Nidhi Jaju (Google)
Victor Vasiliev (Google)
Former Editors:
Peter Thatcher (Google)
Robin Raymond (Optical Tone Ltd.)
Yutaka Hirano (Google)

Abstract

This document defines a set of ECMAScript APIs in WebIDL to allow data to be sent and received between a browser and server, utilizing [WEB-TRANSPORT-HTTP3] and [WEB-TRANSPORT-HTTP2]. This specification is being developed in conjunction with protocol specifications developed by the IETF WEBTRANS Working Group.

Status of this document

This is a public copy of the editors’ draft. It is provided for discussion only and may change at any moment. Its publication here does not imply endorsement of its contents by W3C. Don’t cite this document other than as work in progress.

Feedback and comments on this document are welcome. Please file an issue in this document’s GitHub repository.

This document was produced by the WebTransport Working Group.

This document was produced by a group operating under the W3C Patent Policy. W3C maintains a public list of any patent disclosures made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes contains Essential Claim(s) must disclose the information in accordance with section 6 of the W3C Patent Policy.

This document is governed by the 03 November 2023 W3C Process Document.

1. Introduction

This section is non-normative.

This specification uses [WEB-TRANSPORT-HTTP3] and [WEB-TRANSPORT-HTTP2] to send data to and receive data from servers. It can be used like WebSockets but with support for multiple streams, unidirectional streams, out-of-order delivery, and reliable as well as unreliable transport.

Note: The API presented in this specification represents a preliminary proposal based on work-in-progress within the IETF WEBTRANS WG. Since the [WEB-TRANSPORT-HTTP3] and [WEB-TRANSPORT-HTTP2] specifications are a work-in-progress, both the protocol and API are likely to change significantly going forward.

2. Conformance

As well as sections marked as non-normative, all authoring guidelines, diagrams, examples, and notes in this specification are non-normative. Everything else in this specification is normative.

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

This specification defines conformance criteria that apply to a single product: the user agent that implements the interfaces that it contains.

Conformance requirements phrased as algorithms or specific steps may be implemented in any manner, so long as the end result is equivalent. (In particular, the algorithms defined in this specification are intended to be easy to follow, and not intended to be performant.)

Implementations that use ECMAScript to implement the APIs defined in this specification MUST implement them in a manner consistent with the ECMAScript Bindings defined in the Web IDL specification [WEBIDL], as this specification uses that specification and terminology.

3. Protocol concepts

There are two main protocol concepts for WebTransport: sessions and streams. Each WebTransport session can contain multiple WebTransport streams.

3.1. WebTransport session

A WebTransport session is a session of WebTransport over an HTTP/3 or HTTP/2 underlying connection. There may be multiple WebTransport sessions on one connection, when pooling is enabled.

A WebTransport session has the following capabilities defined in [WEB-TRANSPORT-OVERVIEW]:

capability definition
send a datagram [WEB-TRANSPORT-OVERVIEW] Section 4.2
receive a datagram [WEB-TRANSPORT-OVERVIEW] Section 4.2
create an outgoing unidirectional stream [WEB-TRANSPORT-OVERVIEW] Section 4.3
create a bidirectional stream [WEB-TRANSPORT-OVERVIEW] Section 4.3
receive an incoming unidirectional stream [WEB-TRANSPORT-OVERVIEW] Section 4.3
receive a bidirectional stream [WEB-TRANSPORT-OVERVIEW] Section 4.3

To establish a WebTransport session with an origin origin, follow [WEB-TRANSPORT-OVERVIEW] Section 4.1, with using origin, serialized and isomorphic encoded, as the `Origin` header of the request. When establishing a session, the client MUST NOT provide any credentials. The resulting underlying transport stream is referred to as the session’s CONNECT stream.

A WebTransport session session is draining when the CONNECT stream receives an DRAIN_WEBTRANSPORT_SESSION capsule, or when a GOAWAY frame is received, as described in [WEB-TRANSPORT-HTTP3] Section 4.6.

To terminate a WebTransport session session with an optional integer code and an optional byte sequence reason, follow [WEB-TRANSPORT-OVERVIEW] Section 4.1.

A WebTransport session session is terminated, with optionally an integer code and a byte sequence reason, when the CONNECT stream is closed by the server, as described at [WEB-TRANSPORT-OVERVIEW] Section 4.1.

A WebTransport session has the following signals:

event definition
DRAIN_WEBTRANSPORT_SESSION [WEB-TRANSPORT-HTTP3] Section 4.6
GOAWAY [WEB-TRANSPORT-HTTP3] Section 4.6

3.2. WebTransport stream

A WebTransport stream is a concept for a reliable in-order stream of bytes on a WebTransport session, as described in [WEB-TRANSPORT-OVERVIEW] Section 4.3.

A WebTransport stream is one of incoming unidirectional, outgoing unidirectional or bidirectional.

A WebTransport stream has the following capabilities:

capability definition incoming unidirectional outgoing unidirectional bidirectional
send bytes (potentially with FIN) [WEB-TRANSPORT-OVERVIEW] Section 4.3 No Yes Yes
receive bytes (potentially with FIN) [WEB-TRANSPORT-OVERVIEW] Section 4.3 Yes No Yes
send STOP_SENDING [WEB-TRANSPORT-OVERVIEW] Section 4.3 Yes No Yes
reset a WebTransport stream [WEB-TRANSPORT-OVERVIEW] Section 4.3 No Yes Yes

A WebTransport stream has the following signals:

event definition incoming unidirectional outgoing unidirectional bidirectional
STOP_SENDING [WEB-TRANSPORT-OVERVIEW] Section 4.3 No Yes Yes
RESET_STREAM [WEB-TRANSPORT-OVERVIEW] Section 4.3 Yes No Yes
flow control [WEB-TRANSPORT-OVERVIEW] Section 4.3 No Yes Yes

4. WebTransportDatagramDuplexStream Interface

A WebTransportDatagramDuplexStream is a generic duplex stream.

[Exposed=(Window,Worker), SecureContext]
interface WebTransportDatagramDuplexStream {
  readonly attribute ReadableStream readable;
  readonly attribute WritableStream writable;

  readonly attribute unsigned long maxDatagramSize;
  attribute unrestricted double? incomingMaxAge;
  attribute unrestricted double? outgoingMaxAge;
  attribute unrestricted double incomingHighWaterMark;
  attribute unrestricted double outgoingHighWaterMark;
};

4.1. Internal slots

A WebTransportDatagramDuplexStream object has the following internal slots.

Internal Slot Description (non-normative)
[[Readable]] A ReadableStream for incoming datagrams.
[[Writable]] A WritableStream for outgoing datagrams.
[[IncomingDatagramsQueue]] A queue of pairs of an incoming datagram and a timestamp.
[[IncomingDatagramsPullPromise]] A promise set by pullDatagrams, to wait for an incoming datagram.
[[IncomingDatagramsHighWaterMark]] An unrestricted double representing the high water mark of the incoming datagrams.
[[IncomingDatagramsExpirationDuration]] An unrestricted double representing the expiration duration for incoming datagrams (in milliseconds), or null.
[[OutgoingDatagramsQueue]] A queue of tuples of an outgoing datagram, a timestamp and a promise which is resolved when the datagram is sent or discarded.
[[OutgoingDatagramsHighWaterMark]] An unrestricted double representing the high water mark of the outgoing datagrams.
[[OutgoingDatagramsExpirationDuration]] An unrestricted double value representing the expiration duration for outgoing datagrams (in milliseconds), or null.
[[OutgoingMaxDatagramSize]] An integer representing the maximum size for an outgoing datagram.

The user agent MAY update [[OutgoingMaxDatagramSize]] for any WebTransport object whose [[State]] is either "connecting" or "connected".

To create a WebTransportDatagramDuplexStream given a readable, and a writable, perform the following steps.

  1. Let stream be a new WebTransportDatagramDuplexStream, with:

    [[Readable]]

    readable

    [[Writable]]

    writable

    [[IncomingDatagramsQueue]]

    an empty queue

    [[IncomingDatagramsPullPromise]]

    null

    [[IncomingDatagramsHighWaterMark]]

    an implementation-defined value

    [[IncomingDatagramsExpirationDuration]]

    null

    [[OutgoingDatagramsQueue]]

    an empty queue

    [[OutgoingDatagramsHighWaterMark]]

    an implementation-defined value

    This implementation-defined value should be tuned to ensure decent throughput, without jeopardizing the timeliness of transmitted data.

    [[OutgoingDatagramsExpirationDuration]]

    null

    [[OutgoingMaxDatagramSize]]

    an implementation-defined integer.

  2. Return stream.

4.2. Attributes

readable, of type ReadableStream, readonly

The getter steps are:

  1. Return this.[[Readable]].

writable, of type WritableStream, readonly

The getter steps are:

  1. Return this.[[Writable]].

incomingMaxAge, of type unrestricted double, nullable

The getter steps are:

  1. Return this.[[IncomingDatagramsExpirationDuration]].

The setter steps, given value, are:

  1. If value is negative or NaN, throw a RangeError.

  2. If value is 0, set value to null.

  3. Set this.[[IncomingDatagramsExpirationDuration]] to value.

maxDatagramSize, of type unsigned long, readonly

The maximum size data that may be passed to writable. The getter steps are to return this.[[OutgoingMaxDatagramSize]].

outgoingMaxAge, of type unrestricted double, nullable

The getter steps are:

  1. Return this's [[OutgoingDatagramsExpirationDuration]].

The setter steps, given value, are:

  1. If value is negative or NaN, throw a RangeError.

  2. If value is 0, set value to null.

  3. Set this.[[OutgoingDatagramsExpirationDuration]] to value.

incomingHighWaterMark, of type unrestricted double

The getter steps are:

  1. Return this.[[IncomingDatagramsHighWaterMark]].

The setter steps, given value, are:

  1. If value is negative or NaN, throw a RangeError.

  2. If value is < 1, set value to 1.

  3. Set this.[[IncomingDatagramsHighWaterMark]] to value.

outgoingHighWaterMark, of type unrestricted double

The getter steps are:

  1. Return this.[[OutgoingDatagramsHighWaterMark]].

The setter steps, given value, are:

  1. If value is negative or NaN, throw a RangeError.

  2. If value is < 1, set value to 1.

  3. Set this.[[OutgoingDatagramsHighWaterMark]] to value.

4.3. Procedures

To pullDatagrams, given a WebTransport object transport, run these steps:

  1. Let datagrams be transport.[[Datagrams]].

  2. Assert: datagrams.[[IncomingDatagramsPullPromise]] is null.

  3. Let queue be datagrams.[[IncomingDatagramsQueue]].

  4. If queue is empty, then:

    1. Set datagrams.[[IncomingDatagramsPullPromise]] to a new promise.

    2. Return datagrams.[[IncomingDatagramsPullPromise]].

  5. Let datagram and timestamp be the result of dequeuing queue.

  6. If datagrams.[[Readable]]'s current BYOB request view is not null, then:

    1. Let view be datagrams.[[Readable]]'s current BYOB request view.

    2. If view’s byte length is less than the size of datagram, return a promise rejected with a RangeError.

    3. Let elementSize be the element size specified in the typed array constructors table for view.[[TypedArrayName]]. If view does not have a [[TypedArrayName]] internal slot (i.e. it is a DataView), let elementSize be 0.

    4. If elementSize is not 1, return a promise rejected with a TypeError.

  7. Pull from bytes datagram into datagrams.[[Readable]].

  8. Return a promise resolved with undefined.

To receiveDatagrams, given a WebTransport object transport, run these steps:

  1. Let timestamp be a timestamp representing now.

  2. Let queue be datagrams.[[IncomingDatagramsQueue]].

  3. Let duration be datagrams.[[IncomingDatagramsExpirationDuration]].

  4. If duration is null, then set duration to an implementation-defined value.

  5. Let session be transport.[[Session]].

  6. While there are available incoming datagrams on session:

    1. Let datagram be the result of receiving a datagram with session.

    2. Let timestamp be a timestamp representing now.

    3. Let chunk be a pair of datagram and timestamp.

    4. Enqueue chunk to queue.

  7. Let toBeRemoved be the length of queue minus datagrams.[[IncomingDatagramsHighWaterMark]].

  8. If toBeRemoved is positive, repeat dequeuing queue toBeRemoved (rounded down) times.

  9. While queue is not empty:

    1. Let bytes and timestamp be queue’s first element.

    2. If more than duration milliseconds have passed since timestamp, then dequeue queue.

    3. Otherwise, break this loop.

  10. If queue is not empty and datagrams.[[IncomingDatagramsPullPromise]] is non-null, then:

    1. Let bytes and timestamp be the result of dequeuing queue.

    2. Let promise be datagrams.[[IncomingDatagramsPullPromise]].

    3. Set datagrams.[[IncomingDatagramsPullPromise]] to null.

    4. Queue a network task with transport to run the following steps:

      1. Let chunk be a new Uint8Array object representing bytes.

      2. Enqueue chunk to datagrams.[[Readable]].

      3. Resolve promise with undefined.

The user agent SHOULD run receiveDatagrams for any WebTransport object whose [[State]] is "connected" as soon as reasonably possible whenever the algorithm can make progress.

The writeDatagrams algorithm is given a transport as parameter and data as input. It is defined by running the following steps:

  1. Let timestamp be a timestamp representing now.

  2. If data is not a BufferSource object, then return a promise rejected with a TypeError.

  3. Let datagrams be transport.[[Datagrams]].

  4. If datagrams.[[OutgoingMaxDatagramSize]] is less than data’s [[ByteLength]], return a promise resolved with undefined.

  5. Let promise be a new promise.

  6. Let bytes be a copy of bytes which data represents.

  7. Let chunk be a tuple of bytes, timestamp and promise.

  8. Enqueue chunk to datagrams.[[OutgoingDatagramsQueue]].

  9. If the length of datagrams.[[OutgoingDatagramsQueue]] is less than datagrams.[[OutgoingDatagramsHighWaterMark]], then resolve promise with undefined.

  10. Return promise.

Note: The associated WritableStream calls writeDatagrams only when all the promises that have been returned by writeDatagrams have been resolved. Hence the timestamp and the expiration duration work well only when the web developer pays attention to WritableStreamDefaultWriter.ready.

To sendDatagrams, given a WebTransport object transport, run these steps:

  1. Let queue be datagrams.[[OutgoingDatagramsQueue]].

  2. Let duration be datagrams.[[OutgoingDatagramsExpirationDuration]].

  3. If duration is null, then set duration to an implementation-defined value.

  4. While queue is not empty:

    1. Let bytes, timestamp and promise be queue’s first element.

    2. If more than duration milliseconds have passed since timestamp, then:

      1. Remove the first element from queue.

      2. Queue a network task with transport to resolve promise with undefined.

    3. Otherwise, break this loop.

  5. If transport.[[State]] is not "connected", then return.

  6. Let maxSize be datagrams.[[OutgoingMaxDatagramSize]].

  7. While queue is not empty:

    1. Let bytes, timestamp and promise be queue’s first element.

    2. If bytes’s length ≤ maxSize:

      1. If it is not possible to send bytes to the network immediately, then break this loop.

      2. Send a datagram, with transport.[[Session]] and bytes.

    3. Remove the first element from queue.

    4. Queue a network task with transport to resolve promise with undefined.

The user agent SHOULD run sendDatagrams for any WebTransport object whose [[State]] is "connecting" or "connected" as soon as reasonably possible whenever the algorithm can make progress.

Note: Writing datagrams while the transport’s [[State]] is "connecting" is allowed. The datagrams are stored in [[OutgoingDatagramsQueue]], and they can be discarded in the same manner as when in the "connected" state. Once the transport’s [[State]] becomes "connected", it will start sending the queued datagrams.

5. WebTransport Interface

WebTransport provides an API to the underlying transport functionality defined in [WEB-TRANSPORT-OVERVIEW].

[Exposed=(Window,Worker), SecureContext]
interface WebTransport {
  constructor(USVString url, optional WebTransportOptions options = {});

  Promise<WebTransportConnectionStats> getStats();
  readonly attribute Promise<undefined> ready;
  readonly attribute WebTransportReliabilityMode reliability;
  readonly attribute WebTransportCongestionControl congestionControl;
  [EnforceRange] attribute unsigned short? anticipatedConcurrentIncomingUnidirectionalStreams;
  [EnforceRange] attribute unsigned short? anticipatedConcurrentIncomingBidirectionalStreams;

  readonly attribute Promise<WebTransportCloseInfo> closed;
  readonly attribute Promise<undefined> draining;
  undefined close(optional WebTransportCloseInfo closeInfo = {});

  readonly attribute WebTransportDatagramDuplexStream datagrams;

  Promise<WebTransportBidirectionalStream> createBidirectionalStream(
      optional WebTransportSendStreamOptions options = {});
  /* a ReadableStream of WebTransportBidirectionalStream objects */
  readonly attribute ReadableStream incomingBidirectionalStreams;

  Promise<WebTransportSendStream> createUnidirectionalStream(
      optional WebTransportSendStreamOptions options = {});
  /* a ReadableStream of WebTransportReceiveStream objects */
  readonly attribute ReadableStream incomingUnidirectionalStreams;
  WebTransportSendGroup createSendGroup();

  static readonly attribute boolean supportsReliableOnly;
};

enum WebTransportReliabilityMode {
  "pending",
  "reliable-only",
  "supports-unreliable",
};

5.1. Internal slots

A WebTransport object has the following internal slots.

Internal Slot Description (non-normative)
[[SendStreams]] An ordered set of WebTransportSendStreams owned by this WebTransport.
[[ReceiveStreams]] An ordered set of WebTransportReceiveStreams owned by this WebTransport.
[[IncomingBidirectionalStreams]] A ReadableStream consisting of WebTransportBidirectionalStream objects.
[[IncomingUnidirectionalStreams]] A ReadableStream consisting of WebTransportReceiveStreams.
[[State]] An enum indicating the state of the transport. One of "connecting", "connected", "draining", "closed", and "failed".
[[Ready]] A promise fulfilled when the associated WebTransport session gets established, or rejected if the establishment process failed.
[[Reliability]] A WebTransportReliabilityMode indicating whether the first hop supports unreliable (UDP) transport or whether only reliable (TCP fallback) transport is available. Returns "pending" until a connection has been established.
[[CongestionControl]] A WebTransportCongestionControl indicating whether a preference for a congestion control algorithm optimized for throughput or low latency was requested by the application and satisfied by the user agent, or "default".
[[AnticipatedConcurrentIncomingUnidirectionalStreams]] The number of concurrently open incoming unidirectional streams the application anticipates the server creating, or null.
[[AnticipatedConcurrentIncomingBidirectionalStreams]] The number of concurrently open bidirectional streams the application anticipates the server creating, or null.
[[Closed]] A promise fulfilled when the associated WebTransport object is closed gracefully, or rejected when it is closed abruptly or failed on initialization.
[[Draining]] A promise fulfilled when the associated WebTransport session receives a DRAIN_WEBTRANSPORT_SESSION capsule or a GOAWAY frame.
[[Datagrams]] A WebTransportDatagramDuplexStream.
[[Session]] A WebTransport session for this WebTransport object, or null.

5.2. Constructor

When the WebTransport() constructor is invoked, the user agent MUST run the following steps:
  1. Let parsedURL be the URL record resulting from parsing url.

  2. If parsedURL is a failure, throw a SyntaxError exception.

  3. If parsedURL scheme is not https, throw a SyntaxError exception.

  4. If parsedURL fragment is not null, throw a SyntaxError exception.

  5. Let allowPooling be options's allowPooling.

  6. Let dedicated be the negation of allowPooling.

  7. Let serverCertificateHashes be options's serverCertificateHashes if it exists, and null otherwise.

  8. If dedicated is false and serverCertificateHashes is non-null, then throw a NotSupportedError exception.

  9. Let requireUnreliable be options's requireUnreliable.

  10. Let congestionControl be options's congestionControl.

  11. If congestionControl is not "default", and the user agent does not support any congestion control algorithms that optimize for congestionControl, as allowed by [RFC9002] Section 7, then set congestionControl to "default".

  12. Let anticipatedConcurrentIncomingUnidirectionalStreams be options's anticipatedConcurrentIncomingUnidirectionalStreams.

  13. Let anticipatedConcurrentIncomingBidirectionalStreams be options's anticipatedConcurrentIncomingBidirectionalStreams.

  14. Let incomingDatagrams be a new ReadableStream.

  15. Let outgoingDatagrams be a new WritableStream.

  16. Let datagrams be the result of creating a WebTransportDatagramDuplexStream, its readable set to incomingDatagrams and its writable set to outgoingDatagrams.

  17. Let transport be a newly constructed WebTransport object, with:

    [[SendStreams]]

    an empty ordered set

    [[ReceiveStreams]]

    an empty ordered set

    [[IncomingBidirectionalStreams]]

    a new ReadableStream

    [[IncomingUnidirectionalStreams]]

    a new ReadableStream

    [[State]]

    "connecting"

    [[Ready]]

    a new promise

    [[Reliability]]

    "pending"

    [[CongestionControl]]

    congestionControl

    [[AnticipatedConcurrentIncomingUnidirectionalStreams]]

    anticipatedConcurrentIncomingUnidirectionalStreams

    [[AnticipatedConcurrentIncomingBidirectionalStreams]]

    anticipatedConcurrentIncomingBidirectionalStreams

    [[Closed]]

    a new promise

    [[Draining]]

    a new promise

    [[Datagrams]]

    datagrams

    [[Session]]

    null

  18. Let pullDatagramsAlgorithm be an action that runs pullDatagrams with transport.

  19. Let writeDatagramsAlgorithm be an action that runs writeDatagrams with transport.

Note: Using 64kB buffers with datagrams is recommended because the effective maximum WebTransport datagram frame size has an upper bound of the QUIC maximum datagram frame size which is recommended to be 64kB (See [QUIC-DATAGRAM] Section 3). This will ensure the stream is not errored due to a datagram being larger than the buffer.

  1. Set up with byte reading support incomingDatagrams with pullAlgorithm set to pullDatagramsAlgorithm, and highWaterMark set to 0.

  2. Set up outgoingDatagrams with writeAlgorithm set to writeDatagramsAlgorithm.

  3. Let pullBidirectionalStreamAlgorithm be an action that runs pullBidirectionalStream with transport.

  4. Set up transport.[[IncomingBidirectionalStreams]] with pullAlgorithm set to pullBidirectionalStreamAlgorithm, and highWaterMark set to 0.

  5. Let pullUnidirectionalStreamAlgorithm be an action that runs pullUnidirectionalStream with transport.

  6. Set up transport.[[IncomingUnidirectionalStreams]] with pullAlgorithm set to pullUnidirectionalStreamAlgorithm, and highWaterMark set to 0.

  7. Initialize WebTransport over HTTP with transport, parsedURL, dedicated, requireUnreliable, congestionControl, and serverCertificateHashes.

  8. Return transport.

To initialize WebTransport over HTTP, given a WebTransport object transport, a URL record url, a boolean dedicated, a boolean http3Only, a WebTransportCongestionControl congestionControl, and a sequence<WebTransportHash> serverCertificateHashes, run these steps.
  1. Let client be transport’s relevant settings object.

  2. Let origin be client’s origin.

  3. Let request be a new request whose URL is url, client is client, policy container is client’s policy container, destination is an empty string, and origin is origin.

  4. Run report Content Security Policy violations for request.

  5. If should request be blocked by Content Security Policy? with request returns "Blocked", or if request should be blocked due to a bad port returns blocked, then abort the remaining steps and queue a network task with transport to run these steps:

    1. If transport.[[State]] is "closed" or "failed", then abort these steps.

    2. Let error be a newly created WebTransportError whose source is "session".

    3. Cleanup transport with error.

  6. Let networkPartitionKey be the result of determining the network partition key with transport’s relevant settings object.

  7. Run the following steps in parallel, but abort when transport.[[State]] becomes "closed" or "failed":

    1. Let newConnection be "no" if dedicated is false; otherwise "yes-and-dedicated".

    2. Let connection be the result of obtaining a connection with networkPartitionKey, url, false, newConnection, and http3Only. If the user agent supports more than one congestion control algorithm, choose one appropriate for congestionControl for sending of data on this connection. When obtaining a connection, if serverCertificateHashes is specified, instead of using the default certificate verification algorithm, consider the certificate valid if it meets the custom certificate requirements and if verifying the certificate hash against serverCertificateHashes returns true. If either condition is not met, let connection be failure.

    3. If connection is failure, then abort the remaining steps and queue a network task with transport to run these steps:

      1. If transport.[[State]] is "closed" or "failed", then abort these steps.

      2. Let error be a newly created WebTransportError whose source is "session".

      3. Cleanup transport with error.

    4. Wait for connection to receive the first SETTINGS frame, and let settings be a dictionary that represents the SETTINGS frame.

    5. If settings doesn’t contain SETTINGS_ENABLE_WEBTRANPORT with a value of 1, or it doesn’t contain H3_DATAGRAM with a value of 1, then abort the remaining steps and queue a network task with transport to run these steps:

      1. If transport.[[State]] is "closed" or "failed", then abort these steps.

      2. Let error be a newly created WebTransportError whose source is "session".

      3. Cleanup transport with error.

    6. Establish a WebTransport session with origin on connection.

      Note: This step also contains the transport parameter exchange specified in [QUIC-DATAGRAM].

    7. If the previous step fails, abort the remaining steps and queue a network task with transport to run these steps:

      1. If transport.[[State]] is "closed" or "failed", then abort these steps.

      2. Let error be a newly created WebTransportError whose source is "session".

      3. Cleanup transport with error.

    8. Let session be the established WebTransport session.

    9. Assert: maxDatagramSize is an integer.

    10. Queue a network task with transport to run these steps:

      1. If transport.[[State]] is not "connecting":

        1. In parallel, terminate session.

        2. Abort these steps.

      2. Set transport.[[State]] to "connected".

      3. Set transport.[[Session]] to session.

      4. If the connection is an HTTP/3 connection, set transport’s [[Reliability]] to "supports-unreliable".

      5. If the connection is an HTTP/2 connection [WEB-TRANSPORT-HTTP2], set transport’s [[Reliability]] to "reliable-only".

      6. Resolve transport.[[Ready]] with undefined.

To pullBidirectionalStream, given a WebTransport object transport, run these steps.
  1. If transport.[[State]] is "connecting", then return the result of performing the following steps upon fulfillment of transport.[[Ready]]:

    1. Return the result of pullBidirectionalStream with transport.

  2. If transport.[[State]] is not "connected", then return a new rejected promise with an InvalidStateError.

  3. Let session be transport.[[Session]].

  4. Let p be a new promise.

  5. Run the following steps in parallel:

    1. Wait until there is an available incoming bidirectional stream.

    2. Let internalStream be the result of receiving a bidirectional stream.

    3. Queue a network task with transport to run these steps:

      1. Let stream be the result of creating a WebTransportBidirectionalStream with internalStream and transport.

      2. Enqueue stream to transport.[[IncomingBidirectionalStreams]].

      3. Resolve p with undefined.

  6. Return p.

To pullUnidirectionalStream, given a WebTransport object transport, run these steps.
  1. If transport.[[State]] is "connecting", then return the result of performing the following steps upon fulfillment of transport.[[Ready]]:

    1. Return the result of pullUnidirectionalStream with transport.

  2. If transport.[[State]] is not "connected", then return a new rejected promise with an InvalidStateError.

  3. Let session be transport.[[Session]].

  4. Let p be a new promise.

  5. Run the following steps in parallel:

    1. Wait until there is an available incoming unidirectional stream.

    2. Let internalStream be the result of receiving an incoming unidirectional stream.

    3. Queue a network task with transport to run these steps:

      1. Let stream be the result of creating a WebTransportReceiveStream with internalStream and transport.

      2. Enqueue stream to transport.[[IncomingUnidirectionalStreams]].

      3. Resolve p with undefined.

  6. Return p.

5.3. Attributes

ready, of type Promise<undefined>, readonly

On getting, it MUST return this's [[Ready]].

closed, of type Promise<WebTransportCloseInfo>, readonly

On getting, it MUST return this's [[Closed]].

draining, of type Promise<undefined>, readonly

On getting, it MUST return this's [[Draining]].

datagrams, of type WebTransportDatagramDuplexStream, readonly

A single duplex stream for sending and receiving datagrams over this session. The getter steps for the datagrams attribute SHALL be:

  1. Return this's [[Datagrams]].

incomingBidirectionalStreams, of type ReadableStream, readonly

Returns a ReadableStream of WebTransportBidirectionalStreams that have been received from the server. The getter steps for the incomingBidirectionalStreams attribute SHALL be:

  1. Return this's [[IncomingBidirectionalStreams]].

incomingUnidirectionalStreams, of type ReadableStream, readonly

A ReadableStream of unidirectional streams, each represented by a WebTransportReceiveStream, that have been received from the server. The getter steps for incomingUnidirectionalStreams are:

  1. Return this.[[IncomingUnidirectionalStreams]].

reliability, of type WebTransportReliabilityMode, readonly

Whether connection supports unreliable (over UDP) transport or only reliable (over TCP fallback) transport. Returns "pending" until a connection has been established. The getter steps are to return this's [[Reliability]].

congestionControl, of type WebTransportCongestionControl, readonly

The application’s preference, if requested in the constructor, and satisfied by the user agent, for a congestion control algorithm optimized for either throughput or low latency for sending on this connection. If a preference was requested but not satisfied, then the value is "default" The getter steps are to return this's [[CongestionControl]].

supportsReliableOnly, of type boolean, readonly

Returns true if the user agent supports WebTransport sessions over exclusively reliable connections, otherwise false.

anticipatedConcurrentIncomingUnidirectionalStreams, of type unsigned short, nullable

Optionally lets an application specify the number of concurrently open incoming unidirectional streams it anticipates the server creating. If not null, the user agent SHOULD attempt to reduce future round-trips by taking [[AnticipatedConcurrentIncomingUnidirectionalStreams]] into consideration in its negotiations with the server.

The getter steps are to return this's [[AnticipatedConcurrentIncomingUnidirectionalStreams]].

The setter steps, given value, are to set this's [[AnticipatedConcurrentIncomingUnidirectionalStreams]] to value.

anticipatedConcurrentIncomingBidirectionalStreams, of type unsigned short, nullable

Optionally lets an application specify the number of concurrently open bidirectional streams it anticipates the server creating. If not null, the user agent SHOULD attempt to reduce future round-trips by taking [[AnticipatedConcurrentIncomingBidirectionalStreams]] into consideration in its negotiations with the server.

The getter steps are to return this's [[AnticipatedConcurrentIncomingBidirectionalStreams]].

The setter steps, given value, are to set this's [[AnticipatedConcurrentIncomingBidirectionalStreams]] to value.

Note: Setting anticipatedConcurrentIncomingUnidirectionalStreams or anticipatedConcurrentIncomingBidirectionalStreams does not guarantee the application will receive the number of streams it anticipates.

5.4. Methods

close(closeInfo)

Terminates the WebTransport session associated with the WebTransport object.

When close is called, the user agent MUST run the following steps:

  1. Let transport be this.

  2. If transport.[[State]] is "closed" or "failed", then abort these steps.

  3. If transport.[[State]] is "connecting":

    1. Let error be a newly created WebTransportError whose source is "session".

    2. Cleanup transport with error.

    3. Abort these steps.

  4. Let session be transport.[[Session]].

  5. Let code be closeInfo.closeCode.

  6. Let reasonString be the maximal code unit prefix of closeInfo.reason where the length of the UTF-8 encoded prefix doesn’t exceed 1024.

  7. Let reason be reasonString, UTF-8 encoded.

  8. In parallel, terminate session with code and reason.

    Note: This also resets or sends STOP_SENDING WebTransport streams contained in transport.[[SendStreams]] and [[ReceiveStreams]].

  9. Cleanup transport with AbortError and closeInfo.

getStats()

Gathers stats for this WebTransport's underlying connection and reports the result asynchronously.

When getStats is called, the user agent MUST run the following steps:

  1. Let transport be this.

  2. Let p be a new promise.

  3. If transport.[[State]] is "failed", reject p with an InvalidStateError and abort these steps.

  4. Run the following steps in parallel:

    1. If transport.[[State]] is "connecting", wait until it changes.

    2. If transport.[[State]] is "failed", reject p with an InvalidStateError and abort these steps.

  1. If transport.[[State]] is "closed", resolve p with the most recent stats available for the connection and abort these steps. The exact point at which those stats are collected is implementation-defined.

  2. Gather the stats from the underlying connection, including stats on datagrams.

    1. Queue a network task with transport to run the following steps:

      1. Let stats be a new WebTransportConnectionStats object representing the gathered stats.

      2. Resolve p with stats.

    1. Return p.

createBidirectionalStream()

Creates a WebTransportBidirectionalStream object for an outgoing bidirectional stream. Note that the mere creation of a stream is not immediately visible to the peer until it is used to send data.

When createBidirectionalStream is called, the user agent MUST run the following steps:

  1. Let transport be this.

  2. If transport.[[State]] is "closed" or "failed", return a new rejected promise with an InvalidStateError.

  3. Let sendGroup be options's sendGroup.

  4. Let sendOrder be options's sendOrder.

  5. Let waitUntilAvailable be options's waitUntilAvailable.

  6. Let p be a new promise.

  7. Run the following steps in parallel, but abort when transport’s [[State]] becomes "closed" or "failed", and instead queue a network task with transport to reject p with an InvalidStateError:

    1. Let streamId be a new stream ID that is valid and unique for transport.[[Session]], as defined in [QUIC] Section 19.11. If one is not immediately available due to exhaustion, wait for it to become available if waitUntilAvailable is true, reject p with a QuotaExceededError and abort these steps otherwise.

    2. Let internalStream be the result of creating a bidirectional stream with transport.[[Session]] and streamId.

    3. Queue a network task with transport to run the following steps:

      1. If transport.[[State]] is "closed" or "failed", reject p with an InvalidStateError and abort these steps.

      2. Let stream be the result of creating a WebTransportBidirectionalStream with internalStream, transport, sendGroup, and sendOrder.

      3. Resolve p with stream.

  8. Return p.

createUnidirectionalStream()

Creates a WebTransportSendStream for an outgoing unidirectional stream. Note that the mere creation of a stream is not immediately visible to the server until it is used to send data.

When createUnidirectionalStream() method is called, the user agent MUST run the following steps:

  1. Let transport be this.

  2. If transport.[[State]] is "closed" or "failed", return a new rejected promise with an InvalidStateError.

  3. Let sendGroup be options's sendGroup.

  4. Let sendOrder be options's sendOrder.

  5. Let waitUntilAvailable be options's waitUntilAvailable.

  6. Let p be a new promise.

  7. Run the following steps in parallel, but abort when transport’s [[State]] becomes "closed" or "failed", and instead queue a network task with transport to reject p with an InvalidStateError:

    1. Let streamId be a new stream ID that is valid and unique for transport.[[Session]], as defined in [QUIC] Section 19.11. If one is not immediately available due to exhaustion, wait for it to become available if waitUntilAvailable is true, reject p with a QuotaExceededError and abort these steps otherwise.

    2. Let internalStream be the result of creating an outgoing unidirectional stream with transport.[[Session]] and streamId.

    3. Queue a network task with transport to run the following steps:

      1. If transport.[[State]] is "closed" or "failed", reject p with an InvalidStateError and abort these steps.

      2. Let stream be the result of creating a WebTransportSendStream with internalStream, transport, sendGroup, and sendOrder.

      3. Resolve p with stream.

  8. return p.

createSendGroup()

Creates a WebTransportSendGroup.

When createSendGroup() method is called, the user agent MUST run the following steps:

  1. Let transport be this.

  2. If transport.[[State]] is "closed" or "failed", throw an InvalidStateError.

  3. Return the result of creating a WebTransportSendGroup with transport.

5.5. Procedures

To cleanup a WebTransport transport with error and optionally closeInfo, run these steps:
  1. Let sendStreams be a copy of transport.[[SendStreams]].

  2. Let receiveStreams be a copy of transport.[[ReceiveStreams]].

  3. Let outgoingDatagrams be transport.[[Datagrams]].[[Writable]].

  4. Let incomingDatagrams be transport.[[Datagrams]].[[Readable]].

  5. Let ready be transport.[[Ready]].

  6. Let closed be transport.[[Closed]].

  7. Let incomingBidirectionalStreams be transport.[[IncomingBidirectionalStreams]].

  8. Let incomingUnidirectionalStreams be transport.[[IncomingUnidirectionalStreams]].

  9. Set transport.[[SendStreams]] to an empty set.

  10. Set transport.[[ReceiveStreams]] to an empty set.

  11. Set transport.[[Datagrams]].[[OutgoingDatagramsQueue]] to an empty queue.

  12. Set transport.[[Datagrams]].[[IncomingDatagramsQueue]] to an empty queue.

  13. If closeInfo is given, then set transport.[[State]] to "closed". Otherwise, set transport.[[State]] to "failed".

  14. For each stream in sendStreams, run the following steps:

    1. If stream.[[PendingOperation]] is not null, reject stream.[[PendingOperation]] with error.

    2. Error stream with error.

  15. For each stream in receiveStreams, error stream with error.

    Note: Script authors can inject code which runs in Promise resolution synchronously. Hence from here, do not touch transport as it may be mutated by scripts in an unpredictable way. This applies to logic calling this procedure, too.

  16. If closeInfo is given, then:

    1. Resolve closed with closeInfo.

    2. Assert: ready is settled.

    3. Close incomingBidirectionalStreams.

    4. Close incomingUnidirectionalStreams.

    5. Close outgoingDatagrams.

    6. Close incomingDatagrams.

  17. Otherwise:

    1. Reject closed with error.

    2. Reject ready with error.

    3. Error incomingBidirectionalStreams with error.

    4. Error incomingUnidirectionalStreams with error.

    5. Error outgoingDatagrams with error.

    6. Error incomingDatagrams with error.

To queue a network task with a WebTransport transport and a series of steps steps, run these steps:

  1. Queue a global task on the network task source with transport’s relevant global object to run steps.

5.6. Session termination not initiated by the client

Whenever a WebTransport session which is associated with a WebTransport transport is terminated with optionally code and reasonBytes, run these steps:
  1. Queue a network task with transport to run these steps:

    1. If transport.[[State]] is "closed" or "failed", abort these steps.

    2. Let error be a newly created WebTransportError whose source is "session".

    3. Let closeInfo be a new WebTransportCloseInfo.

    4. If code is given, set closeInfo’s closeCode to code.

    5. If reasonBytes is given, set closeInfo’s reason to reasonBytes, UTF-8 decoded.

      Note: No language or direction metadata is available with reasonBytes. First-strong heuristics can be used for direction when displaying the value.

    6. Cleanup transport with error and closeInfo.

Whenever a WebTransport transport’s underlying connection gets a connection error, run these steps:
  1. Queue a network task with transport to run these steps:

    1. If transport.[[State]] is "closed" or "failed", abort these steps.

    2. Let error be a newly created WebTransportError whose source is "session".

    3. Cleanup transport with error.

5.7. Context cleanup steps

This specification defines context cleanup steps as the following steps, given WebTransport transport:

  1. If transport.[[State]] is "connected", then:

    1. Set transport.[[State]] to "failed".

    2. In parallel, terminate transport.[[Session]].

    3. Queue a network task with transport to run the following steps:

      1. Let error be a newly created WebTransportError whose source is "session".

      2. Cleanup transport with error.

  2. If transport.[[State]] is "connecting", set transport.[[State]] to "failed".

    This needs to be done in workers too. See #127 and whatwg/html#6731.

5.8. Garbage Collection

A WebTransport object whose [[State]] is "connecting" must not be garbage collected if [[IncomingBidirectionalStreams]], [[IncomingUnidirectionalStreams]], any WebTransportReceiveStream, or [[Datagrams]].[[Readable]] are locked, or if the ready, draining, or closed promise is being observed.

A WebTransport object whose [[State]] is "connected" must not be garbage collected if [[IncomingBidirectionalStreams]], [[IncomingUnidirectionalStreams]], any WebTransportReceiveStream, or [[Datagrams]].[[Readable]] are locked, or if the draining or closed promise is being observed.

A WebTransport object whose [[State]] is "draining" must not be garbage collected if [[IncomingBidirectionalStreams]], [[IncomingUnidirectionalStreams]], any WebTransportReceiveStream, or [[Datagrams]].[[Readable]] are locked, or if the closed promise is being observed.

A WebTransport object with an established WebTransport session that has data queued to be transmitted to the network, including datagrams in [[Datagrams]].[[OutgoingDatagramsQueue]], must not be garbage collected.

If a WebTransport object is garbage collected while the underlying connection is still open, the user agent must terminate the WebTransport session with an Application Error Code of 0 and Application Error Message of "".

5.9. Configuration

dictionary WebTransportHash {
  DOMString algorithm;
  BufferSource value;
};

dictionary WebTransportOptions {
  boolean allowPooling = false;
  boolean requireUnreliable = false;
  sequence<WebTransportHash> serverCertificateHashes;
  WebTransportCongestionControl congestionControl = "default";
  [EnforceRange] unsigned short? anticipatedConcurrentIncomingUnidirectionalStreams = null;
  [EnforceRange] unsigned short? anticipatedConcurrentIncomingBidirectionalStreams = null;
};

enum WebTransportCongestionControl {
  "default",
  "throughput",
  "low-latency",
};

WebTransportOptions is a dictionary of parameters that determine how the WebTransport session is established and used.

allowPooling, of type boolean, defaulting to false

When set to true, the WebTransport session can be pooled, that is, its underlying connection can be shared with other WebTransport sessions.

requireUnreliable, of type boolean, defaulting to false

When set to true, the WebTransport session cannot be established over an HTTP/2 connection if an HTTP/3 connection is not possible.

serverCertificateHashes, of type sequence<WebTransportHash>

This option is only supported for transports using dedicated connections. For transport protocols that do not support this feature, having this field non-empty SHALL result in a NotSupportedError exception being thrown.

If supported and non-empty, the user agent SHALL deem a server certificate trusted if and only if it can successfully verify a certificate hash against serverCertificateHashes and satisfies custom certificate requirements. The user agent SHALL ignore any hash that uses an unknown algorithm. If empty, the user agent SHALL use certificate verification procedures it would use for normal fetch operations.

This cannot be used with allowPooling.

congestionControl, of type WebTransportCongestionControl, defaulting to "default"

Optionally specifies an application’s preference for a congestion control algorithm tuned for either throughput or low-latency to be used when sending data over this connection. This is a hint to the user agent.

This configuration option is considered a feature at risk due to the lack of implementation in browsers of a congestion control algorithm, at the time of writing, that optimizes for low latency.

anticipatedConcurrentIncomingUnidirectionalStreams, of type unsigned short, nullable, defaulting to null

Optionally lets an application specify the number of concurrently open incoming unidirectional streams it anticipates the server creating. The user agent MUST initially allow at least 100 incoming unidirectional streams from the server. If not null, the user agent SHOULD attempt to reduce round-trips by taking [[AnticipatedConcurrentIncomingUnidirectionalStreams]] into consideration in its negotiations with the server.

anticipatedConcurrentIncomingBidirectionalStreams, of type unsigned short, nullable, defaulting to null

Optionally lets an application specify the number of concurrently open bidirectional streams it anticipates a server creating. The user agent MUST initially allow the server to create at least 100 bidirectional streams. If not null, the user agent SHOULD attempt to reduce round-trips by taking [[AnticipatedConcurrentIncomingBidirectionalStreams]] into consideration in its negotiations with the server.

To compute a certificate hash, given a certificate, perform the following steps:
  1. Let cert be certificate, represented as a DER encoding of Certificate message defined in [RFC5280].

  2. Compute the SHA-256 hash of cert and return the computed value.

To verify a certificate hash, given a certificate and an array of hashes hashes, perform the following steps:
  1. Let referenceHash be the result of computing a certificate hash with certificate.

  2. For every hash hash in hashes:

    1. If hash.value is not null and hash.algorithm is an ASCII case-insensitive match with "sha-256":

      1. Let hashValue be the byte sequence which hash.value represents.

      2. If hashValue is equal to referenceHash, return true.

  3. Return false.

The custom certificate requirements are as follows: the certificate MUST be an X.509v3 certificate as defined in [RFC5280], the key used in the Subject Public Key field MUST be one of the allowed public key algorithms, the current time MUST be within the validity period of the certificate as defined in Section 4.1.2.5 of [RFC5280] and the total length of the validity period MUST NOT exceed two weeks. The user agent MAY impose additional implementation-defined requirements on the certificate.

The exact list of allowed public key algorithms used in the Subject Public Key Info field (and, as a consequence, in the TLS CertificateVerify message) is implementation-defined; however, it MUST include ECDSA with the secp256r1 (NIST P-256) named group ([RFC3279], Section 2.3.5; [RFC8422]) to provide an interoperable default. It MUST NOT contain RSA keys ([RFC3279], Section 2.3.1).

5.10. WebTransportCloseInfo Dictionary

The WebTransportCloseInfo dictionary includes information relating to the error code for closing a WebTransport. This information is used to set the error code and reason for a CONNECTION_CLOSE frame.

dictionary WebTransportCloseInfo {
  unsigned long closeCode = 0;
  USVString reason = "";
};

The dictionary SHALL have the following attributes:

closeCode, of type unsigned long, defaulting to 0

The error code communicated to the peer.

reason, of type USVString, defaulting to ""

The reason for closing the WebTransport.

5.11. WebTransportSendStreamOptions Dictionary

The WebTransportSendStreamOptions is a dictionary of parameters that affect how WebTransportSendStreams created by createUnidirectionalStream and createBidirectionalStream behave.

dictionary WebTransportSendStreamOptions {
  WebTransportSendGroup? sendGroup = null;
  long long sendOrder = 0;
  boolean waitUntilAvailable = false;
};

The dictionary SHALL have the following attributes:

sendGroup, of type WebTransportSendGroup, nullable, defaulting to null

An optional WebTransportSendGroup to group this WebTransportSendStream under, or null.

sendOrder, of type long long, defaulting to 0

A send order number that, if provided, opts the created WebTransportSendStream in to participating in strict ordering. Bytes currently queued on strictly ordered WebTransportSendStreams will be sent ahead of bytes currently queued on other strictly ordered WebTransportSendStreams created with lower send order numbers.

If no send order number is provided, then the order in which the user agent sends bytes from it relative to other WebTransportSendStreams is implementation-defined. User agents are strongly encouraged however to divide bandwidth fairly between all streams that aren’t starved by lower send order numbers.

Note: This is sender-side data prioritization which does not guarantee reception order.

waitUntilAvailable, of type boolean, defaulting to false

If true, the promise returned by the createUnidirectionalStream or createBidirectionalStream call will not be settled until either the underlying connection has sufficient flow control credit to create the stream, or the connection reaches a state in which no further outgoing streams are possible. If false, the promise will be rejected if no flow control window is available at the time of the call.

5.12. WebTransportConnectionStats Dictionary

The WebTransportConnectionStats dictionary includes information on WebTransport-specific stats about the WebTransport session's underlying connection.

Note: When pooling is used, multiple WebTransport sessions pooled on the same connection all receive the same information, i.e. the information is disclosed across pooled sessions holding the same network partition key.

Note: Any unavailable stats will be absent from the WebTransportConnectionStats dictionary.

dictionary WebTransportConnectionStats {
  unsigned long long bytesSent;
  unsigned long long packetsSent;
  unsigned long long bytesLost;
  unsigned long long packetsLost;
  unsigned long long bytesReceived;
  unsigned long long packetsReceived;
  DOMHighResTimeStamp smoothedRtt;
  DOMHighResTimeStamp rttVariation;
  DOMHighResTimeStamp minRtt;
  WebTransportDatagramStats datagrams;
  unsigned long long? estimatedSendRate;
};

The dictionary SHALL have the following attributes:

bytesSent, of type unsigned long long

The number of bytes sent on the underlying connection, including retransmissions. Does not include UDP or any other outer framing.

packetsSent, of type unsigned long long

The number of packets sent on the underlying connection, including those that are determined to have been lost.

bytesLost, of type unsigned long long

The number of bytes lost on the underlying connection (does not monotonically increase, because packets that are declared lost can subsequently be received). Does not include UDP or any other outer framing.

packetsLost, of type unsigned long long

The number of packets lost on the underlying connection (does not monotonically increase, because packets that are declared lost can subsequently be received).

bytesReceived, of type unsigned long long

The number of total bytes received on the underlying connection, including duplicate data for streams. Does not include UDP or any other outer framing.

packetsReceived, of type unsigned long long

The number of total packets received on the underlying connection, including packets that were not processable.

smoothedRtt, of type DOMHighResTimeStamp

The smoothed round-trip time (RTT) currently observed on the connection, as defined in [RFC9002] Section 5.3.

rttVariation, of type DOMHighResTimeStamp

The mean variation in round-trip time samples currently observed on the connection, as defined in [RFC9002] Section 5.3.

minRtt, of type DOMHighResTimeStamp

The minimum round-trip time observed on the entire connection.

estimatedSendRate, of type unsigned long long, nullable

The estimated rate at which queued data will be sent by the user agent, in bits per second. This rate applies to all streams and datagrams that share a WebTransport session and is calculated by the congestion control algorithm (potentially chosen by congestionControl). If the user agent does not currently have an estimate, the member MUST be the null value. The member can be null even if it was not null in previous results.

5.13. WebTransportDatagramStats Dictionary

The WebTransportDatagramStats dictionary includes statistics on datagram transmission over the underlying connection.

dictionary WebTransportDatagramStats {
  unsigned long long droppedIncoming;
  unsigned long long expiredIncoming;
  unsigned long long expiredOutgoing;
  unsigned long long lostOutgoing;
};

The dictionary SHALL have the following attributes:

droppedIncoming, of type unsigned long long

The number of incoming datagrams that were dropped due to the application not reading from datagrams' readable before new datagrams overflow the receive queue.

expiredIncoming, of type unsigned long long

The number of incoming datagrams that were dropped due to being older than incomingMaxAge before they were read from datagrams' readable.

expiredOutgoing, of type unsigned long long

The number of datagrams queued for sending that were dropped due to being older than outgoingMaxAge before they were able to be sent.

lostOutgoing, of type unsigned long long

The number of sent datagrams that were declared lost, as defined in [RFC9002] Section 6.1.

6. Interface WebTransportSendStream

A WebTransportSendStream is a WritableStream providing outgoing streaming features with an outgoing unidirectional or bidirectional WebTransport stream.

It is a WritableStream of Uint8Array that can be written to, to send data to the server.

[Exposed=(Window,Worker), SecureContext, Transferable]
interface WebTransportSendStream : WritableStream {
  attribute WebTransportSendGroup? sendGroup;
  attribute long long sendOrder;
  Promise<WebTransportSendStreamStats> getStats();
  WebTransportWriter getWriter();
};

A WebTransportSendStream is always created by the create procedure.

The WebTransportSendStream's transfer steps and transfer-receiving steps are those of WritableStream.

6.1. Attributes

sendGroup, of type WebTransportSendGroup, nullable

The getter steps are:

  1. Return this's [[SendGroup]].

The setter steps, given value, are:

  1. If value is non-null, and value.[[Transport]] is not this.[[Transport]], throw an InvalidStateError.

  2. Set this.[[SendGroup]] to value.

sendOrder, of type long long

The getter steps are:

  1. Return this's [[SendOrder]].

The setter steps, given value, are:

  1. Set this.[[SendOrder]] to value.

6.2. Methods

getStats()

Gathers stats specific to this WebTransportSendStream's performance, and reports the result asynchronously.

When getStats is called, the user agent MUST run the following steps:

  1. Let p be a new promise.

  2. Run the following steps in parallel:

    1. Gather the stats specific to this WebTransportSendStream.

    2. Wait for the stats to be ready.

    3. Queue a network task with transport to run the following steps:

      1. Let stats be a new WebTransportSendStreamStats object representing the gathered stats.

      2. Resolve p with stats.

  3. Return p.

getWriter()

This method must be implemented in the same manner as getWriter inherited from WritableStream, except in place of creating a WritableStreamDefaultWriter, it must instead create a WebTransportWriter with this.

6.3. Internal Slots

A WebTransportSendStream has the following internal slots.

Internal Slot Description (non-normative)
[[InternalStream]] An outgoing unidirectional or bidirectional WebTransport stream.
[[PendingOperation]] A promise representing a pending write or close operation, or null.
[[Transport]] A WebTransport which owns this WebTransportSendStream.
[[SendGroup]] An optional WebTransportSendGroup, or null.
[[SendOrder]] An optional send order number, defaulting to 0.
[[AtomicWriteRequests]] An ordered set of promises, keeping track of the subset of write requests that are atomic among those queued to be processed by the underlying sink.

6.4. Procedures

To create a WebTransportSendStream, with an outgoing unidirectional or bidirectional WebTransport stream internalStream, a WebTransport transport, sendGroup, and a sendOrder, run these steps:

  1. Let stream be a new WebTransportSendStream, with:

    [[InternalStream]]

    internalStream

    [[PendingOperation]]

    null

    [[Transport]]

    transport

    [[SendGroup]]

    sendGroup

    [[SendOrder]]

    sendOrder

    [[AtomicWriteRequests]]

    An empty ordered set of promises.

  2. Let writeAlgorithm be an action that writes chunk to stream, given chunk.

  3. Let closeAlgorithm be an action that closes stream.

  4. Let abortAlgorithm be an action that aborts stream with reason, given reason.

  5. Set up stream with writeAlgorithm set to writeAlgorithm, closeAlgorithm set to closeAlgorithm, abortAlgorithm set to abortAlgorithm.

  6. Let abortSignal be stream’s [[controller]].[[abortController]].[[signal]].

  7. Add the following steps to abortSignal.

    1. Let pendingOperation be stream.[[PendingOperation]].

    2. If pendingOperation is null, then abort these steps.

    3. Set stream.[[PendingOperation]] to null.

    4. Let reason be abortSignal’s abort reason.

    5. Let promise be the result of aborting stream with reason.

    6. Upon fulfillment of promise, reject pendingOperation with reason.

  8. Append stream to transport.[[SendStreams]].

  9. Return stream.

To write chunk to a WebTransportSendStream stream, run these steps:
  1. Let transport be stream.[[Transport]].

  2. If chunk is not a BufferSource, return a promise rejected with a TypeError.

  3. Let promise be a new promise.

  4. Let bytes be a copy of the byte sequence which chunk represents.

  5. Set stream.[[PendingOperation]] to promise.

  6. Let inFlightWriteRequest be stream.inFlightWriteRequest.

  7. Let atomic be true if stream.[[AtomicWriteRequests]] contains inFlightWriteRequest, otherwise false.

  8. Run the following steps in parallel:

    1. If atomic is true and the current flow control window is too small for bytes to be sent in its entirety, then abort the remaining steps and queue a network task with transport to run these sub-steps:

      1. Set stream.[[PendingOperation]] to null.

      2. Abort all atomic write requests on stream.

    2. Otherwise, send bytes on stream.[[InternalStream]] and wait for the operation to complete. This sending MAY be interleaved with sending of previously queued streams and datagrams, as well as streams and datagrams yet to be queued to be sent over this transport.

      The user-agent MAY have a buffer to improve the transfer performance. Such a buffer SHOULD have a fixed upper limit, to carry the backpressure information to the user of the WebTransportSendStream.

      This sending MUST starve until all bytes queued for sending on WebTransportSendStreams with the same [[SendGroup]] and a higher [[SendOrder]], that are neither errored nor blocked by flow control, have been sent.

      We access stream.[[SendOrder]] in parallel here. User agents SHOULD respond to live updates of these values during sending, though the details are implementation-defined.

      Note: Ordering of retransmissions is implementation-defined, but user agents are strongly encouraged to prioritize retransmissions of data with higher [[SendOrder]] values.

      This sending MUST NOT starve otherwise, except for flow control reasons or error.

      The user agent SHOULD divide bandwidth fairly between all streams that aren’t starved.

      Note: The definition of fairness here is implementation-defined.

    3. If the previous step failed due to a network error, abort the remaining steps.

      Note: We don’t reject promise here because we handle network errors elsewhere, and those steps reject stream.[[PendingOperation]].

    4. Otherwise, queue a network task with transport to run these steps:

      1. Set stream.[[PendingOperation]] to null.

      2. If stream.[[AtomicWriteRequests]] contains inFlightWriteRequest, remove inFlightWriteRequest.

      3. Resolve promise with undefined.

  9. Return promise.

Note: The fulfillment of the promise returned from this algorithm (or, write(chunk)) does NOT necessarily mean that the chunk is acked by the server [QUIC]. It may just mean that the chunk is appended to the buffer. To make sure that the chunk arrives at the server, the server needs to send an application-level acknowledgment message.

To close a WebTransportSendStream stream, run these steps:
  1. Let transport be stream.[[Transport]].

  2. Let promise be a new promise.

  3. Remove stream from transport.[[SendStreams]].

  4. Set stream.[[PendingOperation]] to promise.

  5. Run the following steps in parallel:

    1. Send FIN on stream.[[InternalStream]] and wait for the operation to complete.

    2. Wait for stream.[[InternalStream]] to enter the "Data Recvd" state. [QUIC]

    3. Queue a network task with transport to run these steps:

      1. Set stream.[[PendingOperation]] to null.

      2. Resolve promise with undefined.

  6. Return promise.

To abort a WebTransportSendStream stream with reason, run these steps:
  1. Let transport be stream.[[Transport]].

  2. Let promise be a new promise.

  3. Let code be 0.

  4. Remove stream from transport.[[SendStreams]].

  5. If reason is a WebTransportError and reason.[[StreamErrorCode]] is not null, then set code to reason.[[StreamErrorCode]].

  6. If code < 0, then set code to 0.

  7. If code > 4294967295, then set code to 4294967295.

    Note: Valid values of code are from 0 to 4294967295 inclusive. If the underlying connection is using HTTP/3, the code will be encoded to a number in [0x52e4a40fa8db, 0x52e5ac983162] as decribed in [WEB-TRANSPORT-HTTP3].

  8. Run the following steps in parallel:

    1. Reset stream.[[InternalStream]] with code.

    2. Queue a network task with transport to resolve promise with undefined.

  9. Return promise.

To abort all atomic write requests on a WebTransportSendStream stream, run these steps:
  1. Let writeRequests be stream.writeRequests.

  2. Let requestsToAbort be stream.[[AtomicWriteRequests]].

  3. If writeRequests contains a promise not in requestsToAbort, then error stream with AbortError, and abort these steps.

  4. Empty stream.[[AtomicWriteRequests]].

  5. For each promise in requestsToAbort, reject promise with AbortError.

  6. In parallel, for each promise in requestsToAbort, abort the sending of bytes associated with promise.

6.5. STOP_SENDING signal coming from the server

Whenever a WebTransport stream associated with a WebTransportSendStream stream gets a STOP_SENDING signal from the server, run these steps:
  1. Let transport be stream.[[Transport]].

  2. Let code be the application protocol error code attached to the STOP_SENDING frame. [QUIC]

    Note: Valid values of code are from 0 to 4294967295 inclusive. If the underlying connection is using HTTP/3, the code will be encoded to a number in [0x52e4a40fa8db, 0x52e5ac983162] as decribed in [WEB-TRANSPORT-HTTP3].

  3. Queue a network task with transport to run these steps:

    1. If transport.[[State]] is "closed" or "failed", abort these steps.

    2. Remove stream from transport.[[SendStreams]].

    3. Let error be a newly created WebTransportError whose source is "stream" and streamErrorCode is code.

    4. If stream.[[PendingOperation]] is not null, reject stream.[[PendingOperation]] with error.

    5. Error stream with error.

6.6. WebTransportSendStreamStats Dictionary

The WebTransportSendStreamStats dictionary includes information on stats specific to one WebTransportSendStream.

dictionary WebTransportSendStreamStats {
  unsigned long long bytesWritten;
  unsigned long long bytesSent;
  unsigned long long bytesAcknowledged;
};

The dictionary SHALL have the following attributes:

bytesWritten, of type unsigned long long

The total number of bytes the application has successfully written to this WebTransportSendStream. This number can only increase.

bytesSent, of type unsigned long long

An indicator of progress on how many of the application bytes written to this WebTransportSendStream has been sent at least once. This number can only increase, and is always less than or equal to bytesWritten.

Note: this is progress of app data sent on a single stream only, and does not include any network overhead.

bytesAcknowledged, of type unsigned long long

An indicator of progress on how many of the application bytes written to this WebTransportSendStream have been sent and acknowledged as received by the server using QUIC’s ACK mechanism. Only sequential bytes up to, but not including, the first non-acknowledged byte, are counted. This number can only increase and is always less than or equal to bytesSent.

Note: This value will match bytesSent when the connection is over HTTP/2.

7. Interface WebTransportSendGroup

A WebTransportSendGroup is an optional organizational object that tracks transmission of data spread across many individual (typically strictly ordered) WebTransportSendStreams.

WebTransportSendStreams can, at their creation or through assignment of their sendGroup attribute, be grouped under at most one WebTransportSendGroup at any time. By default, they are ungrouped.

The user agent considers WebTransportSendGroups as equals when allocating bandwidth for sending WebTransportSendStreams. Each WebTransportSendGroup also establishes a separate numberspace for evaluating sendOrder numbers.

[Exposed=(Window,Worker), SecureContext]
interface WebTransportSendGroup {
  Promise<WebTransportSendStreamStats> getStats();
};

A WebTransportSendGroup is always created by the create procedure.

7.1. Methods

getStats()

Aggregates stats from all WebTransportSendStreams grouped under this sendGroup, and reports the result asynchronously.

When getStats is called, the user agent MUST run the following steps:

  1. Let p be a new promise.

  2. Let streams be all WebTransportSendStreams whose [[SendGroup]] is this.

  3. Run the following steps in parallel:

    1. Gather stream statistics from all streams in streams.

    2. Queue a network task with transport to run the following steps:

      1. Let stats be a new WebTransportSendStreamStats object representing the aggregate numbers of the gathered stats.

      2. Resolve p with stats.

  4. Return p.

7.2. Internal Slots

A WebTransportSendGroup has the following internal slots.

Internal Slot Description (non-normative)
[[Transport]] The WebTransport object owning this WebTransportSendGroup.

7.3. Procedures

To create a WebTransportSendGroup, with a WebTransport transport, run these steps:

  1. Let sendGroup be a new WebTransportSendGroup, with:

    [[Transport]]

    transport

  2. Return sendGroup.

8. Interface WebTransportReceiveStream

A WebTransportReceiveStream is a ReadableStream providing incoming streaming features with an incoming unidirectional or bidirectional WebTransport stream.

It is a ReadableStream of Uint8Array that can be read from, to consume data received from the server. WebTransportReceiveStream is a readable byte stream, and hence it allows its consumers to use a BYOB reader as well as a default reader.

[Exposed=(Window,Worker), SecureContext, Transferable]
interface WebTransportReceiveStream : ReadableStream {
  Promise<WebTransportReceiveStreamStats> getStats();
};

A WebTransportReceiveStream is always created by the create procedure.

The WebTransportReceiveStream's transfer steps and transfer-receiving steps are those of ReadableStream.

8.1. Methods

getStats()

Gathers stats specific to this WebTransportReceiveStream's performance, and reports the result asynchronously.

When getStats is called, the user agent MUST run the following steps:

  1. Let p be a new promise.

  2. Run the following steps in parallel:

    1. Gather the stats specific to this WebTransportReceiveStream.

    2. Queue a network task with transport to run the following steps:

      1. Let stats be a new WebTransportReceiveStreamStats object representing the gathered stats.

      2. Resolve p with stats.

  3. Return p.

8.2. Internal Slots

A WebTransportReceiveStream has the following internal slots.

Internal Slot Description (non-normative)
[[InternalStream]] An incoming unidirectional or bidirectional WebTransport stream.
[[Transport]] The WebTransport object owning this WebTransportReceiveStream.

8.3. Procedures

To create a WebTransportReceiveStream, with an incoming unidirectional or bidirectional WebTransport stream internalStream and a WebTransport transport, run these steps:

  1. Let stream be a new WebTransportReceiveStream, with:

    [[InternalStream]]

    internalStream

    [[Transport]]

    transport

  2. Let pullAlgorithm be an action that pulls bytes from stream.

  3. Let cancelAlgorithm be an action that cancels stream with reason, given reason.

  4. Set up with byte reading support stream with pullAlgorithm set to pullAlgorithm and cancelAlgorithm set to cancelAlgorithm.

  5. Append stream to transport.[[ReceiveStreams]].

  6. Return stream.

To pull bytes from a WebTransportReceiveStream stream, run these steps.

  1. Let transport be stream.[[Transport]].

  2. Let internalStream be stream.[[InternalStream]].

  3. Let promise be a new promise.

  4. Let buffer, offset, and maxBytes be null.

  5. If stream’s current BYOB request view for stream is not null:

    1. Set offset to stream’s current BYOB request view.[[ByteOffset]].

    2. Set maxBytes to stream’s current BYOB request view's byte length.

    3. Set buffer to stream’s current BYOB request view's underlying buffer.

  6. Otherwise:

    1. Set offset to 0.

    2. Set maxBytes to an implementation-defined size.

    3. Set buffer be a new ArrayBuffer with maxBytes size. If allocating the ArrayBuffer fails, return a promise rejected with a RangeError.

  7. Run the following steps in parallel:

    1. Write the bytes that area read from internalStream into buffer with offset offset, up to maxBytes bytes. Wait until either at least one byte is read or FIN is received. Let read be the number of read bytes, and let hasReceivedFIN be whether FIN was accompanied.

      The user-agent MAY have a buffer to improve the transfer performance. Such a buffer SHOULD have a fixed upper limit, to carry the backpressure information to the server.

      Note: This operation may return before filling up all of bytes.

    2. If the previous step failed, abort the remaining steps.

      Note: We don’t reject promise here because we handle network errors elsewhere, and those steps error stream, which rejects any read requests awaiting this pull.

    3. Queue a network task with transport to run these steps:

      Note: If the buffer described above is available in the event loop where this procedure is running, the following steps may run immediately.

      1. If read > 0:

        1. Set view to a new Uint8Array with buffer, offset and read.

        2. Enqueue view into stream.

      2. If hasReceivedFIN is true:

        1. Remove stream from transport.[[ReceiveStreams]].

        2. Close stream.

      3. Resolve promise with undefined.

  8. Return promise.

To cancel a WebTransportReceiveStream stream with reason, run these steps.

  1. Let transport be stream.[[Transport]].

  2. Let internalStream be stream.[[InternalStream]].

  3. Let promise be a new promise.

  4. Let code be 0.

  5. If reason is a WebTransportError and reason.[[StreamErrorCode]] is not null, then set code to reason.[[StreamErrorCode]].

  6. If code < 0, then set code to 0.

  7. If code > 4294967295, then set code to 4294967295.

    Note: Valid values of code are from 0 to 4294967295 inclusive. If the underlying connection is using HTTP/3, the code will be encoded to a number in [0x52e4a40fa8db, 0x52e5ac983162] as decribed in [WEB-TRANSPORT-HTTP3].

  8. Remove stream from transport.[[SendStreams]].

  9. Run the following steps in parallel:

    1. Send STOP_SENDING with internalStream and code.

    2. Queue a network task with transport to run these steps:

      Note: If the buffer described above is available in the event loop where this procedure is running, the following steps may run immediately.

      1. Remove stream from transport.[[ReceiveStreams]].

      2. Resolve promise with undefined.

  10. Return promise.

8.4. Reset signal coming from the server

Whenever a WebTransport stream associated with a WebTransportReceiveStream stream gets a RESET_STREAM signal from the server, run these steps:
  1. Let transport be stream.[[Transport]].

  2. Let code be the application protocol error code attached to the RESET_STREAM frame. [QUIC]

    Note: Valid values of code are from 0 to 4294967295 inclusive. If the underlying connection is using HTTP/3, the code will be encoded to a number in [0x52e4a40fa8db, 0x52e5ac983162] as decribed in [WEB-TRANSPORT-HTTP3].

  3. Queue a network task with transport to run these steps:

    1. If transport.[[State]] is "closed" or "failed", abort these steps.

    2. Remove stream from transport.[[ReceiveStreams]].

    3. Let error be a newly created WebTransportError whose source is "stream" and streamErrorCode is code.

    4. Error stream with error.

8.5. WebTransportReceiveStreamStats Dictionary

The WebTransportReceiveStreamStats dictionary includes information on stats specific to one WebTransportReceiveStream.

dictionary WebTransportReceiveStreamStats {
  unsigned long long bytesReceived;
  unsigned long long bytesRead;
};

The dictionary SHALL have the following attributes:

bytesReceived, of type unsigned long long

An indicator of progress on how many of the server application’s bytes intended for this WebTransportReceiveStream have been received so far. Only sequential bytes up to, but not including, the first missing byte, are counted. This number can only increase.

Note: this is progress of app data received on a single stream only, and does not include any network overhead.

bytesRead, of type unsigned long long

The total number of bytes the application has successfully read from this WebTransportReceiveStream. This number can only increase, and is always less than or equal to bytesReceived.

9. Interface WebTransportBidirectionalStream

[Exposed=(Window,Worker), SecureContext]
interface WebTransportBidirectionalStream {
  readonly attribute WebTransportReceiveStream readable;
  readonly attribute WebTransportSendStream writable;
};

9.1. Internal slots

A WebTransportBidirectionalStream has the following internal slots.

Internal Slot Description (non-normative)
[[Readable]] A WebTransportReceiveStream.
[[Writable]] A WebTransportSendStream.
[[Transport]] The WebTransport object owning this WebTransportBidirectionalStream.

9.2. Attributes

readable, of type WebTransportReceiveStream, readonly

The getter steps are to return this's [[Readable]].

writable, of type WebTransportSendStream, readonly

The getter steps are to return this's [[Writable]].

9.3. Procedures

To create a WebTransportBidirectionalStream with a bidirectional WebTransport stream internalStream, a WebTransport object transport, and a sendOrder, run these steps.
  1. Let readable be the result of creating a WebTransportReceiveStream with internalStream and transport.

  2. Let writable be the result of creating a WebTransportSendStream with internalStream, transport, and sendOrder.

  3. Let stream be a new WebTransportBidirectionalStream, with:

    [[Readable]]

    readable

    [[Writable]]

    writable

    [[Transport]]

    transport

  4. Return stream.

10. WebTransportWriter Interface

WebTransportWriter is a subclass of WritableStreamDefaultWriter that adds one method.

A WebTransportWriter is always created by the create procedure.

[Exposed=*, SecureContext]
interface WebTransportWriter : WritableStreamDefaultWriter {
  Promise<undefined> atomicWrite(optional any chunk);
};

10.1. Methods

atomicWrite(chunk)

The atomicWrite method will reject if the chunk given to it could not be sent in its entirety within the flow control window that is current at the time of sending. This behavior is designed to satisfy niche transactional applications sensitive to flow control deadlocks ([RFC9308] Section 4.4).

Note: atomicWrite can still reject after sending some data. Though it provides atomicity with respect to flow control, other errors may occur. atomicWrite does not prevent data from being split between packets or being interleaved with other data. Only the sender learns if atomicWrite fails due to lack of available flow control credit.

Note: Atomic writes can still block if queued behind non-atomic writes. If the atomic write is rejected, everything queued behind it at that moment will be rejected as well. Any non-atomic writes rejected in this way will error the stream. Applications are therefore encouraged to always await atomic writes.

When atomicWrite is called, the user agent MUST run the following steps:

  1. Let p be the result of write(chunk) on WritableStreamDefaultWriter with chunk.

  2. Append p to stream.[[AtomicWriteRequests]].

  3. Return the result of reacting to p with the following steps:

    1. If stream.[[AtomicWriteRequests]] contains p, remove p.

    2. If p was rejected with reason r, then return a promise rejected with r.

    3. Return undefined.

10.2. Procedures

To create a WebTransportWriter, with a WebTransportSendStream stream, run these steps:

  1. Let writer be a new WebTransportWriter.

  2. Run the new WritableStreamDefaultWriter(stream) constructor steps passing writer as this, and stream as the constructor argument.

  3. Return writer.

11. WebTransportError Interface

WebTransportError is a subclass of DOMException that represents

[Exposed=(Window,Worker), Serializable, SecureContext]
interface WebTransportError : DOMException {
  constructor(optional DOMString message = "", optional WebTransportErrorOptions options = {});

  readonly attribute WebTransportErrorSource source;
  readonly attribute unsigned long? streamErrorCode;
};

dictionary WebTransportErrorOptions {
  WebTransportErrorSource source = "stream";
  [Clamp] unsigned long? streamErrorCode = null;
};

enum WebTransportErrorSource {
  "stream",
  "session",
};

11.1. Internal slots

A WebTransportError has the following internal slots.

Internal Slot Description (non-normative)
[[Source]] A WebTransportErrorSource indicating the source of this error.
[[StreamErrorCode]] The application protocol error code for this error, or null.

11.2. Constructor

The new WebTransportError(message, options) constructor steps are:

  1. Set this’s name to "WebTransportError".

  2. Set this’s message to message.

  3. Set this’s internal slots as follows:

    [[Source]]

    options.source

    [[StreamErrorCode]]

    options.streamErrorCode

    Note: This name does not have a mapping to a legacy code, so this's code is 0.

11.3. Attributes

source, of type WebTransportErrorSource, readonly

The getter steps are to return this's [[Source]].

streamErrorCode, of type unsigned long, readonly, nullable

The getter steps are to return this's [[StreamErrorCode]].

11.4. Serialization

WebTransportError objects are serializable objects. Their serialization steps, given value and serialized, are:

  1. Run the DOMException serialization steps given value and serialized.

  2. Set serialized.[[Source]] to value.[[Source]].

  3. Set serialized.[[StreamErrorCode]] to value.[[StreamErrorCode]].

Their deserialization steps, given serialized and value, are:

  1. Run the DOMException deserialization steps given serialized and value.

  2. Set value.[[Source]] to serialized.[[Source]].

  3. Set value.[[StreamErrorCode]] serialized.[[StreamErrorCode]].

12. Protocol Mappings

This section is non-normative.

This section describes the underlying protocol behavior of methods defined in this specification, utilizing [WEB-TRANSPORT-OVERVIEW]. Cause and effect may not be immediate due to buffering.

WebTransport Protocol Action API Effect
received DRAIN_WEBTRANSPORT_SESSION await wt.draining

If the underlying connection is using HTTP/3, the following protocol behaviors from [WEB-TRANSPORT-HTTP3] apply.

The application streamErrorCode in the WebTransportError error is converted to an httpErrorCode, and vice versa, as specified in [WEB-TRANSPORT-HTTP3] Section 4.3.

API Method QUIC Protocol Action
writable.abort(error) sends RESET_STREAM with httpErrorCode
writable.close() sends STREAM with FIN bit set
writable.getWriter().write(chunk)() sends STREAM
writable.getWriter().close() sends STREAM with FIN bit set
writable.getWriter().abort(error) sends RESET_STREAM with httpErrorCode
readable.cancel(error) sends STOP_SENDING with httpErrorCode
readable.getReader().cancel(error) sends STOP_SENDING with httpErrorCode
wt.close(closeInfo) terminates session with closeInfo
QUIC Protocol Action API Effect
received STOP_SENDING with httpErrorCode errors writable with streamErrorCode
received STREAM (await readable.getReader().read()).value
received STREAM with FIN bit set (await readable.getReader().read()).done
received RESET_STREAM with httpErrorCode errors readable with streamErrorCode
Session cleanly terminated with closeInfo
(await wt.closed).closeInfo, and errors open streams
Network error
(await wt.closed) rejects, and errors open streams

Note: As discussed in [QUIC] Section 3.2, receipt of a RESET_STREAM frame is not always indicated to the application. Receipt of the RESET_STREAM can be signaled immediately, interrupting delivery of stream data with any data not consumed being discarded. However, immediate signaling is not required. Also, if stream data is completely received but has not yet been read by the application, the RESET_STREAM signal can be suppressed.

HTTP/3 Protocol Action API Effect
received GOAWAY await wt.draining

If the underlying connection is using HTTP/2, the following protocol behaviors from [WEB-TRANSPORT-HTTP2] apply. Note that, unlike for HTTP/3, the stream error code does not need to be converted to an HTTP error code, and vice versa.

API Method HTTP/2 Protocol Action
writable.abort(error) sends WT_RESET_STREAM with error
writable.close() sends WT_STREAM with FIN bit set
writable.getWriter().write() sends WT_STREAM
writable.getWriter().close() sends WT_STREAM with FIN bit set
writable.getWriter().abort(error) sends WT_RESET_STREAM with error
readable.cancel(error) sends WT_STOP_SENDING with error
readable.getReader().cancel(error) sends WT_STOP_SENDING with error
wt.close(closeInfo) terminates session with closeInfo
HTTP/2 Protocol Action API Effect
received WT_STOP_SENDING with error errors writable with streamErrorCode
received WT_STREAM (await readable.getReader().read()).value
received WT_STREAM with FIN bit set (await readable.getReader().read()).done
received WT_RESET_STREAM with error errors readable with streamErrorCode
Session cleanly terminated with closeInfo
(await wt.closed).closeInfo, and errors open streams
Network error
(await wt.closed) rejects, and errors open streams
received GOAWAY await wt.draining

13. Privacy and Security Considerations

This section is non-normative; it specifies no new behaviour, but instead summarizes information already present in other parts of the specification.

13.1. Confidentiality of Communications

The fact that communication is taking place cannot be hidden from adversaries that can observe the network, so this has to be regarded as public information.

All of the transport protocols described in this document use either TLS [RFC8446] or a semantically equivalent protocol, thus providing all of the security properties of TLS, including confidentiality and integrity of the traffic. WebTransport over HTTP uses the same certificate verification mechanism as outbound HTTP requests, thus relying on the same public key infrastructure for authentication of the remote server. In WebTransport, certificate verification errors are fatal; no interstitial allowing bypassing certificate validation is available.

13.2. State Persistence

WebTransport does not by itself create any new unique identifiers or new ways to persistently store state, nor does it automatically expose any of the existing persistent state to the server. For instance, neither [WEB-TRANSPORT-HTTP3] nor [WEB-TRANSPORT-HTTP2] send cookies or support HTTP authentication or caching invalidation mechanisms. Since they do use TLS, they inherit TLS persistent state such as TLS session tickets, which while not visible to passive network observers, could be used by the server to correlate different connections from the same client.

13.3. Protocol Security

WebTransport imposes a set of requirements as described in [WEB-TRANSPORT-OVERVIEW], including:

  1. Ensuring that the remote server is aware that the WebTransport protocol is in use and confirming that the remote server is willing to use the WebTransport protocol. [WEB-TRANSPORT-HTTP3] uses a combination of ALPN [RFC7301], an HTTP/3 setting, and a :protocol pseudo-header to identify the WebTransport protocol. [WEB-TRANSPORT-HTTP2] uses a combination of ALPN, an HTTP/2 setting, and a :protocol pseudo-header to identify the WebTransport protocol.

  2. Allowing the server to filter connections based on the origin of the resource originating the transport session. The Origin header field on the session establishment request carries this information.

Protocol security related considerations are described in the Security Considerations sections of [WEB-TRANSPORT-HTTP3] and [WEB-TRANSPORT-HTTP2].

Networking APIs can be commonly used to scan the local network for available hosts, and thus be used for fingerprinting and other forms of attacks. WebTransport follows the WebSocket approach to this problem: the specific connection error is not returned until an endpoint is verified to be a WebTransport endpoint; thus, the Web application cannot distinguish between a non-existing endpoint and the endpoint that is not willing to accept connections from the Web.

13.4. Authentication using Certificate Hashes {#certificate-hashes}

Normally, a user agent authenticates a TLS connection between itself and a remote endpoint by verifying the validity of the TLS server certificate provided against the server name in the URL [RFC9525]. This is accomplished by chaining server certificates to one of the trust anchors maintained by the user agent; the trust anchors in question are responsible for authenticating the server names in the certificates. We will refer to this system as Web PKI.

This API provides web applications with a capability to connect to a remote network endpoint authenticated by a specific server certificate, rather than its server name. This mechanism enables connections to endpoints for which getting long-term certificates can be challenging, including hosts that are ephemeral in nature (e.g. short-lived virtual machines), or that are not publicly routable. Since this mechanism substitutes Web PKI-based authentication for an individual connection, we need to compare the security properties of both.

A remote server will be able to successfully perform a TLS handshake only if it posesses the private key corresponding to the public key of the certificate specified. The API identifies the certificates using their hashes. That is only secure as long as the cryptographic hash function used has second-preimage resistance. The only function defined in this document is SHA-256; the API provides a way to introduce new hash functions through allowing multiple algorithm-hash pairs to be specified.

It is important to note that Web PKI provides additional security mechanisms in addition to simply establishing a chain of trust for a server name. One of them is handling certificate revocation. In cases where the certificate used is ephemeral, such a mechanism is not necessary. In other cases, the Web application has to consider the mechanism by which the certificate hashes are provisioned; for instance, if the hash is provided as a cached HTTP resource, the cache needs to be invalidated if the corresponding certificate has been rotated due to compromise. Another security feature provided by the Web PKI are safeguards against certain issues with key generation, such as rejecting certificates with known weak keys; while this specification does not provide any specific guidance, browsers MAY reject those as a part of implementation-defined behavior.

Web PKI enforces an expiry period requirement on the certificates. This requirement limits the scope of potential key compromise; it also forces server operators to design systems that support and actively perform key rotation. For this reason, WebTransport imposes a similar expiry requirement; as the certificates are expected to be ephemeral or short-lived, the expiry period is limited to two weeks. The two weeks limit is a balance between setting the expiry limit as low as possible to minimize consequences of a key compromise, and maintaining it sufficiently high to accomodate for clock skew across devices, and to lower the costs of synchronizing certificates between the client and the server side.

The WebTransport API lets the application specify multiple certificate hashes at once, allowing the client to accept multiple certificates for a period in which a new certificate is being rolled out.

Unlike a similar mechanism in WebRTC, the server certificate hash API in WebTransport does not provide any means of authenticating the client; the fact that the client knows what the server certificate is or how to contact it is not sufficient. The application has to establish the identity of the client in-band if necessary.

13.5. Fingerprinting and Tracking

This API provides sites with the ability to generate network activity and closely observe the effect of this activity. The information obtained in this way might be identifying.

It is important to recognize that very similar networking capabilities are provided by other web platform APIs (such as fetch and [webrtc]). The net adverse effect on privacy due to adding WebTransport is therefore minimal. The considerations in this section applies equally to other networking capabilities.

Measuring network characteristics requires that the network be used and that the effect of that usage be measured, both of which are enabled by this API. WebTransport provides sites with an ability to generate network activity toward a server of their choice and observe the effects. Observations of both the stable properties of a network path and dynamic effect of network usage are possible.

Information about the network is available to the server either directly through its own networking stack, indirectly through the rate at which data is consumed or transmitted by the client, or as part of the statistics provided by the API (see § 5.12 WebTransportConnectionStats Dictionary). Consequently, restrictions on information in user agents is not the only mechanism that might be needed to manage these privacy risks.

13.5.1. Static Observations

A site can observe available network capacity or round trip time (RTT) between a user agent and a chosen server. This information can be identifying when combined with other tracking vectors. RTT can also reveal something about the physical location of a user agent, especially if multiple measurements can be made from multiple vantage points.

Though networking is shared, network use is often sporadic, which means that sites are often able to observe the capacity and round trip times of an uncontested or lightly loaded network path. These properties are stable for many people as their network location does not change and the position of network bottlenecks--which determine available capacity--can be close to a user agent.

13.5.2. Shared Networking

Contested links present sites with opportunities to enable cross-site recognition, which might be used to perform unsanctioned tracking [UNSANCTIONED-TRACKING]. Network capacity is a finite shared resource, so a user agent that concurrently accesses different sites might reveal a connection between the identity presented to each site.

The use of networking capabilities on one site reduces the capacity available to other sites, which can be observed using networking APIs. Network usage and metrics can change dynamically, so any change can be observed in real time. This might allow sites to increase confidence that activity on different sites originates from the same user.

A user agent could limit or degrade access to feedback mechanisms such as statistics (§ 5.12 WebTransportConnectionStats Dictionary) for sites that are inactive or do not have focus (HTML § 6.6 Focus). As noted, this does not prevent a server from making observations about changes in the network.

13.5.3. Pooled Sessions

Similar to shared networking scenarios, when sessions are pooled on a single connection, information from one session is affected by the activity of another session. One session could infer information about the activity of another session, such as the rate at which another application is sending data.

The use of a shared connection already allows the server to correlate sessions. Use of a network partition key disables pooling where use of a shared session might enable unwanted cross-site recognition.

14. Examples

14.1. Sending a buffer of datagrams

This section is non-normative.

Sending a buffer of datagrams can be achieved by using the datagrams' writable attribute. In the following example datagrams are only sent if the transport is ready to send.

async function sendDatagrams(url, datagrams) {
  const wt = new WebTransport(url);
  const writer = wt.datagrams.writable.getWriter();
  for (const bytes of datagrams) {
    await writer.ready;
    writer.write(bytes).catch(() => {});
  }
}

14.2. Sending datagrams at a fixed rate

This section is non-normative.

Sending datagrams at a fixed rate regardless if the transport is ready to send can be achieved by simply using datagrams' writable and not using the ready attribute. More complex scenarios can utilize the ready attribute.

// Sends datagrams every 100 ms.
async function sendFixedRate(url, createDatagram, ms = 100) {
  const wt = new WebTransport(url);
  await wt.ready;
  const writer = wt.datagrams.writable.getWriter();
  const bytes = createDatagram();
  setInterval(() => writer.write(bytes).catch(() => {}), ms);
}

14.3. Receiving datagrams

This section is non-normative.

Datagrams can be received by reading from the transport.datagrams.readable attribute. Null values may indicate that packets are not being processed quickly enough.

async function receiveDatagrams(url) {
  const wt = new WebTransport(url);
  for await (const datagram of wt.datagrams.readable) {
    // Process the datagram
  }
}

14.4. Receiving datagrams with a BYOB reader

This section is non-normative.

As datagrams are readable byte streams, you can acquire a BYOB reader for them, which allows more precise control over buffer allocation in order to avoid copies. This example reads the datagram into a 64kB memory buffer.

const wt = new WebTransport(url);

for await (const datagram of wt.datagrams.readable) {
  const reader = datagram.getReader({ mode: "byob" });
  
  let array_buffer = new ArrayBuffer(65536);
  const buffer = await readInto(array_buffer);
}

async function readInto(buffer) {
  let offset = 0;

  while (offset < buffer.byteLength) {
    const {value: view, done} = await reader.read(
        new Uint8Array(buffer, offset, buffer.byteLength - offset));
    buffer = view.buffer;
    if (done) {
      break;
    }
    offset += view.byteLength;
  }

  return buffer;
}

14.5. Sending a stream

This section is non-normative.

Sending data as a one-way stream can be achieved by using the createUnidirectionalStream function and the resulting stream’s writer.

async function sendData(url, ...data) {
  const wt = new WebTransport(url);
  const writable = await wt.createUnidirectionalStream();
  const writer = writable.getWriter();
  for (const bytes of data) {
    await writer.ready;
    writer.write(bytes).catch(() => {});
  }
  await writer.close();
}

The streams spec discourages awaiting the promise from write().

Encoding can also be done through pipes from a ReadableStream, for example using TextEncoderStream.

async function sendText(url, readableStreamOfTextData) {
  const wt = new WebTransport(url);
  const writable = await wt.createUnidirectionalStream();
  await readableStreamOfTextData
    .pipeThrough(new TextEncoderStream("utf-8"))
    .pipeTo(writable);
}

14.6. Receiving incoming streams

This section is non-normative.

Reading incoming streams can be achieved by iterating over the incomingUnidirectionalStreams attribute, and then consuming each WebTransportReceiveStream by iterating over its chunks.

async function receiveData(url, processTheData) {
  const wt = new WebTransport(url);
  for await (const readable of wt.incomingUnidirectionalStreams) {
    // consume streams individually using IFFEs, reporting per-stream errors
    ((async () => {
      try {
        for await (const bytes of readable) {
          processTheData(bytes);
        }
      } catch (e) {
        console.error(e);
      }
    })());
  }
}

Decoding can also be done through pipes to new WritableStreams, for example using TextDecoderStream. This example assumes text output should not be interleaved, and therefore only reads one stream at a time.

async function receiveText(url, createWritableStreamForTextData) {
  const wt = new WebTransport(url);
  for await (const readable of wt.incomingUnidirectionalStreams) {
    // consume sequentially to not interleave output, reporting per-stream errors
    try {
      await readable
       .pipeThrough(new TextDecoderStream("utf-8"))
       .pipeTo(createWritableStreamForTextData());
    } catch (e) {
      console.error(e);
    }
  }
}

14.7. Receiving a stream with a BYOB reader

This section is non-normative.

As WebTransportReceiveStreams are readable byte streams, you can acquire a BYOB reader for them, which allows more precise control over buffer allocation in order to avoid copies. This example reads the first 1024 bytes from a WebTransportReceiveStream into a single memory buffer.

const wt = new WebTransport(url);

const reader = wt.incomingUnidirectionalStreams.getReader();
const { value: recv_stream, done } = await reader.read();
const byob_reader = recv_stream.getReader({ mode: "byob" });

let array_buffer = new ArrayBuffer(1024);
const buffer = await readInto(array_buffer);

async function readInto(buffer) {
  let offset = 0;

  while (offset < buffer.byteLength) {
    const {value: view, done} = await reader.read(
        new Uint8Array(buffer, offset, buffer.byteLength - offset));
    buffer = view.buffer;
    if (done) {
      break;
    }
    offset += view.byteLength;
  }

  return buffer;
}

14.8. Sending a transactional chunk on a stream

This section is non-normative.

Sending a transactional piece of data on a unidirectional stream, only if it can be done entirely without blocking on flow control, can be achieved by using the getWriter function and the resulting writer.

async function sendTransactionalData(wt, bytes) {
  const writable = await wt.createUnidirectionalStream();
  const writer = writable.getWriter();
  await writer.ready;
  try {
    await writer.atomicWrite(bytes);
  } catch (e) {
    if (e.name != "AbortError") throw e;
    // rejected to avoid blocking on flow control
    // The writable remains un-errored provided no non-atomic writes are pending
  } finally {
    writer.releaseLock();
  }
}

14.9. Using a server certificate hash

This section is non-normative.

A WebTransport session can override the default trust evaluation performed by the client with a check against the hash of the certificate provided to the server. In the example below, hashValue is a BufferSource containing the SHA-256 hash of a server certificate that the underlying connection should consider to be valid.

const wt = new WebTransport(url, {
  serverCertificateHashes: [
    {
      algorithm: "sha-256",
      value: hashValue,
    }
  ]
});
await wt.ready;

14.10. Complete example

This section is non-normative.

This example illustrates use of the closed and ready promises, opening of uni-directional and bi-directional streams by either the client or the server, and sending and receiving datagrams.

// Adds an entry to the event log on the page, optionally applying a specified
// CSS class.

let wt, streamNumber, datagramWriter;

connect.onclick = async () => {
  try {
    const url = document.getElementById('url').value;

    wt = new WebTransport(url);
    addToEventLog('Initiating connection...');
    await wt.ready;
    addToEventLog(`${(wt.reliability == "reliable-only")? "TCP" : "UDP"} ` +
                  `connection ready.`);
    wt.closed
      .then(() => addToEventLog('Connection closed normally.'))
      .catch(() => addToEventLog('Connection closed abruptly.', 'error'));

    streamNumber = 1;
    datagramWriter = wt.datagrams.writable.getWriter();

    readDatagrams();
    acceptUnidirectionalStreams();
    document.forms.sending.elements.send.disabled = false;
    document.getElementById('connect').disabled = true;
  } catch (e) {
    addToEventLog(`Connection failed. ${e}`, 'error');
  }
}

sendData.onclick = async () => {
  const form = document.forms.sending.elements;
  const data = sending.data.value;
  const bytes = new TextEncoder('utf-8').encode(data);
  try {
    switch (form.sendtype.value) {
      case 'datagram': {
        await datagramWriter.ready;
        datagramWriter.write(bytes).catch(() => {});
        addToEventLog(`Sent datagram: ${data}`);
        break;
      }
      case 'unidi': {
        const writable = await wt.createUnidirectionalStream();
        const writer = writable.getWriter();
        writer.write(bytes).catch(() => {});
        await writer.close();
        addToEventLog(`Sent a unidirectional stream with data: ${data}`);
        break;
      }
      case 'bidi': {
        const duplexStream = await wt.createBidirectionalStream();
        const n = streamNumber++;
        readFromIncomingStream(duplexStream.readable, n);

        const writer = duplexStream.writable.getWriter();
        writer.write(bytes).catch(() => {});
        await writer.close();
        addToEventLog(`Sent bidirectional stream #${n} with data: ${data}`);
        break;
      }
    }
  } catch (e) {
    addToEventLog(`Error while sending data: ${e}`, 'error');
  }
}

// Reads datagrams into the event log until EOF is reached.
async function readDatagrams() {
  try {
    const decoder = new TextDecoderStream('utf-8');

    for await (const data of wt.datagrams.readable.pipeThrough(decoder)) {
      addToEventLog(`Datagram received: ${data}`);
    }
    addToEventLog('Done reading datagrams!');
  } catch (e) {
    addToEventLog(`Error while reading datagrams: ${e}`, 'error');
  }
}

async function acceptUnidirectionalStreams() {
  try {
    for await (const readable of wt.incomingUnidirectionalStreams) {
      const number = streamNumber++;
      addToEventLog(`New incoming unidirectional stream #${number}`);
      readFromIncomingStream(readable, number);
    }
    addToEventLog('Done accepting unidirectional streams!');
  } catch (e) {
    addToEventLog(`Error while accepting streams ${e}`, 'error');
  }
}

async function readFromIncomingStream(readable, number) {
  try {
    const decoder = new TextDecoderStream('utf-8');
    for await (const data of readable.pipeThrough(decoder)) {
      addToEventLog(`Received data on stream #${number}: ${data}`);
    }
    addToEventLog(`Stream #${number} closed`);
  } catch (e) {
    addToEventLog(`Error while reading from stream #${number}: ${e}`, 'error');
    addToEventLog(`    ${e.message}`);
  }
}

function addToEventLog(text, severity = 'info') {
  const log = document.getElementById('event-log');
  const previous = log.lastElementChild;
  const entry = document.createElement('li');
  entry.innerText = text;
  entry.className = `log-${severity}`;
  log.appendChild(entry);

  // If the previous entry in the log was visible, scroll to the new element.
  if (previous &&
      previous.getBoundingClientRect().top < log.getBoundingClientRect().bottom) {
    entry.scrollIntoView();
  }
}

15. Acknowledgements

The editors wish to thank the Working Group chairs and Team Contact, Jan-Ivar Bruaroey, Will Law and Yves Lafon, for their support.

The WebTransport interface is based on the QuicTransport interface initially described in the W3C ORTC CG, and has been adapted for use in this specification.

Index

Terms defined by this specification

Terms defined by reference

References

Normative References

[CSP3]
Mike West; Antonio Sartori. Content Security Policy Level 3. URL: https://w3c.github.io/webappsec-csp/
[DOM]
Anne van Kesteren. DOM Standard. Living Standard. URL: https://dom.spec.whatwg.org/
[ECMASCRIPT-6.0]
Allen Wirfs-Brock. ECMA-262 6th Edition, The ECMAScript 2015 Language Specification. URL: http://www.ecma-international.org/ecma-262/6.0/index.html
[ENCODING]
Anne van Kesteren. Encoding Standard. Living Standard. URL: https://encoding.spec.whatwg.org/
[FETCH]
Anne van Kesteren. Fetch Standard. Living Standard. URL: https://fetch.spec.whatwg.org/
[HR-TIME-3]
Yoav Weiss. High Resolution Time. URL: https://w3c.github.io/hr-time/
[HTML]
Anne van Kesteren; et al. HTML Standard. Living Standard. URL: https://html.spec.whatwg.org/multipage/
[INFRA]
Anne van Kesteren; Domenic Denicola. Infra Standard. Living Standard. URL: https://infra.spec.whatwg.org/
[PRIVACY-PRINCIPLES]
Robin Berjon; Jeffrey Yasskin. Privacy Principles. URL: https://w3ctag.github.io/privacy-principles/
[QUIC]
Jana Iyengar; Martin Thomson. QUIC: A UDP-Based Multiplexed and Secure Transport. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc9000
[QUIC-DATAGRAM]
Tommy Pauly; Eric Kinnear; David Schinazi. An Unreliable Datagram Extension to QUIC. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc9221
[RFC2119]
S. Bradner. Key words for use in RFCs to Indicate Requirement Levels. March 1997. Best Current Practice. URL: https://datatracker.ietf.org/doc/html/rfc2119
[RFC3279]
L. Bassham; W. Polk; R. Housley. Algorithms and Identifiers for the Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile. April 2002. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc3279
[RFC5280]
D. Cooper; et al. Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile. May 2008. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc5280
[RFC8174]
B. Leiba. Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words. May 2017. Best Current Practice. URL: https://www.rfc-editor.org/rfc/rfc8174
[RFC8422]
Y. Nir; S. Josefsson; M. Pegourie-Gonnard. Elliptic Curve Cryptography (ECC) Cipher Suites for Transport Layer Security (TLS) Versions 1.2 and Earlier. August 2018. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc8422
[RFC9002]
J. Iyengar, Ed.; I. Swett, Ed.. QUIC Loss Detection and Congestion Control. May 2021. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc9002
[RFC9525]
P. Saint-Andre; R. Salz. Service Identity in TLS. November 2023. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc9525
[STREAMS]
Adam Rice; et al. Streams Standard. Living Standard. URL: https://streams.spec.whatwg.org/
[URL]
Anne van Kesteren. URL Standard. Living Standard. URL: https://url.spec.whatwg.org/
[WEB-TRANSPORT-HTTP2]
Alan Frindell; et al. WebTransport over HTTP/2. Internet-Draft. URL: https://datatracker.ietf.org/doc/html/draft-ietf-webtrans-http2/
[WEB-TRANSPORT-HTTP3]
Alan Frindell; Eric Kinnear; Victor Vasiliev. WebTransport over HTTP/3. Internet-Draft. URL: https://datatracker.ietf.org/doc/html/draft-ietf-webtrans-http3/
[WEB-TRANSPORT-OVERVIEW]
Victor Vasiliev. WebTransport Protocol Framework. Internet-Draft. URL: https://datatracker.ietf.org/doc/html/draft-ietf-webtrans-overview
[WEBIDL]
Edgar Chen; Timothy Gu. Web IDL Standard. Living Standard. URL: https://webidl.spec.whatwg.org/

Informative References

[RFC7301]
S. Friedl; et al. Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension. July 2014. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc7301
[RFC8446]
E. Rescorla. The Transport Layer Security (TLS) Protocol Version 1.3. August 2018. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc8446
[RFC9308]
M. Kühlewind; B. Trammell. Applicability of the QUIC Transport Protocol. September 2022. Informational. URL: https://www.rfc-editor.org/rfc/rfc9308
[UNSANCTIONED-TRACKING]
Mark Nottingham. Unsanctioned Web Tracking. 17 July 2015. TAG Finding. URL: http://www.w3.org/2001/tag/doc/unsanctioned-tracking/
[WEBRTC]
Cullen Jennings; et al. WebRTC: Real-Time Communication in Browsers. URL: https://w3c.github.io/webrtc-pc/

IDL Index

[Exposed=(Window,Worker), SecureContext]
interface WebTransportDatagramDuplexStream {
  readonly attribute ReadableStream readable;
  readonly attribute WritableStream writable;

  readonly attribute unsigned long maxDatagramSize;
  attribute unrestricted double? incomingMaxAge;
  attribute unrestricted double? outgoingMaxAge;
  attribute unrestricted double incomingHighWaterMark;
  attribute unrestricted double outgoingHighWaterMark;
};

[Exposed=(Window,Worker), SecureContext]
interface WebTransport {
  constructor(USVString url, optional WebTransportOptions options = {});

  Promise<WebTransportConnectionStats> getStats();
  readonly attribute Promise<undefined> ready;
  readonly attribute WebTransportReliabilityMode reliability;
  readonly attribute WebTransportCongestionControl congestionControl;
  [EnforceRange] attribute unsigned short? anticipatedConcurrentIncomingUnidirectionalStreams;
  [EnforceRange] attribute unsigned short? anticipatedConcurrentIncomingBidirectionalStreams;

  readonly attribute Promise<WebTransportCloseInfo> closed;
  readonly attribute Promise<undefined> draining;
  undefined close(optional WebTransportCloseInfo closeInfo = {});

  readonly attribute WebTransportDatagramDuplexStream datagrams;

  Promise<WebTransportBidirectionalStream> createBidirectionalStream(
      optional WebTransportSendStreamOptions options = {});
  /* a ReadableStream of WebTransportBidirectionalStream objects */
  readonly attribute ReadableStream incomingBidirectionalStreams;

  Promise<WebTransportSendStream> createUnidirectionalStream(
      optional WebTransportSendStreamOptions options = {});
  /* a ReadableStream of WebTransportReceiveStream objects */
  readonly attribute ReadableStream incomingUnidirectionalStreams;
  WebTransportSendGroup createSendGroup();

  static readonly attribute boolean supportsReliableOnly;
};

enum WebTransportReliabilityMode {
  "pending",
  "reliable-only",
  "supports-unreliable",
};

dictionary WebTransportHash {
  DOMString algorithm;
  BufferSource value;
};

dictionary WebTransportOptions {
  boolean allowPooling = false;
  boolean requireUnreliable = false;
  sequence<WebTransportHash> serverCertificateHashes;
  WebTransportCongestionControl congestionControl = "default";
  [EnforceRange] unsigned short? anticipatedConcurrentIncomingUnidirectionalStreams = null;
  [EnforceRange] unsigned short? anticipatedConcurrentIncomingBidirectionalStreams = null;
};

enum WebTransportCongestionControl {
  "default",
  "throughput",
  "low-latency",
};

dictionary WebTransportCloseInfo {
  unsigned long closeCode = 0;
  USVString reason = "";
};

dictionary WebTransportSendStreamOptions {
  WebTransportSendGroup? sendGroup = null;
  long long sendOrder = 0;
  boolean waitUntilAvailable = false;
};

dictionary WebTransportConnectionStats {
  unsigned long long bytesSent;
  unsigned long long packetsSent;
  unsigned long long bytesLost;
  unsigned long long packetsLost;
  unsigned long long bytesReceived;
  unsigned long long packetsReceived;
  DOMHighResTimeStamp smoothedRtt;
  DOMHighResTimeStamp rttVariation;
  DOMHighResTimeStamp minRtt;
  WebTransportDatagramStats datagrams;
  unsigned long long? estimatedSendRate;
};

dictionary WebTransportDatagramStats {
  unsigned long long droppedIncoming;
  unsigned long long expiredIncoming;
  unsigned long long expiredOutgoing;
  unsigned long long lostOutgoing;
};

[Exposed=(Window,Worker), SecureContext, Transferable]
interface WebTransportSendStream : WritableStream {
  attribute WebTransportSendGroup? sendGroup;
  attribute long long sendOrder;
  Promise<WebTransportSendStreamStats> getStats();
  WebTransportWriter getWriter();
};

dictionary WebTransportSendStreamStats {
  unsigned long long bytesWritten;
  unsigned long long bytesSent;
  unsigned long long bytesAcknowledged;
};

[Exposed=(Window,Worker), SecureContext]
interface WebTransportSendGroup {
  Promise<WebTransportSendStreamStats> getStats();
};

[Exposed=(Window,Worker), SecureContext, Transferable]
interface WebTransportReceiveStream : ReadableStream {
  Promise<WebTransportReceiveStreamStats> getStats();
};

dictionary WebTransportReceiveStreamStats {
  unsigned long long bytesReceived;
  unsigned long long bytesRead;
};

[Exposed=(Window,Worker), SecureContext]
interface WebTransportBidirectionalStream {
  readonly attribute WebTransportReceiveStream readable;
  readonly attribute WebTransportSendStream writable;
};

[Exposed=*, SecureContext]
interface WebTransportWriter : WritableStreamDefaultWriter {
  Promise<undefined> atomicWrite(optional any chunk);
};

[Exposed=(Window,Worker), Serializable, SecureContext]
interface WebTransportError : DOMException {
  constructor(optional DOMString message = "", optional WebTransportErrorOptions options = {});

  readonly attribute WebTransportErrorSource source;
  readonly attribute unsigned long? streamErrorCode;
};

dictionary WebTransportErrorOptions {
  WebTransportErrorSource source = "stream";
  [Clamp] unsigned long? streamErrorCode = null;
};

enum WebTransportErrorSource {
  "stream",
  "session",
};

Issues Index

This needs to be done in workers too. See #127 and whatwg/html#6731.

This configuration option is considered a feature at risk due to the lack of implementation in browsers of a congestion control algorithm, at the time of writing, that optimizes for low latency.