WebTransport

Editor’s Draft,

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https://w3c.github.io/webtransport/
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Editors:
Bernard Aboba (Microsoft Corporation)
Victor Vasiliev (Google)
Yutaka Hirano (Google)
Former Editors:
Peter Thatcher (Google)
Robin Raymond (Optical Tone Ltd.)

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]. This specification is being developed in conjunction with a protocol specification 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 2 November 2021 W3C Process Document.

1. Introduction

This section is non-normative.

This specification uses [WEB-TRANSPORT-HTTP3] 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] specification is 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 and SHOULD are to be interpreted as described in [RFC2119].

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

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

WebTransport session has the following capabilities defined in [WEB-TRANSPORT-HTTP3].

capability definition
send a datagram [WEB-TRANSPORT-HTTP3] section 4.4
receive a datagram [WEB-TRANSPORT-HTTP3] section 4.4
create an outgoing unidirectional stream [WEB-TRANSPORT-HTTP3] section 4.1
create a bidirectional stream [WEB-TRANSPORT-HTTP3] section 4.2
receive an incoming unidirectional stream [WEB-TRANSPORT-HTTP3] section 4.1
receive a bidirectional stream [WEB-TRANSPORT-HTTP3] section 4.2

To establish a WebTransport session with an origin origin, follow [WEB-TRANSPORT-HTTP3] section 3.3, 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.

To terminate a WebTransport session session with an optional integer code and an optional byte sequence reason, follow [WEB-TRANSPORT-HTTP3] section 5.

A WebTransport session session is terminated, with optionally an integer code and a byte sequence reason, when the HTTP/3 stream associated with the CONNECT request that initiated session is closed by the server, as described at [WEB-TRANSPORT-HTTP3] section 5.

WebTransport stream is a concept for HTTP/3 stream on a WebTransport session.

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

WebTransport stream has the following capabilities:

capability definition incoming unidirectional outgoing unidirectional bidirectional
send bytes (potentially with FIN) [QUIC] section 2.2 No Yes Yes
receive bytes (potentially with FIN) [QUIC] section 2.2 Yes No Yes
send STOP_SENDING [QUIC] section 3.5 Yes No Yes
reset a WebTransport stream [QUIC] section 19.4 No Yes Yes

WebTransport stream has the following signals:

event definition incoming unidirectional outgoing unidirectional bidirectional
STOP_SENDING [QUIC] section 3.5 No Yes Yes
RESET_STREAM [QUIC] section 19.4 Yes No 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 double? incomingMaxAge;
  attribute double? outgoingMaxAge;
  attribute long incomingHighWaterMark;
  attribute long 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 integer representing the high water mark of the incoming datagrams.
[[IncomingDatagramsExpirationDuration]] A double value 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 integer representing the high water mark of the outgoing datagrams.
[[OutgoingDatagramsExpirationDuration]] A 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 integer

    [[IncomingDatagramsExpirationDuration]]

    null

    [[OutgoingDatagramsQueue]]

    an empty queue

    [[OutgoingDatagramsHighWaterMark]]

    an implementation-defined integer

    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 double, nullable

The getter steps are:

  1. Return this.[[IncomingDatagramsExpirationDuration]].

The setter steps are:

  1. Let value be the given value.

  2. If value is null or value > 0:

    1. 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 double, nullable

The getter steps are:

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

The setter steps are:

  1. Let value be the given value.

  2. If value is null or value > 0:

    1. Set this's [[OutgoingDatagramsExpirationDuration]] to value.

incomingHighWaterMark, of type long

The getter steps are:

  1. Return this.[[IncomingDatagramsHighWaterMark]].

The setter steps are:

  1. Let value be the given value.

  2. If value ≥ 0:

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

outgoingHighWaterMark, of type long

The getter steps are:

  1. Return this.[[OutgoingDatagramsHighWaterMark]].

The setter steps are:

  1. Let value be the given value.

  2. If value ≥ 0:

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

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

  7. Enqueue chunk to transport.[[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 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 [[OutgoingDatagramsExpirationDuration]], and they can be discarded in the same manner as the "connected" case. Once the transport’s [[State]] becomes "connected", it will start sending stored datagrams.

5. WebTransport Interface

WebTransport provides an API to the HTTP/3 transport functionality defined in [WEB-TRANSPORT-HTTP3].

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

  Promise<WebTransportStats> getStats();
  readonly attribute Promise<undefined> ready;
  readonly attribute WebTransportReliabilityMode reliability;
  readonly attribute Promise<WebTransportCloseInfo> closed;
  undefined close(optional WebTransportCloseInfo closeInfo = {});

  readonly attribute WebTransportDatagramDuplexStream datagrams;

  Promise<WebTransportBidirectionalStream> createBidirectionalStream();
  /* a ReadableStream of WebTransportBidirectionalStream objects */
  readonly attribute ReadableStream incomingBidirectionalStreams;

  Promise<WebTransportSendStream> createUnidirectionalStream();
  /* a ReadableStream of WebTransportReceiveStream objects */
  readonly attribute ReadableStream incomingUnidirectionalStreams;
};

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", "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 unreliable (UDP) transport is supported or whether only reliable (TCP fallback) transport is used. Returns "pending" until a connection has been established.
[[Closed]] A promise fulfilled when the associated WebTransport object is closed gracefully, or rejected when it is closed abruptly or failed on initialization.
[[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 if it exists, and false otherwise.

  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 TypeError.

  9. Let requireUnreliable be options's requireUnreliable.

  10. Let incomingDatagrams be a new ReadableStream.

  11. Let outgoingDatagrams be a new WritableStream.

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

  13. 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"

    [[Closed]]

    a new promise

    [[Datagrams]]

    datagrams

    [[Session]]

    null

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

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

  16. Set up incomingDatagrams with pullAlgorithm set to pullDatagramsAlgorithm, and highWaterMark set to 0.

  17. Set up outgoingDatagrams with writeAlgorithm set to writeDatagramsAlgorithm.

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

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

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

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

  22. Initialize WebTransport over HTTP with transport, parsedURL, dedicated, and requireUnreliable.

  23. Return transport.

To initialize WebTransport over HTTP, given a WebTransport object transport, a URL record url, a boolean dedicated, and a boolean http3Only, 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 the result of creating a WebTransportError with "session".

    3. Cleanup transport with error, error and true.

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

  7. Run the remaining steps in parallel, but abort them whenever transport.[[State]] becomes "closed" or "failed".

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

  9. Let connection be the result of obtaining a connection with networkPartitionKey, url, false, newConnection, and http3Only.

  10. 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 the result of creating a WebTransportError with "session".

    3. Cleanup transport with error.

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

  12. 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 the result of creating a WebTransportError with "session".

    3. Cleanup transport with error.

  13. Establish a WebTransport session with origin on connection.

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

  14. 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 the result of creating a WebTransportError with "session".

    3. Cleanup transport with error.

  15. Let session be the established WebTransport session.

  16. Assert: maxDatagramSize is an integer.

  17. 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. Set transport’s [[Reliability]] to "supports-unreliable".

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

  3. Let p be a new promise.

  4. Return p and run the remaining steps in parallel.

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

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

  7. 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.

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

  3. Let p be a new promise.

  4. Return p and run the remaining steps in parallel.

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

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

  7. 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.

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]].

This promise MUST be resolved when the transport is closed; an implementation SHOULD include error information in the reason and closeCode fields of WebTransportCloseInfo.

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]].

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 the result of creating a WebTransportError with "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 HTTP/3 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. Return p and continue the following steps in parallel.

    1. Gather the stats from the underlying QUIC connection, including stats on datagrams.

    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 WebTransportStats object representing the gathered stats.

      2. Resolve p with stats.

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 p be a new promise.

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

    1. Wait for transport.[[State]] to be "connected".

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

      Note: This operation may take time, for example when the stream ID is exhausted. [QUIC]

    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 and transport.

      3. Resolve p with stream.

  5. 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 p be a new promise.

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

    1. Wait for transport.[[State]] to be "connected".

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

      Note: This operation may take time, for example when the stream ID is exhausted. [QUIC]

    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 and transport.

      3. Resolve p with stream.

  5. return p.

5.5. Procedures

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

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

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

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

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

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

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

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

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

  10. For each sendStream in sendStreams, error sendStream with error.

  11. For each receiveStream in receiveStreams, error receiveStream 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.

  12. If closeInfo is given, then:

    1. Resolve closed with closeInfo.

    2. Assert: ready is settled.

    3. Close incomingBidirectionalStreams.

    4. Close incomingUnidirectionalStreams.

  13. Otherwise:

    1. Reject closed with error.

    2. Reject ready with error.

    3. Error incomingBidirectionalStreams with error.

    4. Error incomingUnidirectionalStreams 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. Let cleanly be a boolean representing whether the HTTP/3 stream associated with the CONNECT request that initiated transport.[[Session]] is in the "Data Recvd" state. [QUIC]

  2. 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 the result of creating a WebTransportError with "session".

    3. If cleanly is false, then cleanup transport with error, and abort these steps.

    4. Let closeInfo be a new WebTransportCloseInfo.

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

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

    7. Cleanup transport with error and closeInfo.

Whenever a connection associated with a WebTransport transport 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 the result of creating a WebTransportError with "session".

    3. Cleanup transport with error, error and true.

5.7. Context cleanup steps

This specification defines unloading document cleanup steps as the following steps, given a Document document:

  1. Let window be document’s relevant global object.

  2. For each WebTransport transport whose relevant global object is window:

    1. If transport.[[State]] is "connected", set transport.[[State]] to "failed" and terminate transport.[[Session]] in parallel.

    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

The user agent MUST NOT garbage collect a WebTransport object whose [[State]] is either "connecting" or "connected".

5.9. Configuration

dictionary WebTransportHash {
  DOMString algorithm;
  BufferSource value;
};

dictionary WebTransportOptions {
  boolean allowPooling = false;
  boolean requireUnreliable = false;
  sequence<WebTransportHash> serverCertificateHashes;
};

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

allowPooling, of type boolean, defaulting to false

When set to true, the WebTransport connection can be pooled, that is, the network connection for the WebTransport session can be shared with other HTTP/3 sessions.

requireUnreliable, of type boolean, defaulting to false

When set to true, the WebTransport connection cannot be established over HTTP/2 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.

To compute a certificate hash, do the following:
  1. Let cert be the input 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, do the following:
  1. Let hashes be the input array of hashes.

  2. Let referenceHash be the computed hash of the input certificate.

  3. For every hash hash in hashes:

    1. If hash.value is not null:

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

      2. If algorithm of hash is equal to "sha-256", and hashValue is equal to referenceHash, the certificate is valid. Return true.

  4. 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;
  DOMString 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 DOMString, defaulting to ""

The reason for closing the WebTransport.

5.11. WebTransportStats Dictionary

The WebTransportStats dictionary includes information on HTTP/3 connection stats.

Now that quic-transport has been removed, this section needs to be revised. Some of those are safe to expose for HTTP/2 and HTTP/3 connections (like min-RTT), while most would either result in information disclosure or are impossible to define for pooled connections.

dictionary WebTransportStats {
  DOMHighResTimeStamp timestamp;
  unsigned long long bytesSent;
  unsigned long long packetsSent;
  unsigned long long packetsLost;
  unsigned long numOutgoingStreamsCreated;
  unsigned long numIncomingStreamsCreated;
  unsigned long long bytesReceived;
  unsigned long long packetsReceived;
  DOMHighResTimeStamp smoothedRtt;
  DOMHighResTimeStamp rttVariation;
  DOMHighResTimeStamp minRtt;
  WebTransportDatagramStats datagrams;
};

The dictionary SHALL have the following attributes:

timestamp, of type DOMHighResTimeStamp

The timestamp for when the stats are gathered, relative to the UNIX epoch (Jan 1, 1970, UTC).

bytesSent, of type unsigned long long

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

packetsLost, of type unsigned long long

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

numOutgoingStreamsCreated, of type unsigned long

The number of outgoing QUIC streams created on the QUIC connection.

numIncomingStreamsCreated, of type unsigned long

The number of incoming QUIC streams created on the QUIC connection.

bytesReceived, of type unsigned long long

The number of total bytes received on the QUIC 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 QUIC 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.

5.12. WebTransportDatagramStats Dictionary

The WebTransportDatagramStats dictionary includes statistics on datagram transmission over the HTTP/3 connection.

dictionary WebTransportDatagramStats {
  DOMHighResTimeStamp timestamp;
  unsigned long long expiredOutgoing;
  unsigned long long droppedIncoming;
  unsigned long long lostOutgoing;
};

The dictionary SHALL have the following attributes:

timestamp, of type DOMHighResTimeStamp

The timestamp for when the stats are gathered, relative to the UNIX epoch (Jan 1, 1970, UTC).

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.

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.

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]
interface WebTransportSendStream : WritableStream {
  Promise<WebTransportSendStreamStats> getStats();
};

A WebTransportSendStream is always created by the create procedure.

6.1. 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. Return p and continue 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.

6.2. 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.

6.3. Procedures

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

  1. Let stream be a new WebTransportSendStream, with:

    [[InternalStream]]

    internalStream

    [[PendingOperation]]

    null

    [[Transport]]

    transport

  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. Add the following steps to stream’s [[controller]]'s [[signal]].

    1. If stream.[[PendingOperation]] is null, then abort these steps.

    2. Let reason be stream’s [[controller]]'s [[signal]]'s abort reason.

    3. Let abortPromise be the result of aborting stream with reason.

    4. Upon fulfillment of abortPromise, reject promise with reason.

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

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

    7. Resolve pendingOperation with promise.

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

  8. 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. Return promise and run the remaining steps in parallel.

  7. 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.

    Note: Solving prioritization is an outstanding issue.

  8. 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 and reject the result of this write operation.

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

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

    2. Resolve promise with undefined.

Note: 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 WebTransportSendStream. This also means the fulfillment of the promise returned from this algorithm (or, WritableStreamDefaultWriter.write) 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, use an application-level protocol.

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. Return promise and run the remaining steps in parallel.

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

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

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

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

    2. Resolve promise with undefined.

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 reasons.[[StreamErrorCode]] is not null, then set code to reason.[[StreamErrorCode]].

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

  7. If code > 255, then set code to 255.

    Note: Valid values of code are from 0 to 255 inclusive. The code will be encoded to a number in [0x52e4a40fa8db, 0x52e4a40fa9e2] as decribed in [WEB-TRANSPORT-HTTP3].

  8. Return promise and run the remaining steps in parallel.

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

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

6.4. 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 255 inclusive. The code has been decoded from a number in [0x52e4a40fa8db, 0x52e4a40fa9e2] 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 the result of creating a WebTransportError with "stream".

    4. Set error.[[StreamErrorCode]] to code.

    5. Error stream with error.

6.5. WebTransportSendStreamStats Dictionary

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

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

The dictionary SHALL have the following attributes:

timestamp, of type DOMHighResTimeStamp

The timestamp for when the stats are gathered, relative to the UNIX epoch (Jan 1, 1970, UTC).

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 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.

A WebTransportReceiveStream is always created by the create procedure.

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

A WebTransportReceiveStream is always created by the create procedure.

7.1. 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. Return p and continue the following steps in parallel.

    1. Gather the stats specific to this WebTransportReceiveStream.

    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 WebTransportReceiveStreamStats object representing the gathered stats.

      2. Resolve p with stats.

7.2. 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.

7.3. 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. Return promise and run the remaining steps in parallel.

  8. 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. Note: 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.

  9. 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 and reject the result of this read operation.

  10. 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.

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 reasons.[[StreamErrorCode]] is not null, then set code to reason.[[StreamErrorCode]].

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

  7. If code > 255, then set code to 255.

    Note: Valid values of code are from 0 to 255 inclusive. The code will be encoded to a number in [0x52e4a40fa8db, 0x52e4a40fa9e2] as decribed in [WEB-TRANSPORT-HTTP3].

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

  9. Return promise and run the remaining steps in parallel.

  10. Send STOP_SENDING with internalStream and code.

  11. 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.

7.4. 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 255 inclusive. The code has been decoded from a number in [0x52e4a40fa8db, 0x52e4a40fa9e2] 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 the result of creating a WebTransportError with "stream".

    4. Set error.[[StreamErrorCode]] to code.

    5. Error stream with error.

7.5. WebTransportReceiveStreamStats Dictionary

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

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

The dictionary SHALL have the following attributes:

timestamp, of type DOMHighResTimeStamp

The timestamp for when the stats are gathered, relative to the UNIX epoch (Jan 1, 1970, UTC).

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.

8. Interface WebTransportBidirectionalStream

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

8.1. 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.

8.2. Attributes

readable, of type ReadableStream, readonly

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

writable, of type WritableStream, readonly

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

8.3. Procedures

To create a WebTransportBidirectionalStream with a bidirectional WebTransport stream internalStream and a WebTransport object transport, 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 and transport.

  3. Let stream be a new WebTransportBidirectionalStream, with:

    [[Readable]]

    readable

    [[Writable]]

    writable

    [[Transport]]

    transport

  4. Return stream.

9. WebTransportError Interface

WebTransportError is a subclass of DOMException that represents

[Exposed=(Window,Worker), SecureContext]
interface WebTransportError : DOMException {
  constructor(optional WebTransportErrorInit init = {});

  readonly attribute WebTransportErrorSource source;
  readonly attribute octet? streamErrorCode;
};

dictionary WebTransportErrorInit {
  [Clamp] octet streamErrorCode;
  DOMString message;
};

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

9.1. 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.

9.2. Constructor

The new WebTransportError(init) constructor steps are:

  1. Let error be this.

  2. Let message be init.message if it exists, and "" otherwise.

  3. Set up error with message and "stream".

  4. Set error.[[StreamErrorCode]] to init.streamErrorCode if it exists.

9.3. Attributes

source, of type WebTransportErrorSource, readonly

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

streamErrorCode, of type octet, readonly, nullable

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

9.4. Procedures

To create a WebTransportError error with a WebTransportErrorSource source, run these steps:

  1. Let message be an implementation-defined string.

  2. Let error be a new WebTransportError.

  3. Set up error with message and source.

To set up a WebTransportError error with a DOMString message and a WebTransportErrorSource source, run these steps:

  1. Set error’s internal slots as:

    [[StreamErrorCode]]

    null

    [[Source]]

    source

  2. Run new DOMException(message, name) constructor steps with error, message and "WebTransportError".

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

10. Protocol Mappings

This section is non-normative.

This section describes the [QUIC] protocol behavior of methods defined in this specification, utilizing [WEB-TRANSPORT-HTTP3]. Cause and effect may not be immediate due to buffering.

API Method QUIC Protocol Action
writable.abort(errorCode) sends RESET_STREAM with errorCode
writable.close() sends STREAM_FINAL
writable.getWriter().write() sends STREAM
writable.getWriter().close() sends STREAM_FINAL
writable.getWriter().abort(errorCode) sends RESET_STREAM with errorCode
readable.cancel(errorCode) sends STOP_SENDING with errorCode
readable.getReader().cancel(errorCode) sends STOP_SENDING with errorCode
wt.close(closeInfo) terminates session with closeInfo
QUIC Protocol Action API Effect
received STOP_SENDING with errorCode errors writable with streamErrorCode
received STREAM (await readable.getReader().read()).value
received STREAM_FINAL (await readable.getReader().read()).done
received RESET_STREAM with errorCode 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

11. 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.

11.1. 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.

11.2. State Persistence

WebTransport by itself does not 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, none of the transports defined in this document automatically send cookies, support HTTP authentication or caching invalidation mechanisms. Since they use TLS, they may support TLS session tickets, which could be used by the server (though not by passive network observers) to correlate different connections from the same client. This is not specific to WebTransport by itself, but rather an inherent property of all TLS-based protocols; thus, this is out-of-scope for this specification.

11.3. 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.

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

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.

11.4. 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 [RFC6125]. 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.

12. Examples

12.1. 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 datagram of datagrams) {
    await writer.ready;
    writer.write(datagram).catch(() => {});
  }
}

12.2. 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 datagram = createDatagram();
  setInterval(() => writer.write(datagram).catch(() => {}), ms);
}

12.3. 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
  }
}

12.4. 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();
  await writer.write(data);
  await writer.close();
}

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);
}

12.5. 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, reporting per-stream errors
    ((async () => {
      try {
        for await (const chunk of readable) {
          processTheData(chunk);
        }
      } 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);
    }
  }
}

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

        const writer = duplexStream.writable.getWriter();
        await writer.write(data);
        await writer.close();
        addToEventLog(`Sent bidirectional stream #${n} with data: ${rawData}`);
        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 chunk of readable.pipeThrough(decoder)) {
      addToEventLog(`Received data on stream #${number}: ${chunk}`);
    }
    addToEventLog(`Stream #${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();
  }
}

13. 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. 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-2]
Ilya Grigorik. High Resolution Time Level 2. 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/
[QUIC]
Jana Iyengar; Martin Thomson. QUIC: A UDP-Based Multiplexed and Secure Transport. Internet-Draft. URL: https://tools.ietf.org/html/draft-ietf-quic-transport
[QUIC-DATAGRAM]
Tommy Pauly; Eric Kinnear; David Schinazi. An Unreliable Datagram Extension to QUIC. Internet-Draft. URL: https://datatracker.ietf.org/doc/html/draft-ietf-quic-datagram/
[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
[RFC6125]
P. Saint-Andre; J. Hodges. Representation and Verification of Domain-Based Application Service Identity within Internet Public Key Infrastructure Using X.509 (PKIX) Certificates in the Context of Transport Layer Security (TLS). March 2011. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc6125
[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
[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-HTTP3]
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

IDL Index

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

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

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

  Promise<WebTransportStats> getStats();
  readonly attribute Promise<undefined> ready;
  readonly attribute WebTransportReliabilityMode reliability;
  readonly attribute Promise<WebTransportCloseInfo> closed;
  undefined close(optional WebTransportCloseInfo closeInfo = {});

  readonly attribute WebTransportDatagramDuplexStream datagrams;

  Promise<WebTransportBidirectionalStream> createBidirectionalStream();
  /* a ReadableStream of WebTransportBidirectionalStream objects */
  readonly attribute ReadableStream incomingBidirectionalStreams;

  Promise<WebTransportSendStream> createUnidirectionalStream();
  /* a ReadableStream of WebTransportReceiveStream objects */
  readonly attribute ReadableStream incomingUnidirectionalStreams;
};

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

dictionary WebTransportHash {
  DOMString algorithm;
  BufferSource value;
};

dictionary WebTransportOptions {
  boolean allowPooling = false;
  boolean requireUnreliable = false;
  sequence<WebTransportHash> serverCertificateHashes;
};

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

dictionary WebTransportStats {
  DOMHighResTimeStamp timestamp;
  unsigned long long bytesSent;
  unsigned long long packetsSent;
  unsigned long long packetsLost;
  unsigned long numOutgoingStreamsCreated;
  unsigned long numIncomingStreamsCreated;
  unsigned long long bytesReceived;
  unsigned long long packetsReceived;
  DOMHighResTimeStamp smoothedRtt;
  DOMHighResTimeStamp rttVariation;
  DOMHighResTimeStamp minRtt;
  WebTransportDatagramStats datagrams;
};

dictionary WebTransportDatagramStats {
  DOMHighResTimeStamp timestamp;
  unsigned long long expiredOutgoing;
  unsigned long long droppedIncoming;
  unsigned long long lostOutgoing;
};

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

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

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

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

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

[Exposed=(Window,Worker), SecureContext]
interface WebTransportError : DOMException {
  constructor(optional WebTransportErrorInit init = {});

  readonly attribute WebTransportErrorSource source;
  readonly attribute octet? streamErrorCode;
};

dictionary WebTransportErrorInit {
  [Clamp] octet streamErrorCode;
  DOMString message;
};

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

Issues Index

This needs to be done in workers too. See #127 and whatwg/html#6731.
Now that quic-transport has been removed, this section needs to be revised. Some of those are safe to expose for HTTP/2 and HTTP/3 connections (like min-RTT), while most would either result in information disclosure or are impossible to define for pooled connections.