WebTransport

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.

These should not be confused with protocol names which is an application-level API construct.

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]:

To establish a WebTransport session with an origin origin and a protocols array, 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. Additionally, if the protocols array is non-empty, add a WT-Available-Protocols header field to the CONNECT request, containing isomorphic encoded protocols from protocols in the order given, following [WEB-TRANSPORT-HTTP3] Section 3.4.

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:

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:

A WebTransport stream has the following signals:

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.

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 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 DOMString protocol;

  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.

5.2. Constructor

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.

Note: Whether the incoming streams already have data on them will depend on server behavior.

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.

Note: Whether the incoming streams already have data on them will depend on server behavior.

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.

protocol, of type DOMString, readonly

Once a WebTransport session has been established and the protocols constructor option was used to provide a non-empty array, returns the application-level protocol selected by the server, if any. Otherwise, an empty string. The getter steps are to return this's [[Protocol]].

5.4. Methods

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.

   3. 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.

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

   5. 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.

5. 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.

Note: There is no expectation that the server will be aware of the stream until data is sent on it.

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.

Note: There is no expectation that the server will be aware of the stream until data is sent on it.

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

5.6. Session termination not initiated by the client

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;
  sequence<DOMString> protocols = [];
};

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.

protocols, of type sequence<DOMString>, defaulting to []

An optionally provided array of application-level protocol names. The connection will only be established if the server reports that it has selected one of these application-level protocols.

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 = 0;
  unsigned long long packetsSent = 0;
  unsigned long long bytesLost = 0;
  unsigned long long packetsLost = 0;
  unsigned long long bytesReceived = 0;
  unsigned long long packetsReceived = 0;
  required DOMHighResTimeStamp smoothedRtt;
  required DOMHighResTimeStamp rttVariation;
  required DOMHighResTimeStamp minRtt;
  required WebTransportDatagramStats datagrams;
  unsigned long long? estimatedSendRate = null;
  boolean atSendCapacity = false;
};

The dictionary SHALL have the following attributes:

bytesSent, of type unsigned long long, defaulting to 0

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, defaulting to 0

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

bytesLost, of type unsigned long long, defaulting to 0

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, defaulting to 0

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, defaulting to 0

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, defaulting to 0

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, defaulting to null

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.

atSendCapacity, of type boolean, defaulting to false

A value of false indicates the estimatedSendRate might be application limited, meaning the application is sending significantly less data than the congestion controller allows. A congestion controller might produce a poor estimate of the available network capacity while it is application limited.

A value of true indicates the application is sending data at network capacity, and the estimatedSendRate reflects the network capacity available to the application.

When atSendCapacity is true, the estimatedSendRate reflects a ceiling. As long as the application send rate is sustained, the estimatedSendRate will adapt to network conditions. However, estimatedSendRate is allowed to be null while atSendCapacity is true.

5.13. WebTransportDatagramStats Dictionary

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

dictionary WebTransportDatagramStats {
  unsigned long long droppedIncoming = 0;
  unsigned long long expiredIncoming = 0;
  unsigned long long expiredOutgoing = 0;
  unsigned long long lostOutgoing = 0;
};

The dictionary SHALL have the following attributes:

droppedIncoming, of type unsigned long long, defaulting to 0

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, defaulting to 0

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, defaulting to 0

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, defaulting to 0

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.

6.4. Procedures

6.5. STOP_SENDING signal coming from the server

6.6. WebTransportSendStreamStats Dictionary

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

dictionary WebTransportSendStreamStats {
  unsigned long long bytesWritten = 0;
  unsigned long long bytesSent = 0;
  unsigned long long bytesAcknowledged = 0;
};

The dictionary SHALL have the following attributes:

bytesWritten, of type unsigned long long, defaulting to 0

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

bytesSent, of type unsigned long long, defaulting to 0

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, defaulting to 0

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.

7.3. Procedures

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.

8.3. Procedures

8.4. Reset signal coming from the server

8.5. WebTransportReceiveStreamStats Dictionary

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

dictionary WebTransportReceiveStreamStats {
  unsigned long long bytesReceived = 0;
  unsigned long long bytesRead = 0;
};

The dictionary SHALL have the following attributes:

bytesReceived, of type unsigned long long, defaulting to 0

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, defaulting to 0

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.

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

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

11. WebTransportError Interface

WebTransportError is a subclass of DOMException that represents

• An error coming from the server or the network, or

• A reason for a client-initiated abort operation.

[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.

11.2. Constructor

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.

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.

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.

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.

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

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();
}

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