WebRTC: Real-Time Communication in Browsers

Abstract

This document defines a set of ECMAScript APIs in WebIDL to allow media and generic application data to be sent to and received from another browser or device implementing the appropriate set of real-time protocols. This specification is being developed in conjunction with a protocol specification developed by the IETF RTCWEB group and an API specification to get access to local media devices.

Status of This Document

This section describes the status of this document at the time of its publication. A list of current W3C publications and the latest revision of this technical report can be found in the W3C technical reports index at https://www.w3.org/TR/.

This document includes Candidate Amendments.

Its associated test suite has been used to build an implementation report of the API at the time of its initial publication as a Recommendation. That test suite has been updated to integrate proposed and candidates amendments identified since then, and an updated implementation report focused on the implementation status of these amendments has been used to select features with double implementation as proposed amendments, now fully incorporated in this version of the Recommendation.

This document was published by the Web Real-Time Communications Working Group as a Recommendation using the Recommendation track. It includes candidate amendments, introducing substantive changes and new features since the previous Recommendation.

W3C recommends the wide deployment of this specification as a standard for the Web.

A W3C Recommendation is a specification that, after extensive consensus-building, is endorsed by W3C and its Members, and has commitments from Working Group members to royalty-free licensing for implementations. Future updates to this Recommendation may incorporate new features.

Candidate additions are marked in the document.

Candidate corrections are marked in the document.

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

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

This section is non-normative.

There are a number of facets to peer-to-peer communications and video-conferencing in HTML covered by this specification:

Connecting to remote peers using NAT-traversal technologies such as ICE, STUN, and TURN.
Sending the locally-produced tracks to remote peers and receiving tracks from remote peers.
Sending arbitrary data directly to remote peers.

This document defines the APIs used for these features. This specification is being developed in conjunction with a protocol specification developed by the IETF RTCWEB group and an API specification to get access to local media devices [GETUSERMEDIA] developed by the WebRTC Working Group. An overview of the system can be found in [RFC8825] and [RFC8826].

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

This 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.

The EventHandler interface, representing a callback used for event handlers, is defined in [HTML].

The concepts queue a task and networking task source are defined in [HTML].

The concept fire an event is defined in [DOM].

The terms event, event handlers and event handler event types are defined in [HTML].

Performance.timeOrigin and Performance.now() are defined in [hr-time].

The terms serializable objects, serialization steps, and deserialization steps are defined in [HTML].

The terms MediaStream, MediaStreamTrack, and MediaStreamConstraints are defined in [GETUSERMEDIA]. Note that MediaStream is extended in 9.2 MediaStream in this document while MediaStreamTrack is extended in 9.3 MediaStreamTrack in this document.

The term Blob is defined in [FILEAPI].

The term media description is defined in [RFC4566].

The term media transport is defined in [RFC7656].

The term generation is defined in [RFC8838] Section 2.

The terms stats object and monitored object are defined in [WEBRTC-STATS].

When referring to exceptions, the terms throw and created are defined in [WEBIDL].

The callback VoidFunction is defined in [WEBIDL].

The term "throw" is used as specified in [INFRA]: it terminates the current processing steps.

The terms fulfilled, rejected, resolved, and settled used in the context of Promises are defined in [ECMASCRIPT-6.0].

The AlgorithmIdentifier is defined in [WebCryptoAPI].

Note

The general principles for Javascript APIs apply, including the principle of run-to-completion and no-data-races as defined in [API-DESIGN-PRINCIPLES]. That is, while a task is running, external events do not influence what's visible to the Javascript application. For example, the amount of data buffered on a data channel will increase due to "send" calls while Javascript is executing, and the decrease due to packets being sent will be visible after a task checkpoint.
It is the responsibility of the user agent to make sure the set of values presented to the application is consistent - for instance that getContributingSources() (which is synchronous) returns values for all sources measured at the same time.

This section is non-normative.

An RTCPeerConnection instance allows an application to establish peer-to-peer communications with another RTCPeerConnection instance in another browser, or to another endpoint implementing the required protocols. Communications are coordinated by the exchange of control messages (called a signaling protocol) over a signaling channel which is provided by unspecified means, but generally by a script in the page via the server, e.g. using WebSocket or XMLHttpRequest.

The RTCConfiguration defines a set of parameters to configure how the peer-to-peer communication established via RTCPeerConnection is established or re-established.

dictionary RTCConfiguration {
  sequence<RTCIceServer> iceServers = [];
  RTCIceTransportPolicy iceTransportPolicy = "all";
  RTCBundlePolicy bundlePolicy = "balanced";
  RTCRtcpMuxPolicy rtcpMuxPolicy = "require";
  sequence<RTCCertificate> certificates = [];
  [EnforceRange] octet iceCandidatePoolSize = 0;
};

iceServers of type sequence<RTCIceServer>, defaulting to [].

An array of objects describing servers available to be used by ICE, such as STUN and TURN servers. If the number of ICE servers exceeds an implementation-defined limit, ignore the ICE servers above the threshold. This implementation defined limit MUST be at least 32.

iceTransportPolicy of type RTCIceTransportPolicy, defaulting to "all".

Indicates which candidates the ICE Agent is allowed to use.

bundlePolicy of type RTCBundlePolicy, defaulting to "balanced".

Indicates which media-bundling policy to use when gathering ICE candidates.

rtcpMuxPolicy of type RTCRtcpMuxPolicy, defaulting to "require".

Indicates which rtcp-mux policy to use when gathering ICE candidates.

certificates of type sequence<RTCCertificate>, defaulting to [].

A set of certificates that the RTCPeerConnection uses to authenticate.

Valid values for this parameter are created through calls to the generateCertificate() function.

Although any given DTLS connection will use only one certificate, this attribute allows the caller to provide multiple certificates that support different algorithms. The final certificate will be selected based on the DTLS handshake, which establishes which certificates are allowed. The RTCPeerConnection implementation selects which of the certificates is used for a given connection; how certificates are selected is outside the scope of this specification.

Note

Existing implementations only utilize the first certificate provided; the others are ignored.

If this value is absent, then a default set of certificates is generated for each RTCPeerConnection instance.

This option allows applications to establish key continuity. An RTCCertificate can be persisted in [INDEXEDDB] and reused. Persistence and reuse also avoids the cost of key generation.

The value for this configuration option cannot change after its value is initially selected.

iceCandidatePoolSize of type octet, defaulting to 0

Size of the prefetched ICE pool as defined in [RFC9429] (section 3.5.4. and section 4.1.1.).

The RTCIceServer dictionary is used to describe the STUN and TURN servers that can be used by the ICE Agent to establish a connection with a peer.

dictionary RTCIceServer {
  required (DOMString or sequence<DOMString>) urls;
  DOMString username;
  DOMString credential;
};

urls of type (DOMString or sequence<DOMString>), required

STUN or TURN URI(s) as defined in [RFC7064] and [RFC7065] or other URI types.

username of type DOMString

If this RTCIceServer object represents a TURN server, then this attribute specifies the username to use with that TURN server.

credential of type DOMString

If this RTCIceServer object represents a TURN server, then this attribute specifies the credential to use with that TURN server.

credential represents a long-term authentication password, as described in [RFC5389], Section 10.2.

An example array of RTCIceServer objects is:

Example 1

[
  {urls: 'stun:stun1.example.net'},
  {urls: ['turns:turn.example.org', 'turn:turn.example.net'],
    username: 'user',
    credential: 'myPassword',
];

As described in [RFC9429] (section 4.1.1.), if the iceTransportPolicy member of the RTCConfiguration is specified, it defines the ICE candidate policy [RFC9429] (section 3.5.3.) the browser uses to surface the permitted candidates to the application; only these candidates will be used for connectivity checks.

enum RTCIceTransportPolicy {
  "relay",
  "all"
};

`RTCIceTransportPolicy` Enumeration description
Enum value	Description
`relay`	The ICE Agent uses only media relay candidates such as candidates passing through a TURN server. Note This can be used to prevent the remote endpoint from learning the user's IP addresses, which may be desired in certain use cases. For example, in a "call"-based application, the application may want to prevent an unknown caller from learning the callee's IP addresses until the callee has consented in some way.
`all`	The ICE Agent can use any type of candidate when this value is specified. Note The implementation can still use its own candidate filtering policy in order to limit the IP addresses exposed to the application, as noted in the description of `RTCIceCandidate`.`address`.

As described in [RFC9429] (section 4.1.1.), bundle policy affects which media tracks are negotiated if the remote endpoint is not bundle-aware, and what ICE candidates are gathered. If the remote endpoint is bundle-aware, all media tracks and data channels are bundled onto the same transport.

enum RTCBundlePolicy {
  "balanced",
  "max-compat",
  "max-bundle"
};

`RTCBundlePolicy` Enumeration description
Enum value	Description
`balanced`	Gather ICE candidates for each media type in use (audio, video, and data). If the remote endpoint is not bundle-aware, negotiate only one audio and video track on separate transports.
`max-compat`	Gather ICE candidates for each track. If the remote endpoint is not bundle-aware, negotiate all media tracks on separate transports.
`max-bundle`	Gather ICE candidates for only one track. If the remote endpoint is not bundle-aware, negotiate only one media track.

As described in [RFC9429] (section 4.1.1.), the RTCRtcpMuxPolicy affects what ICE candidates are gathered to support non-multiplexed RTCP. The only value defined in this spec is "require".

enum RTCRtcpMuxPolicy {
  "require"
};

`RTCRtcpMuxPolicy` Enumeration description
Enum value	Description
`require`	Gather ICE candidates only for RTP and multiplex RTCP on the RTP candidates. If the remote endpoint is not capable of rtcp-mux, session negotiation will fail.

These dictionaries describe the options that can be used to control the offer/answer creation process.

dictionary RTCOfferAnswerOptions {};

dictionary RTCOfferOptions : RTCOfferAnswerOptions {
  boolean iceRestart = false;
};

iceRestart of type boolean, defaulting to false

When the value of this dictionary member is true, or the relevant RTCPeerConnection object's [[LocalIceCredentialsToReplace]] slot is not empty, then the generated description will have ICE credentials that are different from the current credentials (as visible in the currentLocalDescription attribute's SDP). Applying the generated description will restart ICE, as described in section 9.1.1.1 of [RFC5245].

When the value of this dictionary member is false, and the relevant RTCPeerConnection object's [[LocalIceCredentialsToReplace]] slot is empty, and the currentLocalDescription attribute has valid ICE credentials, then the generated description will have the same ICE credentials as the current value from the currentLocalDescription attribute.

Note

Performing an ICE restart is recommended when iceConnectionState transitions to "failed". An application may additionally choose to listen for the iceConnectionState transition to "disconnected" and then use other sources of information (such as using getStats to measure if the number of bytes sent or received over the next couple of seconds increases) to determine whether an ICE restart is advisable.

The RTCAnswerOptions dictionary describe options specific to session description of type "answer" (none in this version of the specification).

dictionary RTCAnswerOptions : RTCOfferAnswerOptions {};

enum RTCSignalingState {
  "stable",
  "have-local-offer",
  "have-remote-offer",
  "have-local-pranswer",
  "have-remote-pranswer",
  "closed"
};

`RTCSignalingState` Enumeration description
Enum value	Description
`stable`	There is no offer/answer exchange in progress. This is also the initial state, in which case the local and remote descriptions are empty.
`have-local-offer`	A local description, of type "`offer`", has been successfully applied.
`have-remote-offer`	A remote description, of type "`offer`", has been successfully applied.
`have-local-pranswer`	A remote description of type "`offer`" has been successfully applied and a local description of type "`pranswer`" has been successfully applied.
`have-remote-pranswer`	A local description of type "`offer`" has been successfully applied and a remote description of type "`pranswer`" has been successfully applied.
`closed`	The `RTCPeerConnection` has been closed; its `[[IsClosed]]` slot is `true`.

signaling state transition diagram — Figure 1 Non-normative signaling state transitions diagram. Method calls abbreviated.

An example set of transitions might be:

Caller transition:

new RTCPeerConnection(): "stable"
setLocalDescription(offer): "have-local-offer"
setRemoteDescription(pranswer): "have-remote-pranswer"
setRemoteDescription(answer): "stable"

Callee transition:

new RTCPeerConnection(): "stable"
setRemoteDescription(offer): "have-remote-offer"
setLocalDescription(pranswer): "have-local-pranswer"
setLocalDescription(answer): "stable"

enum RTCIceGatheringState {
  "new",
  "gathering",
  "complete"
};

`RTCIceGatheringState` Enumeration description
Enum value	Description
`new`	Any of the `RTCIceTransport`s are in the "`new`" gathering state and none of the transports are in the "`gathering`" state, or there are no transports.
`gathering`	Any of the `RTCIceTransport`s are in the "`gathering`" state.
`complete`	At least one `RTCIceTransport` exists, and all `RTCIceTransport`s are in the "`complete`" gathering state.

The set of transports considered is the one presently referenced by the RTCPeerConnection's set of transceivers and the RTCPeerConnection's [[SctpTransport]] internal slot if not null.

enum RTCPeerConnectionState {
  "closed",
  "failed",
  "disconnected",
  "new",
  "connecting",
  "connected"
};

`RTCPeerConnectionState` Enumeration description
Enum value	Description
`closed`	`[[IceConnectionState]]` is "`closed`".
`failed`	The previous state doesn't apply, and either `[[IceConnectionState]]` is "`failed`" or any `RTCDtlsTransport`s are in the "`failed`" state.
`disconnected`	None of the previous states apply, and `[[IceConnectionState]]` is "`disconnected`".
`new`	None of the previous states apply, and either `[[IceConnectionState]]` is "`new`", and all `RTCDtlsTransport`s are in the "`new`" or "`closed`" state, or there are no transports.
`connected`	None of the previous states apply, `[[IceConnectionState]]` is "`connected`", and all `RTCDtlsTransport`s are in the "`connected`" or "`closed`" state.
`connecting`	None of the previous states apply.

Note

In the "connecting" state, one or more RTCIceTransports are in the "new" or "checking" state, or one or more RTCDtlsTransports are in the "new" or "connecting" state.

The set of transports considered is the one presently referenced by the RTCPeerConnection's set of transceivers and the RTCPeerConnection's [[SctpTransport]] internal slot if not null.

enum RTCIceConnectionState {
  "closed",
  "failed",
  "disconnected",
  "new",
  "checking",
  "completed",
  "connected"
};

`RTCIceConnectionState` Enumeration description
Enum value	Description
`closed`	The `RTCPeerConnection` object's `[[IsClosed]]` slot is `true`.
`failed`	The previous state doesn't apply and any `RTCIceTransport`s are in the "`failed`" state.
`disconnected`	None of the previous states apply and any `RTCIceTransport`s are in the "`disconnected`" state.
`new`	None of the previous states apply and all `RTCIceTransport`s are in the "`new`" or "`closed`" state, or there are no transports.
`checking`	None of the previous states apply and any `RTCIceTransport`s are in the "`new`" or "`checking`" state.
`completed`	None of the previous states apply and all `RTCIceTransport`s are in the "`completed`" or "`closed`" state.
`connected`	None of the previous states apply and all `RTCIceTransport`s are in the "`connected`", "`completed`" or "`closed`" state.

The set of transports considered is the one presently referenced by the RTCPeerConnection's set of transceivers and the RTCPeerConnection's [[SctpTransport]] internal slot if not null.

Note that if an RTCIceTransport is discarded as a result of signaling (e.g. RTCP mux or bundling), or created as a result of signaling (e.g. adding a new media description), the state may advance directly from one state to another.

The [RFC9429] specification, as a whole, describes the details of how the RTCPeerConnection operates. References to specific subsections of [RFC9429] are provided as appropriate.

Calling new RTCPeerConnection(configuration) creates an RTCPeerConnection object.

configuration.iceServers contains information used to find and access the servers used by ICE. The application can supply multiple servers of each type, and any TURN server MAY also be used as a STUN server for the purposes of gathering server reflexive candidates.

An RTCPeerConnection object has a [[SignalingState]], and the aggregated states [[ConnectionState]], [[IceGatheringState]], and [[IceConnectionState]]. These are initialized when the object is created.

The ICE protocol implementation of an RTCPeerConnection is represented by an ICE agent [RFC5245]. Certain RTCPeerConnection methods involve interactions with the ICE Agent, namely addIceCandidate, setConfiguration, setLocalDescription, setRemoteDescription and close. These interactions are described in the relevant sections in this document and in [RFC9429]. The ICE Agent also provides indications to the user agent when the state of its internal representation of an RTCIceTransport changes, as described in 5.6 RTCIceTransport Interface.

The task source for the tasks listed in this section is the networking task source.

Note

The state of the SDP negotiation is represented by the internal variables [[SignalingState]], [[CurrentLocalDescription]], [[CurrentRemoteDescription]], [[PendingLocalDescription]] and [[PendingRemoteDescription]]. These are only set inside the setLocalDescription and setRemoteDescription operations, and modified by the addIceCandidate operation and the surface a candidate procedure. In each case, all the modifications to all the five variables are completed before the procedures fire any events or invoke any callbacks, so the modifications are made visible at a single point in time.

As one of the unloading document cleanup steps, run the following steps:

Let window be document's relevant global object.
For each RTCPeerConnection object connection whose relevant global object is window, close the connection with connection and the value true.

When the RTCPeerConnection.constructor() is invoked, the user agent MUST run the following steps:

If any of the steps enumerated below fails for a reason not specified here, throw an UnknownError with the message attribute set to an appropriate description.
Let connection be a newly created RTCPeerConnection object.
Let connection have a [[DocumentOrigin]] internal slot, initialized to the relevant settings object's origin.
Let configuration be the method's first argument.
If the certificates value in configuration is non-empty, run the following steps for each certificate in certificates:
1. If the value of certificate.expires is less than the current time, throw an InvalidAccessError.
2. If certificate.[[Origin]] is not same origin with connection.[[DocumentOrigin]], throw an InvalidAccessError.
3. Store certificate.
Else, generate one or more new RTCCertificate instances with this RTCPeerConnection instance and store them. This MAY happen asynchronously and the value of certificates remains undefined for the subsequent steps. As noted in Section 4.3.2.3 of [RFC8826], WebRTC utilizes self-signed rather than Public Key Infrastructure (PKI) certificates, so that the expiration check is to ensure that keys are not used indefinitely and additional certificate checks are unnecessary.
Initialize connection's ICE Agent.
Let connection have a [[Configuration]] internal slot, initialized to null. Set the configuration specified by configuration.
Let connection have an [[IsClosed]] internal slot, initialized to false.
Let connection have a [[NegotiationNeeded]] internal slot, initialized to false.
Let connection have an [[SctpTransport]] internal slot, initialized to null.
Let connection have a [[DataChannels]] internal slot, initialized to an empty ordered set.
Let connection have an [[Operations]] internal slot, representing an operations chain, initialized to an empty list.
Let connection have a [[UpdateNegotiationNeededFlagOnEmptyChain]] internal slot, initialized to false.
Let connection have an [[LastCreatedOffer]] internal slot, initialized to "".
Let connection have an [[LastCreatedAnswer]] internal slot, initialized to "".
Let connection have an [[EarlyCandidates]] internal slot, initialized to an empty list.
Let connection have an [[SignalingState]] internal slot, initialized to "stable".
Let connection have an [[IceConnectionState]] internal slot, initialized to "new".
Let connection have an [[IceGatheringState]] internal slot, initialized to "new".
Let connection have an [[ConnectionState]] internal slot, initialized to "new".
Let connection have a [[PendingLocalDescription]] internal slot, initialized to null.
Let connection have a [[CurrentLocalDescription]] internal slot, initialized to null.
Let connection have a [[PendingRemoteDescription]] internal slot, initialized to null.
Let connection have a [[CurrentRemoteDescription]] internal slot, initialized to null.
Let connection have a [[LocalIceCredentialsToReplace]] internal slot, initialized to an empty set.
Return connection.

An RTCPeerConnection object has an operations chain, [[Operations]], which ensures that only one asynchronous operation in the chain executes concurrently. If subsequent calls are made while the returned promise of a previous call is still not settled, they are added to the chain and executed when all the previous calls have finished executing and their promises have settled.

To chain an operation to an RTCPeerConnection object's operations chain, run the following steps:

Let connection be the RTCPeerConnection object.
If connection.[[IsClosed]] is true, return a promise rejected with a newly created InvalidStateError.
Let operation be the operation to be chained.
Let p be a new promise.
Append operation to [[Operations]].
If the length of [[Operations]] is exactly 1, execute operation.
Upon fulfillment or rejection of the promise returned by the operation, run the following steps:
1. If connection.[[IsClosed]] is true, abort these steps.
2. If the promise returned by operation was fulfilled with a value, fulfill p with that value.
3. If the promise returned by operation was rejected with a value, reject p with that value.
4. Upon fulfillment or rejection of p, execute the following steps:
  1. If connection.[[IsClosed]] is true, abort these steps.
  2. Remove the first element of [[Operations]].
  3. If [[Operations]] is non-empty, execute the operation represented by the first element of [[Operations]], and abort these steps.
  4. If connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] is false, abort these steps.
  5. Set connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to false.
  6. Update the negotiation-needed flag for connection.
Return p.

An RTCPeerConnection object has an aggregated [[ConnectionState]]. Whenever the state of an RTCDtlsTransport changes, the user agent MUST queue a task that runs the following steps:

Let connection be this RTCPeerConnection object associated with the RTCDtlsTransport object whose state changed.
If connection.[[IsClosed]] is true, abort these steps.
Let newState be the value of deriving a new state value as described by the RTCPeerConnectionState enum.
If connection.[[ConnectionState]] is equal to newState, abort these steps.
Set connection.[[ConnectionState]] to newState.
Fire an event named connectionstatechange at connection.

To set a local session description description on an RTCPeerConnection object connection, set the session description description on connection with the additional value false.

To set a remote session description description on an RTCPeerConnection object connection, set the session description description on connection with the additional value true.

To set a session description description on an RTCPeerConnection object connection, given a remote boolean, run the following steps:

Let p be a new promise.
If description.type is "rollback" and connection.[[SignalingState]] is either "stable", "have-local-pranswer", or "have-remote-pranswer", then reject p with a newly created InvalidStateError and abort these steps.
Let jsepSetOfTransceivers be a shallow copy of connection's set of transceivers.
In parallel, start the process to apply description as described in [RFC9429] (section 5.5. and section 5.6.), with these additional restrictions:
1. Use jsepSetOfTransceivers as the source of truth with regard to what "RtpTransceivers" exist, and their [[JsepMid]] internal slot as their "mid property".
2. Candidate Correction 5:Forbid ICE gathering and connectivity checks on administrative prohibited candidates (PR #2708)
  
  If remote is false and this triggers the ICE candidate gathering process in [RFC9429] (section 5.9.), the ICE Agent MUST NOT gather candidates that would be administratively prohibited.
3. Candidate Correction 5:Forbid ICE gathering and connectivity checks on administrative prohibited candidates (PR #2708)
  
  If remote is true and this triggers ICE connectivity checks in [RFC9429] (section 5.10.), the ICE Agent MUST NOT attempt to connect to candidates that are administratively prohibited.
4. If remote is true, validate back-to-back offers as if answers were applied in between, by running the check for subsequent offers as if it were in stable state.
5. Candidate Correction 37:Don't fail sRD(offer) over rid mismatch, just answer with unicast. (PR #2794)
6. If applying description leads to modifying a transceiver transceiver, and transceiver.[[Sender]].[[SendEncodings]] is non-empty, and not equal to the encodings that would result from processing description, the process of applying description fails. This specification does not allow remotely initiated RID renegotiation.
7. If the process to apply description fails for any reason, then the user agent MUST queue a task that runs the following steps:
  1. If connection.[[IsClosed]] is true, then abort these steps.
  2. If description.type is invalid for the current connection.[[SignalingState]] as described in [RFC9429] (section 5.5. and section 5.6.), then reject p with a newly created InvalidStateError and abort these steps.
  3. If the content of description is not valid SDP syntax, then reject p with an RTCError (with errorDetail set to "sdp-syntax-error" and the sdpLineNumber attribute set to the line number in the SDP where the syntax error was detected) and abort these steps.
  4. If remote is true, the connection's RTCRtcpMuxPolicy is require and the description does not use RTCP mux, then reject p with a newly created InvalidAccessError and abort these steps.
  5. If the description attempted to renegotiate RIDs, as described above, then reject p with a newly created InvalidAccessError and abort these steps.
  6. If the content of description is invalid, then reject p with a newly created InvalidAccessError and abort these steps.
  7. For all other errors, reject p with a newly created OperationError.
8. If description is applied successfully, the user agent MUST queue a task that runs the following steps:
  1. If connection.[[IsClosed]] is true, then abort these steps.
  2. If remote is true and description is of type "offer", then if any addTrack() methods on connection succeeded during the process to apply description, abort these steps and start the process over as if they had succeeded prior, to include the extra transceiver(s) in the process.
  3. If any promises from setParameters methods on RTCRtpSenders associated with connection are not settled, abort these steps and start the process over.
  4. If description is of type "offer" and connection.[[SignalingState]] is "stable" then for each transceiver in connection's set of transceivers, run the following steps:
    1. Set transceiver.[[Sender]].[[LastStableStateSenderTransport]] to transceiver.[[Sender]].[[SenderTransport]].
    2. Candidate Correction 13:Rollback restores ridless encoding trounced by sRD(simulcastOffer). (PR #2797)
      
      If transceiver.[[Sender]].[[SendEncodings]].length is 1 and the lone encoding contains no rid member, then set transceiver.[[Sender]].[[LastStableRidlessSendEncodings]] to transceiver.[[Sender]].[[SendEncodings]]; Otherwise, set transceiver.[[Sender]].[[LastStableRidlessSendEncodings]] to null.
    3. Set transceiver.[[Receiver]].[[LastStableStateReceiverTransport]] to transceiver.[[Receiver]].[[ReceiverTransport]].
    4. Set transceiver.[[Receiver]].[[LastStableStateAssociatedRemoteMediaStreams]] to transceiver.[[Receiver]].[[AssociatedRemoteMediaStreams]].
    5. Set transceiver.[[Receiver]].[[LastStableStateReceiveCodecs]] to transceiver.[[Receiver]].[[ReceiveCodecs]].
  5. If remote is false, then run one of the following steps:
    1. If description is of type "offer", set connection.[[PendingLocalDescription]] to a new RTCSessionDescription object constructed from description, set connection.[[SignalingState]] to "have-local-offer", and release early candidates.
    2. If description is of type "answer", then this completes an offer answer negotiation. Set connection.[[CurrentLocalDescription]] to a new RTCSessionDescription object constructed from description, and set connection.[[CurrentRemoteDescription]] to connection.[[PendingRemoteDescription]]. Set both connection.[[PendingRemoteDescription]] and connection.[[PendingLocalDescription]] to null. Set both connection.[[LastCreatedOffer]] and connection.[[LastCreatedAnswer]] to "", set connection.[[SignalingState]] to "stable", and release early candidates. Finally, if none of the ICE credentials in connection.[[LocalIceCredentialsToReplace]] are present in description, then set connection.[[LocalIceCredentialsToReplace]] to an empty set.
    3. If description is of type "pranswer", then set connection.[[PendingLocalDescription]] to a new RTCSessionDescription object constructed from description, set connection.[[SignalingState]] to "have-local-pranswer", and release early candidates.
  6. Otherwise, (if remote is true) run one of the following steps:
    1. If description is of type "offer", set connection.[[PendingRemoteDescription]] attribute to a new RTCSessionDescription object constructed from description, and set connection.[[SignalingState]] to "have-remote-offer".
    2. If description is of type "answer", then this completes an offer answer negotiation. Set connection.[[CurrentRemoteDescription]] to a new RTCSessionDescription object constructed from description, and set connection.[[CurrentLocalDescription]] to connection.[[PendingLocalDescription]]. Set both connection.[[PendingRemoteDescription]] and connection.[[PendingLocalDescription]] to null. Set both connection.[[LastCreatedOffer]] and connection.[[LastCreatedAnswer]] to "", and set connection.[[SignalingState]] to "stable". Finally, if none of the ICE credentials in connection.[[LocalIceCredentialsToReplace]] are present in the newly set connection.[[CurrentLocalDescription]], then set connection.[[LocalIceCredentialsToReplace]] to an empty set.
    3. If description is of type "pranswer", then set connection.[[PendingRemoteDescription]] to a new RTCSessionDescription object constructed from description and set connection.[[SignalingState]] to "have-remote-pranswer".
  7. If description is of type "answer", and it initiates the closure of an existing SCTP association, as defined in [RFC8841], Sections 10.3 and 10.4, set the value of connection.[[SctpTransport]] to null.
  8. Let trackEventInits, muteTracks, addList, removeList and errorList be empty lists.
  9. If description is of type "answer" or "pranswer", then run the following steps:
    1. If description initiates the establishment of a new SCTP association, as defined in [RFC8841], Sections 10.3 and 10.4, create an RTCSctpTransport with an initial state of "connecting" and assign the result to the [[SctpTransport]] slot. Otherwise, if an SCTP association is established, but the max-message-size SDP attribute is updated, update the data max message size of connection.[[SctpTransport]].
    2. If description negotiates the DTLS role of the SCTP transport, then for each RTCDataChannel, channel, with a null id, run the following step:
      1. Give channel a new ID generated according to [RFC8832]. If no available ID could be generated, set channel.[[ReadyState]] to "closed", and add channnel to errorList.
  10. If description is not of type "rollback", then run the following steps:
    1. If remote is false, then run the following steps for each media description in description:
      
      Candidate Correction 26:Prune createAnswer()'s encodings and SendEncodings in sLD(answer). (PR #2801)
      1. If the media description was not yet associated with an RTCRtpTransceiver object then run the following steps:
        Let transceiver be the RTCRtpTransceiver used to create the media description.
        Set transceiver.[[Mid]] to transceiver.[[JsepMid]].
        If transceiver.[[Stopped]] is true, abort these sub steps.
        If the media description is indicated as using an existing ~~media~~ media transport according to [RFC8843], let transport be the RTCDtlsTransport object representing the RTP/RTCP component of that transport.
        Otherwise, let transport be a newly created RTCDtlsTransport object with a new underlying RTCIceTransport.
        Set transceiver.[[Sender]].[[SenderTransport]] to transport.
        Set transceiver.[[Receiver]].[[ReceiverTransport]] to transport.
      2. Let transceiver be the RTCRtpTransceiver associated with the media description.
      3. If transceiver.[[Stopped]] is true, abort these sub steps.
      4. Let direction be an RTCRtpTransceiverDirection value representing the direction from the ~~media~~ media description.
      5. If direction is "sendrecv" or "recvonly", set transceiver.[[Receptive]] to true, otherwise set it to false.
      6. Set transceiver.[[Receiver]].[[ReceiveCodecs]] to the codecs that description negotiates for receiving and which the user agent is currently prepared to receive.
        Note
        If the direction is "sendonly" or "inactive", the receiver is not prepared to receive anything, and the list will be empty.
      7. If description is of type "answer" or "pranswer", then run the following steps:
        If transceiver. [[Sender]].[[SendEncodings]] .length is greater than 1, then run the following steps:
        If description is missing all of the previously negotiated layers, then remove all dictionaries in transceiver.[[Sender]].[[SendEncodings]] except the first one, and skip the next step.
        If description is missing any of the previously negotiated layers, then remove the dictionaries that correspond to the missing layers from transceiver.[[Sender]].[[SendEncodings]].
        Set transceiver.[[Sender]].[[SendCodecs]] to the codecs that description negotiates for sending and which the user agent is currently capable of sending, and set transceiver.[[Sender]].[[LastReturnedParameters]] to null.
        If direction is "sendonly" or "inactive", and transceiver.[[FiredDirection]] is either "sendrecv" or "recvonly", then run the following steps:
        Set the associated remote streams given transceiver.[[Receiver]], an empty list, another empty list, and removeList.
        process the removal of a ~~remote~~ remote track for the media description, given transceiver and muteTracks.
        Set transceiver.[[CurrentDirection]] and transceiver.[[FiredDirection]] to direction.
    2. Otherwise, (if remote is true) run the following steps for each media description in description:
      
      Candidate Correction 12:Remove interaction between encoding.active and simulcast ~rid (PR #2754)
      Candidate Correction 14:Make RTCTransceiver.direction reflects local preference in offers and answers (PR #2759)
      Candidate Correction 22:Allow remote offer rid pruning of encodings through the client answer. (PR #2758)
      Candidate Correction 37:Don't fail sRD(offer) over rid mismatch, just answer with unicast. (PR #2794)
      Candidate Correction 25:Remove duplicate rids in proposedSendEncodings. (PR #2800)
      Candidate Correction 27:Ignore comma-separated rid alternatives. (PR #2813)
      1. If the description is of type "offer" and the media description contains a request to receive simulcast, use the order of the rid values specified in the simulcast attribute to create an RTCRtpEncodingParameters dictionary for each of the simulcast layers, populating the rid member according to the corresponding rid ~~value~~value (using only the first value if comma-separated alternatives exist), and let ~~sendEncodings~~proposedSendEncodings be ~~the list~~ the list containing the created dictionaries. Otherwise, let ~~sendEncodings~~ proposedSendEncodings be an an empty list.
      2. For each encoding, encoding, in proposedSendEncodings in reverse order, if encoding's rid matches that of another encoding in proposedSendEncodings, remove encoding from proposedSendEncodings.
      3. Let supportedEncodings be the maximum number of encodings that the implementation can support. If the length of ~~sendEncodings~~proposedSendEncodings is greater than supportedEncodings, truncate ~~sendEncodings~~proposedSendEncodings so that its length is supportedEncodings.
      4. If ~~sendEncodings~~proposedSendEncodings is non-empty, ~~set~~ set each encoding's scaleResolutionDownBy to 2^(length of sendEncodingsproposedSendEncodings - encoding index - 1).
      5. As described by [~~RFC8829~~RFC9429] (section 5.10.), attempt to find an existing RTCRtpTransceiver object, transceiver, to represent the media description.
      6. If a suitable transceiver was found (transceiver is set), and ~~sendEncodings~~proposedSendEncodings is non-empty, ~~set transceiver.[[Sender]].[[SendEncodings]]~~ ~~to sendEncodings, and set transceiver.[[Sender]].[[LastReturnedParameters]] to null.~~run the following steps:
        If the length of transceiver.[[Sender]].[[SendEncodings]] is 1, and the lone encoding contains no rid member, set transceiver.[[Sender]].[[SendEncodings]] to proposedSendEncodings, and set transceiver.[[Sender]].[[LastReturnedParameters]] to null.
      7. If no suitable transceiver was found (transceiver is unset), run the following steps:
        Create an RTCRtpSender, sender, from the ~~media~~ media description using ~~sendEncodings~~proposedSendEncodings.
        Create an RTCRtpReceiver, receiver, from the ~~media~~ media description.
        Create an RTCRtpTransceiver with sender, receiver and an RTCRtpTransceiverDirection value of "recvonly", and let transceiver be the result.
        Add transceiver to the connection's set of of transceivers.
      8. If description is of type "answer" or "pranswer", and transceiver. [[Sender]].[[SendEncodings]] .length is greater than 1, then run the following steps:
        If description indicates that simulcast is not supported or desired, or description is missing all of the previously negotiated layers, then remove all dictionaries in transceiver.[[Sender]].[[SendEncodings]] except the first one and abort these sub steps.
        If description ~~rejects~~ is missing any of the ~~offered~~ previously negotiated layers, ~~then~~ then remove ~~the~~ the dictionaries that correspond ~~to rejected~~ to the missing layers from transceiver.[[Sender]].[[SendEncodings]].
        Update the paused status as indicated by [RFC8853] of each simulcast layer by setting the active member on the corresponding dictionaries in transceiver.[[Sender]].[[SendEncodings]] to true for unpaused or to false for paused.
      9. Set transceiver.[[Mid]] to transceiver.[[JsepMid]].
      10. Let direction be an RTCRtpTransceiverDirection value representing the direction from the ~~media~~ media description, but with the send and receive directions reversed to represent this peer's point of view. If the media description is rejected, set direction to "inactive".
      11. If direction is "sendrecv" or "recvonly", let msids be a list of the MSIDs that the media description indicates transceiver.[[Receiver]].[[ReceiverTrack]] is to be associated with. Otherwise, let msids be an empty list.
        Note
        msids will be an empty list here if media description is rejected.
      12. Process remote tracks with transceiver, direction, msids, addList, removeList, and trackEventInits.
      13. Set transceiver.[[Receiver]].[[ReceiveCodecs]] to the codecs that description negotiates for receiving and which the user agent is currently prepared to receive.
      14. If description is of type "answer" or "pranswer", then run the following steps:
        Set transceiver.[[Sender]].[[SendCodecs]] to the codecs that description negotiates for sending and which the user agent is currently capable of sending.
        Set transceiver.[[CurrentDirection]] ~~and transceiver.[[Direction]]s~~ to direction.
        Let transport be the RTCDtlsTransport object representing the RTP/RTCP component of the ~~media~~ media transport used by transceiver's associated media description, according to [RFC8843].
        Set transceiver.[[Sender]].[[SenderTransport]] to transport.
        Set transceiver.[[Receiver]].[[ReceiverTransport]] to transport.
        Set the [[IceRole]] of transport according to the rules of [RFC8445].
        Note
        The rules of [RFC8445] that apply here are:
        If [[IceRole]] is not unknown, do not modify [[IceRole]].
        If description is a local offer, set it to controlling.
        If description is a remote offer, and contains a=ice-lite, set [[IceRole]] to controlling.
        If description is a remote offer, and does not contain a=ice-lite, set [[IceRole]] to controlled.
        This ensures that [[IceRole]] always has a value after the first offer is processed.
      15. If the media description is rejected, and transceiver.[[Stopped]] is false, then stop ~~the~~ the RTCRtpTransceiver transceiver.
  11. Otherwise, (if description is of type "rollback") run the following steps:
    1. Let pendingDescription be either connection.[[PendingLocalDescription]] or connection.[[PendingRemoteDescription]], whichever one is not null.
    2. For each transceiver in the connection's set of transceivers run the following steps:
      1. If transceiver was not associated with a media description prior to pendingDescription being set, disassociate it and set both transceiver.[[JsepMid]] and transceiver.[[Mid]] to null.
      2. Set transceiver.[[Sender]].[[SenderTransport]] to transceiver.[[Sender]].[[LastStableStateSenderTransport]].
      3. Candidate Correction 13:Rollback restores ridless encoding trounced by sRD(simulcastOffer). (PR #2797)
        
        If transceiver.[[Sender]].[[LastStableRidlessSendEncodings]] is not null, and any encoding in transceiver.[[Sender]].[[SendEncodings]] contains a rid member, then set transceiver.[[Sender]].[[SendEncodings]] to transceiver.[[Sender]].[[LastStableRidlessSendEncodings]].
      4. Set transceiver.[[Receiver]].[[ReceiverTransport]] to transceiver.[[Receiver]].[[LastStableStateReceiverTransport]].
      5. Set transceiver.[[Receiver]].[[ReceiveCodecs]] to transceiver.[[Receiver]].[[LastStableStateReceiveCodecs]].
      6. If connection.[[SignalingState]] is "have-remote-offer", run the following sub steps:
        
        Let msids be a list of the ids of all MediaStream objects in transceiver.[[Receiver]].[[LastStableStateAssociatedRemoteMediaStreams]], or an empty list if there are none.
        
        Process remote tracks with transceiver, transceiver.[[CurrentDirection]], msids, addList, removeList, and trackEventInits.
      7. If transceiver was created when pendingDescription was set, and a track has never been attached to it via addTrack(), then stop the RTCRtpTransceiver transceiver, and remove it from connection's set of transceivers.
    3. Set connection.[[PendingLocalDescription]] and connection.[[PendingRemoteDescription]] to null, and set connection.[[SignalingState]] to "stable".
  12. If description is of type "answer", then run the following steps:
    1. For each transceiver in the connection's set of transceivers run the following steps:
      1. If transceiver is stopped, associated with an m= section and the associated m= section is rejected in connection.[[CurrentLocalDescription]] or connection.[[CurrentRemoteDescription]], remove the transceiver from the connection's set of transceivers.
  13. If connection.[[SignalingState]] is now "stable", run the following steps:
    1. For any transceiver that was removed from the set of transceivers in a previous step, if any of its transports (transceiver.[[Sender]].[[SenderTransport]] or transceiver.[[Receiver]].[[ReceiverTransport]]) are still not closed and they're no longer referenced by a non-stopped transceiver, close the RTCDtlsTransports and their associated RTCIceTransports. This results in events firing on these objects in a queued task.
    2. Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)
      For each transceiver in connection's set of transceivers:
      
      Let codecs be transceiver.[[Sender]].[[SendCodecs]].
      
      If codecs is not an empty list:
      
      For each encoding in transceiver.[[Sender]].[[SendEncodings]], if encoding.codec does not match any entry in codecs, using the codec dictionary match algorithm with ignoreLevels set to true, remove encoding.codec.
    3. Clear the negotiation-needed flag and update the negotiation-needed flag.
  14. If connection.[[SignalingState]] changed above, fire an event named signalingstatechange at connection.
  15. For each channel in errorList, fire an event named error using the RTCErrorEvent interface with the errorDetail attribute set to "data-channel-failure" at channel.
  16. For each track in muteTracks, set the muted state of track to the value true.
  17. For each stream and track pair in removeList, remove the track track from stream.
  18. For each stream and track pair in addList, add the track track to stream.
  19. For each entry entry in trackEventInits, fire an event named track using the RTCTrackEvent interface with its receiver attribute initialized to entry.receiver, its track attribute initialized to entry.track, its streams attribute initialized to entry.streams and its transceiver attribute initialized to entry.transceiver at the connection object.
  20. Resolve p with undefined.
Return p.

To set a configuration with configuration, run the following steps:

Let connection be the target RTCPeerConnection object.
Let oldConfig be connection.[[Configuration]].
If oldConfig is not null, run the following steps, and if any of them fail, throw an InvalidModificationError:
1. If the length of configuration.certificates is different from the length of oldConfig.certificates, fail.
2. Let index be 0.
3. While index is less than the length of configuration.certificates, run the following steps:
  1. If the ECMAScript object represented by the value of configuration.certificates at index is not the same as the ECMAScript object represented by the value of oldConfig.certificates at index, then fail.
  2. Increment index by 1.
4. If the value of configuration.bundlePolicy differs from oldConfig.bundlePolicy, then fail.
5. If the value of configuration.rtcpMuxPolicy differs from oldConfig.rtcpMuxPolicy, then fail.
6. If the value of configuration.iceCandidatePoolSize differs from oldConfig.iceCandidatePoolSize, and setLocalDescription has already been called, then fail.
Let iceServers be configuration.iceServers.
Truncate iceServers to the maximum number of supported elements.
For each server in iceServers, run the following steps:
1. Let urls be server.urls.
2. If urls is a string, set urls to a list consisting of just that string.
3. If urls is empty, throw a "SyntaxError" DOMException.
4. For each url in urls, run the validate an ICE server URL algorithm on url.
Set the ICE Agent's ICE transports setting to the value of configuration.iceTransportPolicy. As defined in [RFC9429] (section 4.1.18.), if the new ICE transports setting changes the existing setting, no action will be taken until the next gathering phase. If a script wants this to happen immediately, it should do an ICE restart.
Set the ICE Agent's prefetched ICE candidate pool size as defined in [RFC9429] (section 3.5.4. and section 4.1.1.) to the value of configuration.iceCandidatePoolSize. If the new ICE candidate pool size changes the existing setting, this may result in immediate gathering of new pooled candidates, or discarding of existing pooled candidates, as defined in [RFC9429] (section 4.1.18.).
Set the ICE Agent's ICE servers list to iceServers.

As defined in [RFC9429] (section 4.1.18.), if a new list of servers replaces the ICE Agent's existing ICE servers list, no action will be taken until the next gathering phase. If a script wants this to happen immediately, it should do an ICE restart. However, if the ICE candidate pool has a nonzero size, any existing pooled candidates will be discarded, and new candidates will be gathered from the new servers.
Store configuration in the [[Configuration]] internal slot.

To validate an ICE server URL url, run the following steps:

Parse the url using the generic URI syntax defined in [RFC3986] and obtain the scheme name. If the parsing based on the syntax defined in [RFC3986] fails, throw a SyntaxError. If the scheme name is not implemented by the browser throw a NotSupportedError. If scheme name is turn or turns, and parsing the url using the syntax defined in [RFC7065] fails, throw a SyntaxError. If scheme name is stun or stuns, and parsing the url using the syntax defined in [RFC7064] fails, throw a SyntaxError.
Let parsedURL be the result of parsing url.
If any of the following conditions apply, then throw a "SyntaxError" DOMException:
- parsedURL is failure
- parsedURL's scheme is neither "stun", "stuns", "turn", nor "turns"
- parsedURL does not have an opaque path
- parsedURL's opaque path contains one or more "/" or "@"
- parsedURL's fragment is non-null
- parsedURL's scheme is "stun" or "stuns", and parsedURL's query is non-null
If parsedURL's scheme is not implemented by the user agent, then throw a NotSupportedError.
Let hostAndPortURL be result of parsing the concatenation of "https://" and parsedURL's path.
If hostAndPortURL is failure, then throw a "SyntaxError" DOMException.
If hostAndPortURL's path, username, or password is non-null, then throw a "SyntaxError" DOMException.
Note
For "stun" and "stuns" schemes, this validates [RFC7064] section 3.1.
For "turn" and "turns" schemes, this and the steps below validate [RFC7065] section 3.1.
If parsedURL's query is non-null and if parsedURL's query is different from either "transport=udp" or "transport=tcp", throw a "SyntaxError" DOMException.
If ~~scheme name~~parsedURL's' scheme is ~~turn~~"turn" ~~or turns~~or "turns", and either of server.username or server.credential ~~are omitted~~do not exist, then throw an InvalidAccessError.
If scheme name is turn or turns, and server.credentialType is "password", and server.credential is not a DOMString, then throw an InvalidAccessError.

The RTCPeerConnection interface presented in this section is extended by several partial interfaces throughout this specification. Notably, the RTP Media API section, which adds the APIs to send and receive MediaStreamTrack objects.

[Exposed=Window]
interface RTCPeerConnection : EventTarget  {
  constructor(optional RTCConfiguration configuration = {});
  Promise<RTCSessionDescriptionInit> createOffer(optional RTCOfferOptions options = {});
  Promise<RTCSessionDescriptionInit> createAnswer(optional RTCAnswerOptions options = {});
  Promise<undefined> setLocalDescription(optional RTCLocalSessionDescriptionInit description = {});
  readonly attribute RTCSessionDescription? localDescription;
  readonly attribute RTCSessionDescription? currentLocalDescription;
  readonly attribute RTCSessionDescription? pendingLocalDescription;
  Promise<undefined> setRemoteDescription(RTCSessionDescriptionInit description);
  readonly attribute RTCSessionDescription? remoteDescription;
  readonly attribute RTCSessionDescription? currentRemoteDescription;
  readonly attribute RTCSessionDescription? pendingRemoteDescription;
  Promise<undefined> addIceCandidate(optional RTCIceCandidateInit candidate = {});
  readonly attribute RTCSignalingState signalingState;
  readonly attribute RTCIceGatheringState iceGatheringState;
  readonly attribute RTCIceConnectionState iceConnectionState;
  readonly attribute RTCPeerConnectionState connectionState;
  readonly attribute boolean? canTrickleIceCandidates;
  undefined restartIce();
  RTCConfiguration getConfiguration();
  undefined setConfiguration(optional RTCConfiguration configuration = {});
  undefined close();
  attribute EventHandler onnegotiationneeded;
  attribute EventHandler onicecandidate;
  attribute EventHandler onicecandidateerror;
  attribute EventHandler onsignalingstatechange;
  attribute EventHandler oniceconnectionstatechange;
  attribute EventHandler onicegatheringstatechange;
  attribute EventHandler onconnectionstatechange;

  // Legacy Interface Extensions
  // Supporting the methods in this section is optional.
  // If these methods are supported
  // they must be implemented as defined
  // in section "Legacy Interface Extensions"
  Promise<undefined> createOffer(RTCSessionDescriptionCallback successCallback,
                            RTCPeerConnectionErrorCallback failureCallback,
                            optional RTCOfferOptions options = {});
  Promise<undefined> setLocalDescription(RTCLocalSessionDescriptionInit description,
                                    VoidFunction successCallback,
                                    RTCPeerConnectionErrorCallback failureCallback);
  Promise<undefined> createAnswer(RTCSessionDescriptionCallback successCallback,
                             RTCPeerConnectionErrorCallback failureCallback);
  Promise<undefined> setRemoteDescription(RTCSessionDescriptionInit description,
                                     VoidFunction successCallback,
                                     RTCPeerConnectionErrorCallback failureCallback);
  Promise<undefined> addIceCandidate(RTCIceCandidateInit candidate,
                                VoidFunction successCallback,
                                RTCPeerConnectionErrorCallback failureCallback);
};

localDescription of type RTCSessionDescription, readonly, nullable

The localDescription attribute MUST return [[PendingLocalDescription]] if it is not null and otherwise it MUST return [[CurrentLocalDescription]].

Note that [[CurrentLocalDescription]].sdp and [[PendingLocalDescription]].sdp need not be string-wise identical to the sdp value passed to the corresponding setLocalDescription call (i.e. SDP may be parsed and reformatted, and ICE candidates may be added).

currentLocalDescription of type RTCSessionDescription, readonly, nullable

The currentLocalDescription attribute MUST return [[CurrentLocalDescription]].

It represents the local description that was successfully negotiated the last time the RTCPeerConnection transitioned into the stable state plus any local candidates that have been generated by the ICE Agent since the offer or answer was created.

pendingLocalDescription of type RTCSessionDescription, readonly, nullable

The pendingLocalDescription attribute MUST return [[PendingLocalDescription]].

It represents a local description that is in the process of being negotiated plus any local candidates that have been generated by the ICE Agent since the offer or answer was created. If the RTCPeerConnection is in the stable state, the value is null.

remoteDescription of type RTCSessionDescription, readonly, nullable

The remoteDescription attribute MUST return [[PendingRemoteDescription]] if it is not null and otherwise it MUST return [[CurrentRemoteDescription]].

Note that [[CurrentRemoteDescription]].sdp and [[PendingRemoteDescription]].sdp need not be string-wise identical to the sdp value passed to the corresponding setRemoteDescription call (i.e. SDP may be parsed and reformatted, and ICE candidates may be added).

currentRemoteDescription of type RTCSessionDescription, readonly, nullable

The currentRemoteDescription attribute MUST return [[CurrentRemoteDescription]].

It represents the last remote description that was successfully negotiated the last time the RTCPeerConnection transitioned into the stable state plus any remote candidates that have been supplied via addIceCandidate() since the offer or answer was created.

pendingRemoteDescription of type RTCSessionDescription, readonly, nullable

The pendingRemoteDescription attribute MUST return [[PendingRemoteDescription]].

It represents a remote description that is in the process of being negotiated, complete with any remote candidates that have been supplied via addIceCandidate() since the offer or answer was created. If the RTCPeerConnection is in the stable state, the value is null.

signalingState of type RTCSignalingState, readonly

The signalingState attribute MUST return the RTCPeerConnection object's [[SignalingState]].

iceGatheringState of type RTCIceGatheringState, readonly

The iceGatheringState attribute MUST return the RTCPeerConnection object's [[IceGatheringState]].

iceConnectionState of type RTCIceConnectionState, readonly

The iceConnectionState attribute MUST return the RTCPeerConnection object's [[IceConnectionState]].

connectionState of type RTCPeerConnectionState, readonly

The connectionState attribute MUST return the RTCPeerConnection object's [[ConnectionState]].

canTrickleIceCandidates of type boolean, readonly, nullable

The canTrickleIceCandidates attribute indicates whether the remote peer is able to accept trickled ICE candidates [RFC8838]. The value is determined based on whether a remote description indicates support for trickle ICE, as defined in [RFC9429] (section 4.1.17.). Prior to the completion of setRemoteDescription, this value is null.

onnegotiationneeded of type EventHandler

The event type of this event handler is negotiationneeded.

onicecandidate of type EventHandler

The event type of this event handler is icecandidate.

onicecandidateerror of type EventHandler

The event type of this event handler is icecandidateerror.

onsignalingstatechange of type EventHandler

The event type of this event handler is signalingstatechange.

oniceconnectionstatechange of type EventHandler

The event type of this event handler is iceconnectionstatechange

onicegatheringstatechange of type EventHandler

The event type of this event handler is icegatheringstatechange.

onconnectionstatechange of type EventHandler

The event type of this event handler is connectionstatechange.

createOffer

The createOffer method generates a blob of SDP that contains an RFC 3264 offer with the supported configurations for the session, including descriptions of the local MediaStreamTracks attached to this RTCPeerConnection, the codec/RTP/RTCP capabilities supported by this implementation, and parameters of the ICE agent and the DTLS connection. The options parameter may be supplied to provide additional control over the offer generated.

If a system has limited resources (e.g. a finite number of decoders), createOffer needs to return an offer that reflects the current state of the system, so that setLocalDescription will succeed when it attempts to acquire those resources. The session descriptions MUST remain usable by setLocalDescription without causing an error until at least the end of the fulfillment callback of the returned promise.

Creating the SDP MUST follow the appropriate process for generating an offer described in [RFC9429], except the user agent MUST treat a stopping transceiver as stopped for the purposes of RFC9429 in this case.

As an offer, the generated SDP will contain the full set of codec/RTP/RTCP capabilities supported or preferred by the session (as opposed to an answer, which will include only a specific negotiated subset to use). In the event createOffer is called after the session is established, createOffer will generate an offer that is compatible with the current session, incorporating any changes that have been made to the session since the last complete offer-answer exchange, such as addition or removal of tracks. If no changes have been made, the offer will include the capabilities of the current local description as well as any additional capabilities that could be negotiated in an updated offer.

The generated SDP will also contain the ICE agent's usernameFragment, password and ICE options (as defined in [RFC5245], Section 14) and may also contain any local candidates that have been gathered by the agent.

The certificates value in configuration for the RTCPeerConnection provides the certificates configured by the application for the RTCPeerConnection. These certificates, along with any default certificates are used to produce a set of certificate fingerprints. These certificate fingerprints are used in the construction of SDP.

The process of generating an SDP exposes a subset of the media capabilities of the underlying system, which provides generally persistent cross-origin information on the device. It thus increases the fingerprinting surface of the application. In privacy-sensitive contexts, browsers can consider mitigations such as generating SDP matching only a common subset of the capabilities.

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

Let connection be the RTCPeerConnection object on which the method was invoked.
If connection.[[IsClosed]] is true, return a promise rejected with a newly created InvalidStateError.
Return the result of chaining the result of creating an offer with connection to connection's operations chain.

To create an offer given connection run the following steps:

If connection.[[SignalingState]] is neither "stable" nor "have-local-offer", return a promise rejected with a newly created InvalidStateError.
Let p be a new promise.
In parallel, begin the in-parallel steps to create an offer given connection and p.
Return p.

The in-parallel steps to create an offer given connection and a promise p are as follows:

If connection was not constructed with a set of certificates, and one has not yet been generated, wait for it to be generated.
Inspect the offerer's system state to determine the currently available resources as necessary for generating the offer, as described in [RFC9429] (section 4.1.8.).
If this inspection failed for any reason, reject p with a newly created OperationError and abort these steps.
Queue a task that runs the final steps to create an offer given connection and p.

The final steps to create an offer given connection and a promise p are as follows:

If connection.[[IsClosed]] is true, then abort these steps.
If connection was modified in such a way that additional inspection of the offerer's system state is necessary, then in parallel begin the in-parallel steps to create an offer again, given connection and p, and abort these steps.

Note
This may be necessary if, for example, createOffer was called when only an audio RTCRtpTransceiver was added to connection, but while performing the in-parallel steps to create an offer, a video RTCRtpTransceiver was added, requiring additional inspection of video system resources.
Given the information that was obtained from previous inspection, the current state of connection and its RTCRtpTransceivers, generate an SDP offer, sdpString, as described in [RFC9429] (section 5.2.).
1. As described in [RFC8843] (Section 7), if bundling is used (see RTCBundlePolicy) an offerer tagged m= section must be selected in order to negotiate a BUNDLE group. The user agent MUST choose the m= section that corresponds to the first non-stopped transceiver in the set of transceivers as the offerer tagged m= section. This allows the remote endpoint to predict which transceiver is the offerer tagged m= section without having to parse the SDP.
2. Let filteredCodecs be the result of applying the following filter on transceiver.[[PreferredCodecs]]. The filtering MUST NOT change the order of the codec preferences:
  1. Let kind be transceiver's [[Receiver]]'s [[ReceiverTrack]]'s kind.
  3. If transceiver.direction is "sendonly" or "sendrecv", exclude any codecs not included in the list of implemented send codecs for kind, using the codec dictionary match algorithm with ignoreLevels set to true.
  4. If transceiver.direction is "recvonly" or "sendrecv", exclude any codecs not included in the list of implemented receive codecs for kind, using the codec dictionary match algorithm with ignoreLevels set to true.
  The codec preferences of a media description's associated transceiver, transceiver, is said to be the value of filteredCodecs if non-empty and said to be unset otherwise.
3. If the length of the [[SendEncodings]] slot of the RTCRtpSender is larger than 1, then for each encoding given in [[SendEncodings]] of the RTCRtpSender, add an a=rid send line to the corresponding media section, and add an a=simulcast:send line giving the RIDs in the same order as given in the encodings field. No RID restrictions are set.
  
  Note
  [RFC8853] section 5.2 specifies that the order of RIDs in the a=simulcast line suggests a proposed order of preference. If the browser decides not to transmit all encodings, one should expect it to stop sending the last encoding in the list first.
Let offer be a newly created RTCSessionDescriptionInit dictionary with its type member initialized to the string "offer" and its sdp member initialized to sdpString.
Set the [[LastCreatedOffer]] internal slot to sdpString.
Resolve p with offer.

createAnswer

The createAnswer method generates an [SDP] answer with the supported configuration for the session that is compatible with the parameters in the remote configuration. Like createOffer, the returned blob of SDP contains descriptions of the local MediaStreamTracks attached to this RTCPeerConnection, the codec/RTP/RTCP options negotiated for this session, and any candidates that have been gathered by the ICE Agent. The options parameter may be supplied to provide additional control over the generated answer.

Like createOffer, the returned description SHOULD reflect the current state of the system. The session descriptions MUST remain usable by setLocalDescription without causing an error until at least the end of the fulfillment callback of the returned promise.

As an answer, the generated SDP will contain a specific codec/RTP/RTCP configuration that, along with the corresponding offer, specifies how the media plane should be established. The generation of the SDP MUST follow the appropriate process for generating an answer described in [RFC9429].

An answer can be marked as provisional, as described in [RFC9429] (section 4.1.10.1.), by setting the type to "pranswer".

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

Let connection be the RTCPeerConnection object on which the method was invoked.
If connection.[[IsClosed]] is true, return a promise rejected with a newly created InvalidStateError.
Return the result of chaining the result of creating an answer with connection to connection's operations chain.

To create an answer given connection run the following steps:

If connection.[[SignalingState]] is neither "have-remote-offer" nor "have-local-pranswer", return a promise rejected with a newly created InvalidStateError.
Let p be a new promise.
In parallel, begin the in-parallel steps to create an answer given connection and p.
Return p.

The in-parallel steps to create an answer given connection and a promise p are as follows:

If connection was not constructed with a set of certificates, and one has not yet been generated, wait for it to be generated.
Inspect the answerer's system state to determine the currently available resources as necessary for generating the answer, as described in [RFC9429] (section 4.1.9.).
If this inspection failed for any reason, reject p with a newly created OperationError and abort these steps.
Queue a task that runs the final steps to create an answer given p.

The final steps to create an answer given a promise p are as follows:

If connection.[[IsClosed]] is true, then abort these steps.
If connection was modified in such a way that additional inspection of the answerer's system state is necessary, then in parallel begin the in-parallel steps to create an answer again given connection and p, and abort these steps.

Note
This may be necessary if, for example, createAnswer was called when an RTCRtpTransceiver's direction was "recvonly", but while performing the in-parallel steps to create an answer, the direction was changed to "sendrecv", requiring additional inspection of video encoding resources.
Given the information that was obtained from previous inspection and the current state of connection and its RTCRtpTransceivers, generate an SDP answer, sdpString, as described in [RFC9429] (section 5.3.).

Candidate Correction 26:Prune createAnswer()'s encodings and SendEncodings in sLD(answer). (PR #2801)
Candidate Correction 27:Ignore comma-separated rid alternatives. (PR #2813)
1. The codec preferences of an m= section's associated transceiver is said to be the value of the RTCRtpTransceiver.[[PreferredCodecs]] with the following filtering applied (or said not to be set if [[PreferredCodecs]] is empty):
  If the direction is "sendrecv", exclude any codecs not included in the intersection of RTCRtpSender.getCapabilities(kind).codecs and RTCRtpReceiver.getCapabilities(kind).codecs.
  If the direction is "sendonly", exclude any codecs not included in RTCRtpSender.getCapabilities(kind).codecs.
  If the direction is "recvonly", exclude any codecs not included in RTCRtpReceiver.getCapabilities(kind).codecs.
  The filtering MUST NOT change the order of the codec preferences.
2. If the length of the [[SendEncodings]] slot of the RTCRtpSender is larger than 1, then for each encoding given in [[SendEncodings]] of the RTCRtpSender, add an a=rid send line to the corresponding media section, and add an a=simulcast:send line giving the RIDs in the same order as given in the encodings field. No RID restrictions are set.
3. Let filteredCodecs be the result of applying the following filter on transceiver.[[PreferredCodecs]]. The filtering MUST NOT change the order of the codec preferences:
  1. Let kind be transceiver's [[Receiver]]'s [[ReceiverTrack]]'s kind.
  2. If transceiver.direction is "sendonly" or "sendrecv", exclude any codecs not included in the list of implemented send codecs for kind, using the codec dictionary match algorithm with ignoreLevels set to true.
  3. If transceiver.direction is "recvonly" or "sendrecv", exclude any codecs not included in the list of implemented receive codecs for kind, using the codec dictionary match algorithm with ignoreLevels set to true.
  The codec preferences of a media description's associated transceiver, transceiver, is said to be the value of filteredCodecs if non-empty and said to be unset otherwise.
4. If this is an answer to an offer to receive simulcast, then for each media section requesting to receive simulcast, run the following steps:
  1. If the a=simulcast attribute contains comma-separated alternatives for RIDs, remove all but the first ones.
  2. If there are any identically named RIDs in the a=simulcast attribute, remove all but the first one. No RID restrictions are set.
  3. Exclude from the media section in the answer any RID not found in the corresponding transceiver's [[Sender]].[[SendEncodings]].
  Note
  When a setRemoteDescription(offer) establishes a sender's proposed envelope, the sender's [[SendEncodings]] is updated in "have-remote-offer", exposing it to rollback. However, once a simulcast envelope has been established for the sender, subsequent pruning of the sender's [[SendEncodings]] happen when this answer is set with setLocalDescription.
Let answer be a newly created RTCSessionDescriptionInit dictionary with its type member initialized to the string "answer" and its sdp member initialized to sdpString.
Set the [[LastCreatedAnswer]] internal slot to sdpString.
Resolve p with answer.

setLocalDescription

The setLocalDescription method instructs the RTCPeerConnection to apply the supplied RTCLocalSessionDescriptionInit as the local description.

This API changes the local media state. In order to successfully handle scenarios where the application wants to offer to change from one media format to a different, incompatible format, the RTCPeerConnection MUST be able to simultaneously support use of both the current and pending local descriptions (e.g. support codecs that exist in both descriptions) until a final answer is received, at which point the RTCPeerConnection can fully adopt the pending local description, or rollback to the current description if the remote side rejected the change.

Passing in a description is optional. If left out, then setLocalDescription will implicitly create an offer or create an answer, as needed. As noted in [RFC9429] (section 5.4.), if a description with SDP is passed in, that SDP is not allowed to have changed from when it was returned from either createOffer or createAnswer.

When the method is invoked, the user agent MUST run the following steps:

Let description be the method's first argument.
Let connection be the RTCPeerConnection object on which the method was invoked.
Let sdp be description.sdp.
Return the result of chaining the following steps to connection's operations chain:
1. Let type be description.type if present, or "offer" if not present and connection.[[SignalingState]] is either "stable", "have-local-offer", or "have-remote-pranswer"; otherwise "answer".
2. If type is "offer", and sdp is not the empty string and not equal to connection.[[LastCreatedOffer]], then return a promise rejected with a newly created InvalidModificationError and abort these steps.
3. If type is "answer" or "pranswer", and sdp is not the empty string and not equal to connection.[[LastCreatedAnswer]], then return a promise rejected with a newly created InvalidModificationError and abort these steps.
4. If sdp is the empty string, and type is "offer", then run the following sub steps:
  1. Set sdp to the value of connection.[[LastCreatedOffer]].
  2. If sdp is the empty string, or if it no longer accurately represents the offerer's system state of connection, then let p be the result of creating an offer with connection, and return the result of reacting to p with a fulfillment step that sets the local session description indicated by its first argument.
5. If sdp is the empty string, and type is "answer" or "pranswer", then run the following sub steps:
  1. Set sdp to the value of connection.[[LastCreatedAnswer]].
  2. If sdp is the empty string, or if it no longer accurately represents the answerer's system state of connection, then let p be the result of creating an answer with connection, and return the result of reacting to p with the following fulfillment steps:
    1. Let answer be the first argument to these fulfillment steps.
    2. Return the result of setting the local session description indicated by {type, answer.sdp}.
6. Return the result of setting the local session description indicated by {type, sdp}.

Note

As noted in [RFC9429] (section 5.9.), calling this method may trigger the ICE candidate gathering process by the ICE Agent.

setRemoteDescription

The setRemoteDescription method instructs the RTCPeerConnection to apply the supplied RTCSessionDescriptionInit as the remote offer or answer. This API changes the local media state.

When the method is invoked, the user agent MUST run the following steps:

Let description be the method's first argument.
Let connection be the RTCPeerConnection object on which the method was invoked.
Return the result of chaining the following steps to connection's operations chain:
1. If description.type is "offer" and is invalid for the current connection.[[SignalingState]] as described in [RFC9429] (section 5.5. and section 5.6.), then run the following sub steps:
  1. Let p be the result of setting the local session description indicated by {type: "rollback"}.
  2. Return the result of reacting to p with a fulfillment step that sets the remote session description description, and abort these steps.
2. Return the result of setting the remote session description description.

addIceCandidate

The addIceCandidate method provides a remote candidate to the ICE Agent. This method can also be used to indicate the end of remote candidates when called with an empty string for the candidate member. The only members of the argument used by this method are candidate, sdpMid, sdpMLineIndex, and usernameFragment; the rest are ignored. When the method is invoked, the user agent MUST run the following steps:

Let candidate be the method's argument.
Let connection be the RTCPeerConnection object on which the method was invoked.
If candidate.candidate is not an empty string and both candidate.sdpMid and candidate.sdpMLineIndex are null, return a promise rejected with a newly created TypeError.
Return the result of chaining the following steps to connection's operations chain:
1. If remoteDescription is null return a promise rejected with a newly created InvalidStateError.
2. If candidate.sdpMid is not null, run the following steps:
  1. If candidate.sdpMid is not equal to the mid of any media description in remoteDescription, return a promise rejected with a newly created OperationError.
3. Else, if candidate.sdpMLineIndex is not null, run the following steps:
  1. If candidate.sdpMLineIndex is equal to or larger than the number of media descriptions in remoteDescription, return a promise rejected with a newly created OperationError.
4. If either candidate.sdpMid or candidate.sdpMLineIndex indicate a media description in remoteDescription whose associated transceiver is stopped, return a promise resolved with undefined.
5. If candidate.usernameFragment is not null, and is not equal to any username fragment present in the corresponding media description of an applied remote description, return a promise rejected with a newly created OperationError.
6. Let p be a new promise.
7. In parallel, if the candidate is not administratively prohibited, add the ICE candidate candidate as described in [RFC9429] (section 4.1.19.). Use candidate.usernameFragment to identify the ICE generation; if usernameFragment is null, process the candidate for the most recent ICE generation.
  
  If candidate.candidate is an empty string, process candidate as an end-of-candidates indication for the corresponding media description and ICE candidate generation. If both candidate.sdpMid and candidate.sdpMLineIndex are null, then this end-of-candidates indication applies to all media descriptions.
  1. If candidate could not be successfully added the user agent MUST queue a task that runs the following steps:
    1. If connection.[[IsClosed]] is true, then abort these steps.
    2. Reject p with a newly created OperationError and abort these steps.
  2. If candidate is applied successfully, or if the candidate was administratively prohibited the user agent MUST queue a task that runs the following steps:
    1. If connection.[[IsClosed]] is true, then abort these steps.
    2. If connection.[[PendingRemoteDescription]] is not null, and represents the ICE generation for which candidate was processed, add candidate to connection.[[PendingRemoteDescription]].sdp.
    3. If connection.[[CurrentRemoteDescription]] is not null, and represents the ICE generation for which candidate was processed, add candidate to connection.[[CurrentRemoteDescription]].sdp.
    4. Resolve p with undefined.
8. Return p.

A candidate is administratively prohibited if the UA has decided not to allow connection attempts to this address.

For privacy reasons, there is no indication to the developer about whether or not an address/port is blocked; it behaves exactly as if there was no response from the address.

The UA MUST prohibit connections to addresses on the [Fetch] block bad port list, and MAY choose to prohibit connections to other addresses.

If the iceTransportPolicy member of the RTCConfiguration is relay, candidates requiring external resolution, such as mDNS candidates and DNS candidates, MUST be prohibited.

Note

Due to WebIDL processing, addIceCandidate(null) is interpreted as a call with the default dictionary present, which, in the above algorithm, indicates end-of-candidates for all media descriptions and ICE candidate generation. This is by design for legacy reasons.

restartIce

The restartIce method tells the RTCPeerConnection that ICE should be restarted. Subsequent calls to createOffer will create descriptions that will restart ICE, as described in section 9.1.1.1 of [RFC5245].

When this method is invoked, the user agent MUST run the following steps:

Let connection be the RTCPeerConnection on which the method was invoked.
Empty connection.[[LocalIceCredentialsToReplace]], and populate it with all ICE credentials (ice-ufrag and ice-pwd as defined in section 15.4 of [RFC5245]) found in connection.[[CurrentLocalDescription]], as well as all ICE credentials found in connection.[[PendingLocalDescription]].
Update the negotiation-needed flag for connection.

getConfiguration

Returns an RTCConfiguration object representing the current configuration of this RTCPeerConnection object.

When this method is called, the user agent MUST return the RTCConfiguration object stored in the [[Configuration]] internal slot.

setConfiguration

The setConfiguration method updates the configuration of this RTCPeerConnection object. This includes changing the configuration of the ICE Agent. As noted in [RFC9429] (section 3.5.1.), when the ICE configuration changes in a way that requires a new gathering phase, an ICE restart is required.

When the setConfiguration method is invoked, the user agent MUST run the following steps:

Let connection be the RTCPeerConnection on which the method was invoked.
If connection.[[IsClosed]] is true, throw an InvalidStateError.
Set the configuration specified by configuration.

close

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

Let connection be the RTCPeerConnection object on which the method was invoked.
close the connection with connection and the value false.

The close the connection algorithm given a connection and a disappear boolean, is as follows:

If connection.[[IsClosed]] is true, abort these steps.
Set connection.[[IsClosed]] to true.
Set connection.[[SignalingState]] to "closed". This does not fire any event.
Let transceivers be the result of executing the CollectTransceivers algorithm. For every RTCRtpTransceiver transceiver in transceivers, run the following steps:
1. If transceiver.[[Stopped]] is true, abort these sub steps.
2. Stop the RTCRtpTransceiver with transceiver and disappear.
Set the [[ReadyState]] slot of each of connection's RTCDataChannels to "closed".

Note
The RTCDataChannels will be closed abruptly and the closing procedure will not be invoked.
If connection.[[SctpTransport]] is not null, tear down the underlying SCTP association by sending an SCTP ABORT chunk and set the [[SctpTransportState]] to "closed".
Set the [[DtlsTransportState]] slot of each of connection's RTCDtlsTransports to "closed".
Destroy connection's ICE Agent, abruptly ending any active ICE processing and releasing any relevant resources (e.g. TURN permissions).
Set the [[IceTransportState]] slot of each of connection's RTCIceTransports to "closed".
Set connection.[[IceConnectionState]] to "closed". This does not fire any event.
Set connection.[[ConnectionState]] to "closed". This does not fire any event.

Note

The IDL definition of these methods are documented in the main definition of the RTCPeerConnection interface since overloaded functions are not allowed to be defined in partial interfaces.

Supporting the methods in this section is optional. However, if these methods are supported it is mandatory to implement according to what is specified here.

Note

The addStream method that used to exist on RTCPeerConnection is easy to polyfill as:

RTCPeerConnection.prototype.addStream = function(stream) {
  stream.getTracks().forEach((track) => this.addTrack(track, stream));
};

createOffer

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

Let successCallback be the method's first argument.
Let failureCallback be the callback indicated by the method's second argument.
Let options be the callback indicated by the method's third argument.
Run the steps specified by RTCPeerConnection's createOffer() method with options as the sole argument, and let p be the resulting promise.
Upon fulfillment of p with value offer, invoke successCallback with offer as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with undefined.

setLocalDescription

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

Let description be the method's first argument.
Let successCallback be the callback indicated by the method's second argument.
Let failureCallback be the callback indicated by the method's third argument.
Run the steps specified by RTCPeerConnection's setLocalDescription method with description as the sole argument, and let p be the resulting promise.
Upon fulfillment of p, invoke successCallback with undefined as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with undefined.

createAnswer

Note

The legacy createAnswer method does not take an RTCAnswerOptions parameter, since no known legacy createAnswer implementation ever supported it.

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

Let successCallback be the method's first argument.
Let failureCallback be the callback indicated by the method's second argument.
Run the steps specified by RTCPeerConnection's createAnswer() method with no arguments, and let p be the resulting promise.
Upon fulfillment of p with value answer, invoke successCallback with answer as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with undefined.

setRemoteDescription

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

Let description be the method's first argument.
Let successCallback be the callback indicated by the method's second argument.
Let failureCallback be the callback indicated by the method's third argument.
Run the steps specified by RTCPeerConnection's setRemoteDescription method with description as the sole argument, and let p be the resulting promise.
Upon fulfillment of p, invoke successCallback with undefined as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with undefined.

addIceCandidate

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

Let candidate be the method's first argument.
Let successCallback be the callback indicated by the method's second argument.
Let failureCallback be the callback indicated by the method's third argument.
Run the steps specified by RTCPeerConnection's addIceCandidate() method with candidate as the sole argument, and let p be the resulting promise.
Upon fulfillment of p, invoke successCallback with undefined as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with undefined.

These callbacks are only used on the legacy APIs.

callback RTCSessionDescriptionCallback = undefined (RTCSessionDescriptionInit description);

description of type RTCSessionDescriptionInit: The object containing the SDP [SDP].

This section describes a set of legacy extensions that may be used to influence how an offer is created, in addition to the media added to the RTCPeerConnection. Developers are encouraged to use the RTCRtpTransceiver API instead.

When createOffer is called with any of the legacy options specified in this section, run the followings steps instead of the regular createOffer steps:

Let options be the methods first argument.
Let connection be the current RTCPeerConnection object.
For each offerToReceive<Kind> member in options with kind, kind, run the following steps:
1. If the value of the dictionary member is false,
  1. For each non-stopped "sendrecv" transceiver of transceiver kind kind, set transceiver.[[Direction]] to "sendonly".
  2. For each non-stopped "recvonly" transceiver of transceiver kind kind, set transceiver.[[Direction]] to "inactive".
  Continue with the next option, if any.
2. If connection has any non-stopped "sendrecv" or "recvonly" transceivers of transceiver kind kind, continue with the next option, if any.
3. Let transceiver be the result of invoking the equivalent of connection.addTransceiver(kind), except that this operation MUST NOT update the negotiation-needed flag.
4. If transceiver is unset because the previous operation threw an error, abort these steps.
5. Set transceiver.[[Direction]] to "recvonly".
Run the steps specified by createOffer to create the offer.

partial dictionary RTCOfferOptions {
  boolean offerToReceiveAudio;
  boolean offerToReceiveVideo;
};

offerToReceiveAudio of type boolean: This setting provides additional control over the directionality of audio. For example, it can be used to ensure that audio can be received, regardless if audio is sent or not.
offerToReceiveVideo of type boolean: This setting provides additional control over the directionality of video. For example, it can be used to ensure that video can be received, regardless if video is sent or not.

An RTCPeerConnection object MUST not be garbage collected as long as any event can cause an event handler to be triggered on the object. When the object's [[IsClosed]] internal slot is true, no such event handler can be triggered and it is therefore safe to garbage collect the object.

All RTCDataChannel and MediaStreamTrack objects that are connected to an RTCPeerConnection have a strong reference to the RTCPeerConnection object.

All methods that return promises are governed by the standard error handling rules of promises. Methods that do not return promises may throw exceptions to indicate errors.

The RTCSdpType enum describes the type of an RTCSessionDescriptionInit, RTCLocalSessionDescriptionInit, or RTCSessionDescription instance.

enum RTCSdpType {
  "offer",
  "pranswer",
  "answer",
  "rollback"
};

`RTCSdpType` Enumeration description
Enum value	Description
`offer`	An `RTCSdpType` of "`offer`" indicates that a description MUST be treated as an [SDP] offer.
`pranswer`	An `RTCSdpType` of "`pranswer`" indicates that a description MUST be treated as an [SDP] answer, but not a final answer. A description used as an SDP pranswer may be applied as a response to an SDP offer, or an update to a previously sent SDP pranswer.
`answer`	An `RTCSdpType` of "`answer`" indicates that a description MUST be treated as an [SDP] final answer, and the offer-answer exchange MUST be considered complete. A description used as an SDP answer may be applied as a response to an SDP offer or as an update to a previously sent SDP pranswer.
`rollback`	An `RTCSdpType` of "`rollback`" indicates that a description MUST be treated as canceling the current SDP negotiation and moving the SDP [SDP] offer back to what it was in the previous stable state. Note the local or remote SDP descriptions in the previous stable state could be `null` if there has not yet been a successful offer-answer negotiation. An "`answer`" or "`pranswer`" cannot be rolled back.

The RTCSessionDescription class is used by RTCPeerConnection to expose local and remote session descriptions.

[Exposed=Window]
interface RTCSessionDescription {
  constructor(RTCSessionDescriptionInit descriptionInitDict);
  readonly attribute RTCSdpType type;
  readonly attribute DOMString sdp;
  [Default] RTCSessionDescriptionInit toJSON();
};

constructor(): The RTCSessionDescription() constructor takes a dictionary argument, description, whose content is used to initialize the new RTCSessionDescription object. This constructor is deprecated; it exists for legacy compatibility reasons only.

type of type RTCSdpType, readonly: The type of this session description.
sdp of type DOMString, readonly, defaulting to "": The string representation of the SDP [SDP].

toJSON(): When called, run [WEBIDL]'s default toJSON steps.

dictionary RTCSessionDescriptionInit {
  required RTCSdpType type;
  DOMString sdp = "";
};

type of type RTCSdpType, required: The type of this session description.
sdp of type DOMString: The string representation of the SDP [SDP]; if type is "rollback", this member is unused.

dictionary RTCLocalSessionDescriptionInit {
  RTCSdpType type;
  DOMString sdp = "";
};

type of type RTCSdpType: The type of this description. If not present, then setLocalDescription will infer the type based on the RTCPeerConnection's [[SignalingState]].
sdp of type DOMString: The string representation of the SDP [SDP]; if type is "rollback", this member is unused.

Many changes to state of an RTCPeerConnection will require communication with the remote side via the signaling channel, in order to have the desired effect. The app can be kept informed as to when it needs to do signaling, by listening to the negotiationneeded event. This event is fired according to the state of the connection's negotiation-needed flag, represented by a [[NegotiationNeeded]] internal slot.

This section is non-normative.

If an operation is performed on an RTCPeerConnection that requires signaling, the connection will be marked as needing negotiation. Examples of such operations include adding or stopping an RTCRtpTransceiver, or adding the first RTCDataChannel.

Internal changes within the implementation can also result in the connection being marked as needing negotiation.

Note that the exact procedures for updating the negotiation-needed flag are specified below.

This section is non-normative.

The negotiation-needed flag is cleared when a session description of type "answer" is set successfully, and the supplied description matches the state of the RTCRtpTransceivers and RTCDataChannels that currently exist on the RTCPeerConnection. Specifically, this means that all non-stopped transceivers have an associated section in the local description with matching properties, and, if any data channels have been created, a data section exists in the local description.

Note that the exact procedures for updating the negotiation-needed flag are specified below.

The process below occurs where referenced elsewhere in this document. It also may occur as a result of internal changes within the implementation that affect negotiation. If such changes occur, the user agent MUST update the negotiation-needed flag.

To update the negotiation-needed flag for connection, run the following steps:

If the length of connection.[[Operations]] is not 0, then set connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to true, and abort these steps.
Queue a task to run the following steps:
1. If connection.[[IsClosed]] is true, abort these steps.
2. If the length of connection.[[Operations]] is not 0, then set connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to true, and abort these steps.
3. If connection.[[SignalingState]] is not "stable", abort these steps.
  
  Note
  The negotiation-needed flag will be updated once the state transitions to "stable", as part of the steps for setting a session description.
4. If the result of checking if negotiation is needed is false, clear the negotiation-needed flag by setting connection.[[NegotiationNeeded]] to false, and abort these steps.
5. If connection.[[NegotiationNeeded]] is already true, abort these steps.
6. Set connection.[[NegotiationNeeded]] to true.
7. Fire an event named negotiationneeded at connection.
Note

The task queueing prevents negotiationneeded from firing prematurely, in the common situation where multiple modifications to connection are being made at once.

Additionally, we avoid racing with negotiation methods by only firing negotiationneeded when the operations chain is empty.

To check if negotiation is needed for connection, perform the following checks:

If any implementation-specific negotiation is required, as described at the start of this section, return true.
If connection.[[LocalIceCredentialsToReplace]] is not empty, return true.
Let description be connection.[[CurrentLocalDescription]].
If connection has created any RTCDataChannels, and no m= section in description has been negotiated yet for data, return true.
For each transceiver in connection's set of transceivers, perform the following checks:
1. If transceiver.[[Stopping]] is true and transceiver.[[Stopped]] is false, return true.
2. If transceiver isn't stopped and isn't yet associated with an m= section in description, return true.
3. If transceiver isn't stopped and is associated with an m= section in description then perform the following checks:
  1. If transceiver.[[Direction]] is "sendrecv" or "sendonly", and the associated m= section in description either doesn't contain a single a=msid line, or the number of MSIDs from the a=msid lines in this m= section, or the MSID values themselves, differ from what is in transceiver.sender.[[AssociatedMediaStreamIds]], return true.
  2. If description is of type "offer", and the direction of the associated m= section in neither connection.[[CurrentLocalDescription]] nor connection.[[CurrentRemoteDescription]] matches transceiver.[[Direction]], return true. In this step, when the direction is compared with a direction found in [[CurrentRemoteDescription]], the description's direction must be reversed to represent the peer's point of view.
  3. If description is of type "answer", and the direction of the associated m= section in the description does not match transceiver.[[Direction]] intersected with the offered direction (as described in [RFC9429] (section 5.3.1.)), return true.
4. If transceiver is stopped and is associated with an m= section, but the associated m= section is not yet rejected in connection.[[CurrentLocalDescription]] or connection.[[CurrentRemoteDescription]], return true.
If all the preceding checks were performed and true was not returned, nothing remains to be negotiated; return false.

This interface describes an ICE candidate, described in [RFC5245] Section 2. Other than candidate, sdpMid, sdpMLineIndex, and usernameFragment, the remaining attributes are derived from parsing the candidate member in candidateInitDict, if it is well formed.

Candidate Addition 16:Add RTCIceCandidate.relayProtocol (PR #2763)

Candidate Addition 23:Add RTCIceCandidate.url (PR #2773)

[Exposed=Window]
interface RTCIceCandidate {
  constructor(optional RTCIceCandidateInit candidateInitDict = {});
  readonly attribute DOMString candidate;
  readonly attribute DOMString? sdpMid;
  readonly attribute unsigned short? sdpMLineIndex;
  readonly attribute DOMString? foundation;
  readonly attribute RTCIceComponent? component;
  readonly attribute unsigned long? priority;
  readonly attribute DOMString? address;
  readonly attribute RTCIceProtocol? protocol;
  readonly attribute unsigned short? port;
  readonly attribute RTCIceCandidateType? type;
  readonly attribute RTCIceTcpCandidateType? tcpType;
  readonly attribute DOMString? relatedAddress;
  readonly attribute unsigned short? relatedPort;
  readonly attribute DOMString? usernameFragment;
  readonly attribute RTCIceServerTransportProtocol? relayProtocol;
  readonly attribute DOMString? url;
  RTCIceCandidateInit toJSON();
};

constructor()

The RTCIceCandidate() constructor takes a dictionary argument, candidateInitDict, whose content is used to initialize the new RTCIceCandidate object.

When invoked, run the following steps:

If both the sdpMid and sdpMLineIndex members of candidateInitDict are null, throw a TypeError.
Return the result of creating an RTCIceCandidate with candidateInitDict.

To create an RTCIceCandidate with a candidateInitDict dictionary, run the following steps:

Let iceCandidate be a newly created RTCIceCandidate object.
Create internal slots for the following attributes of iceCandidate, initilized to null: foundation, component, priority, address, protocol, port, type, tcpType, relatedAddress, and relatedPort.
Create internal slots for the following attributes of iceCandidate, initilized to their namesakes in candidateInitDict: candidate, sdpMid, sdpMLineIndex, usernameFragment.
Let candidate be the candidate dictionary member of candidateInitDict. If candidate is not an empty string, run the following steps:
1. Parse candidate using the candidate-attribute grammar.
2. If parsing of candidate-attribute has failed, abort these steps.
3. If any field in the parse result represents an invalid value for the corresponding attribute in iceCandidate, abort these steps.
4. Set the corresponding internal slots in iceCandidate to the field values of the parsed result.
Return iceCandidate.

Note

The constructor for RTCIceCandidate only does basic parsing and type checking for the dictionary members in candidateInitDict. Detailed validation on the well-formedness of candidate, sdpMid, sdpMLineIndex, usernameFragment with the corresponding session description is done when passing the RTCIceCandidate object to addIceCandidate().

To maintain backward compatibility, any error on parsing the candidate attribute is ignored. In such case, the candidate attribute holds the raw candidate string given in candidateInitDict, but derivative attributes such as foundation, priority, etc are set to null.

Most attributes below are defined in section 15.1 of [RFC5245].

candidate of type DOMString, readonly

This carries the candidate-attribute as defined in section 15.1 of [RFC5245]. If this RTCIceCandidate represents an end-of-candidates indication or a peer reflexive remote candidate, candidate is an empty string.

sdpMid of type DOMString, readonly, nullable

If not null, this contains the media stream "identification-tag" defined in [RFC5888] for the media component this candidate is associated with.

sdpMLineIndex of type unsigned short, readonly, nullable

If not null, this indicates the index (starting at zero) of the media description in the SDP this candidate is associated with.

foundation of type DOMString, readonly, nullable

A unique identifier that allows ICE to correlate candidates that appear on multiple RTCIceTransports.

component of type RTCIceComponent, readonly, nullable

The assigned network component of the candidate ("rtp" or "rtcp"). This corresponds to the component-id field in candidate-attribute, decoded to the string representation as defined in RTCIceComponent.

priority of type unsigned long, readonly, nullable

The assigned priority of the candidate.

address of type DOMString, readonly, nullable

The address of the candidate, allowing for IPv4 addresses, IPv6 addresses, and fully qualified domain names (FQDNs). This corresponds to the connection-address field in candidate-attribute.

Remote candidates may be exposed, for instance via [[SelectedCandidatePair]].remote. By default, the user agent MUST leave the address attribute as null for any exposed remote candidate. Once a RTCPeerConnection instance learns on an address by the web application using addIceCandidate, the user agent can expose the address attribute value in any RTCIceCandidate of the RTCPeerConnection instance representing a remote candidate with that newly learnt address.

Note

The addresses exposed in candidates gathered via ICE and made visibile to the application in RTCIceCandidate instances can reveal more information about the device and the user (e.g. location, local network topology) than the user might have expected in a non-WebRTC enabled browser.

These addresses are always exposed to the application, and potentially exposed to the communicating party, and can be exposed without any specific user consent (e.g. for peer connections used with data channels, or to receive media only).

These addresses can also be used as temporary or persistent cross-origin states, and thus contribute to the fingerprinting surface of the device.

Applications can avoid exposing addresses to the communicating party, either temporarily or permanently, by forcing the ICE Agent to report only relay candidates via the iceTransportPolicy member of RTCConfiguration.

To limit the addresses exposed to the application itself, browsers can offer their users different policies regarding sharing local addresses, as defined in [RFC8828].

protocol of type RTCIceProtocol, readonly, nullable

The protocol of the candidate ("udp"/"tcp"). This corresponds to the transport field in candidate-attribute.

port of type unsigned short, readonly, nullable

The port of the candidate.

type of type RTCIceCandidateType, readonly, nullable

The type of the candidate. This corresponds to the candidate-types field in candidate-attribute.

tcpType of type RTCIceTcpCandidateType, readonly, nullable

If protocol is "tcp", tcpType represents the type of TCP candidate. Otherwise, tcpType is null. This corresponds to the tcp-type field in candidate-attribute.

relatedAddress of type DOMString, readonly, nullable

For a candidate that is derived from another, such as a relay or reflexive candidate, the relatedAddress is the IP address of the candidate that it is derived from. For host candidates, the relatedAddress is null. This corresponds to the rel-address field in candidate-attribute.

relatedPort of type unsigned short, readonly, nullable

For a candidate that is derived from another, such as a relay or reflexive candidate, the relatedPort is the port of the candidate that it is derived from. For host candidates, the relatedPort is null. This corresponds to the rel-port field in candidate-attribute.

usernameFragment of type DOMString, readonly, nullable

This carries the ufrag as defined in section 15.4 of [RFC5245].

Candidate Addition 16:Add RTCIceCandidate.relayProtocol (PR #2763)

relayProtocol of type RTCIceServerTransportProtocol, readonly, nullable

For local candidates of type "relay" this is the protocol used by the endpoint to communicate with the TURN server. For all other candidates it is null.

Candidate Addition 23:Add RTCIceCandidate.url (PR #2773)

url of type DOMString, readonly, nullable

For local candidates of type "srflx" or type "relay" this is the URL of the ICE server from which the candidate was obtained. For all other candidates it is null.

toJSON()

To invoke the toJSON() operation of the RTCIceCandidate interface, run the following steps:

Let json be a new RTCIceCandidateInit dictionary.
For each attribute identifier attr in «candidate, sdpMid, sdpMLineIndex, usernameFragment»:
1. Let value be the result of getting the underlying value of the attribute identified by attr, given this RTCIceCandidate object.
2. Set json[attr] to value.
Return json.

dictionary RTCIceCandidateInit {
  DOMString candidate = "";
  DOMString? sdpMid = null;
  unsigned short? sdpMLineIndex = null;
  DOMString? usernameFragment = null;
};

candidate of type DOMString, defaulting to "": This carries the candidate-attribute as defined in section 15.1 of [RFC5245]. If this represents an end-of-candidates indication, candidate is an empty string.
sdpMid of type DOMString, nullable, defaulting to null: If not null, this contains the media stream "identification-tag" defined in [RFC5888] for the media component this candidate is associated with.
sdpMLineIndex of type unsigned short, nullable, defaulting to null: If not null, this indicates the index (starting at zero) of the media description in the SDP this candidate is associated with.
usernameFragment of type DOMString, nullable, defaulting to null: If not null, this carries the ufrag as defined in section 15.4 of [RFC5245].

The candidate-attribute grammar is used to parse the candidate member of candidateInitDict in the RTCIceCandidate() constructor.

The primary grammar for candidate-attribute is defined in section 15.1 of [RFC5245]. In addition, the browser MUST support the grammar extension for ICE TCP as defined in section 4.5 of [RFC6544].

The browser MAY support other grammar extensions for candidate-attribute as defined in other RFCs.

The RTCIceProtocol represents the protocol of the ICE candidate.

enum RTCIceProtocol {
  "udp",
  "tcp"
};

`RTCIceProtocol` Enumeration description
Enum value	Description
`udp`	A UDP candidate, as described in [RFC5245].
`tcp`	A TCP candidate, as described in [RFC6544].

The RTCIceTcpCandidateType represents the type of the ICE TCP candidate, as defined in [RFC6544].

enum RTCIceTcpCandidateType {
  "active",
  "passive",
  "so"
};

`RTCIceTcpCandidateType` Enumeration description
Enum value	Description
`active`	An "`active`" TCP candidate is one for which the transport will attempt to open an outbound connection but will not receive incoming connection requests.
`passive`	A "`passive`" TCP candidate is one for which the transport will receive incoming connection attempts but not attempt a connection.
`so`	An "`so`" candidate is one for which the transport will attempt to open a connection simultaneously with its peer.

Note

The user agent will typically only gather active ICE TCP candidates.

The RTCIceCandidateType represents the type of the ICE candidate, as defined in [RFC5245] section 15.1.

enum RTCIceCandidateType {
  "host",
  "srflx",
  "prflx",
  "relay"
};

`RTCIceCandidateType` Enumeration description
Enum value	Description
`host`	A host candidate, as defined in Section 4.1.1.1 of [RFC5245].
`srflx`	A server reflexive candidate, as defined in Section 4.1.1.2 of [RFC5245].
`prflx`	A peer reflexive candidate, as defined in Section 4.1.1.2 of [RFC5245].
`relay`	A relay candidate, as defined in Section 7.1.3.2.1 of [RFC5245].

Candidate Addition 16:Add RTCIceCandidate.relayProtocol (PR #2763)

The RTCIceServerTransportProtocol represents the type of the transport protocol used between the client and the server, as defined in [RFC8656] section 3.1.

enum RTCIceServerTransportProtocol {
  "udp",
  "tcp",
  "tls",
};

`RTCIceServerTransportProtocol` Enumeration description
Enum value	Description
`udp`	The TURN client is using UDP as transport to the server.
`tcp`	The TURN client is using TCP as transport to the server.
`tls`	The TURN client is using TLS as transport to the server.

The icecandidate event of the RTCPeerConnection uses the RTCPeerConnectionIceEvent interface.

When firing an RTCPeerConnectionIceEvent event that contains an RTCIceCandidate object, it MUST include values for both sdpMid and sdpMLineIndex. If the RTCIceCandidate is of type "srflx" or type "relay", the url property of the event MUST be set to the URL of the ICE server from which the candidate was obtained.

Note

The icecandidate event is used for three different types of indications:

A candidate has been gathered. The candidate member of the event will be populated normally. It should be signaled to the remote peer and passed into addIceCandidate.
An RTCIceTransport has finished gathering a generation of candidates, and is providing an end-of-candidates indication as defined by Section 8.2 of [RFC8838]. This is indicated by candidate.candidate being set to an empty string. The candidate object should be signaled to the remote peer and passed into addIceCandidate like a typical ICE candidate, in order to provide the end-of-candidates indication to the remote peer.
All RTCIceTransports have finished gathering candidates, and the RTCPeerConnection's RTCIceGatheringState has transitioned to "complete". This is indicated by the candidate member of the event being set to null. This only exists for backwards compatibility, and this event does not need to be signaled to the remote peer. It's equivalent to an icegatheringstatechange event with the "complete" state.

[Exposed=Window]
interface RTCPeerConnectionIceEvent : Event {
  constructor(DOMString type, optional RTCPeerConnectionIceEventInit eventInitDict = {});
  readonly attribute RTCIceCandidate? candidate;
  readonly attribute DOMString? url;
};

RTCPeerConnectionIceEvent.constructor()

candidate of type RTCIceCandidate, readonly, nullable

The candidate attribute is the RTCIceCandidate object with the new ICE candidate that caused the event.

This attribute is set to null when an event is generated to indicate the end of candidate gathering.

Note

Even where there are multiple media components, only one event containing a null candidate is fired.

url of type DOMString, readonly, nullable

The url attribute is the STUN or TURN URL that identifies the STUN or TURN server used to gather this candidate. If the candidate was not gathered from a STUN or TURN server, this parameter will be set to null.

Candidate Correction 23:Mark RTCPeerConnectionIceEvent.url as deprecated (PR #2773)

This attribute is deprecated; it exists for legacy compatibility reasons only. Prefer the candidate url.

dictionary RTCPeerConnectionIceEventInit : EventInit {
  RTCIceCandidate? candidate;
  DOMString? url;
};

candidate of type RTCIceCandidate, nullable: See the candidate attribute of the RTCPeerConnectionIceEvent interface.
url of type DOMString, nullable: The url attribute is the STUN or TURN URL that identifies the STUN or TURN server used to gather this candidate.

The icecandidateerror event of the RTCPeerConnection uses the RTCPeerConnectionIceErrorEvent interface.

[Exposed=Window]
interface RTCPeerConnectionIceErrorEvent : Event {
  constructor(DOMString type, RTCPeerConnectionIceErrorEventInit eventInitDict);
  readonly attribute DOMString? address;
  readonly attribute unsigned short? port;
  readonly attribute DOMString url;
  readonly attribute unsigned short errorCode;
  readonly attribute USVString errorText;
};

RTCPeerConnectionIceErrorEvent.constructor()

address of type DOMString, readonly, nullable

The address attribute is the local IP address used to communicate with the STUN or TURN server.

On a multihomed system, multiple interfaces may be used to contact the server, and this attribute allows the application to figure out on which one the failure occurred.

If the local IP address value is not already exposed as part of a local candidate, the address attribute will be set to null.

port of type unsigned short, readonly, nullable

The port attribute is the port used to communicate with the STUN or TURN server.

If the address attribute is null, the port attribute is also set to null.

url of type DOMString, readonly

The url attribute is the STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred.

errorCode of type unsigned short, readonly

The errorCode attribute is the numeric STUN error code returned by the STUN or TURN server [STUN-PARAMETERS].

If no host candidate can reach the server, errorCode will be set to the value 701 which is outside the STUN error code range. This error is only fired once per server URL while in the RTCIceGatheringState of "gathering".

errorText of type USVString, readonly

The errorText attribute is the STUN reason text returned by the STUN or TURN server [STUN-PARAMETERS].

If the server could not be reached, errorText will be set to an implementation-specific value providing details about the error.

dictionary RTCPeerConnectionIceErrorEventInit : EventInit {
  DOMString? address;
  unsigned short? port;
  DOMString url;
  required unsigned short errorCode;
  USVString errorText;
};

address of type DOMString, nullable: The local address used to communicate with the STUN or TURN server, or null.
port of type unsigned short, nullable: The local port used to communicate with the STUN or TURN server, or null.
url of type DOMString: The STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred.
errorCode of type unsigned short, required: The numeric STUN error code returned by the STUN or TURN server.
errorText of type USVString: The STUN reason text returned by the STUN or TURN server.

The certificates that RTCPeerConnection instances use to authenticate with peers use the RTCCertificate interface. These objects can be explicitly generated by applications using the generateCertificate method and can be provided in the RTCConfiguration when constructing a new RTCPeerConnection instance.

The explicit certificate management functions provided here are optional. If an application does not provide the certificates configuration option when constructing an RTCPeerConnection a new set of certificates MUST be generated by the user agent. That set MUST include an ECDSA certificate with a private key on the P-256 curve and a signature with a SHA-256 hash.

partial interface RTCPeerConnection {
  static Promise<RTCCertificate>
      generateCertificate(AlgorithmIdentifier keygenAlgorithm);
};

generateCertificate, static

The generateCertificate function causes the user agent to create an X.509 certificate [X509V3] and corresponding private key. A handle to information is provided in the form of the RTCCertificate interface. The returned RTCCertificate can be used to control the certificate that is offered in the DTLS sessions established by RTCPeerConnection.

The keygenAlgorithm argument is used to control how the private key associated with the certificate is generated. The keygenAlgorithm argument uses the WebCrypto [WebCryptoAPI] AlgorithmIdentifier type.

The following values MUST be supported by a user agent: { name: "RSASSA-PKCS1-v1_5", modulusLength: 2048, publicExponent: new Uint8Array([1, 0, 1]), hash: "SHA-256" }, and { name: "ECDSA", namedCurve: "P-256" }.

Note

It is expected that a user agent will have a small or even fixed set of values that it will accept.

The certificate produced by this process also contains a signature. The validity of this signature is only relevant for compatibility reasons. Only the public key and the resulting certificate fingerprint are used by RTCPeerConnection, but it is more likely that a certificate will be accepted if the certificate is well formed. The browser selects the algorithm used to sign the certificate; a browser SHOULD select SHA-256 [FIPS-180-4] if a hash algorithm is needed.

The resulting certificate MUST NOT include information that can be linked to a user or user agent. Randomized values for distinguished name and serial number SHOULD be used.

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

Let keygenAlgorithm be the first argument to generateCertificate.
Let expires be a value of 2592000000 (30*24*60*60*1000)

Note
This means the certificate will by default expire in 30 days from the time of the generateCertificate call.
If keygenAlgorithm is an object, run the following steps:
1. Let certificateExpiration be the result of converting the ECMAScript object represented by keygenAlgorithm to an RTCCertificateExpiration dictionary.
2. If the conversion fails with an error, return a promise that is rejected with error.
3. If certificateExpiration.expires is not undefined, set expires to certificateExpiration.expires.
4. If expires is greater than 31536000000, set expires to 31536000000.
  
  Note
  This means the certificate cannot be valid for longer than 365 days from the time of the generateCertificate call.
  
  A user agent MAY further cap the value of expires.
Let normalizedKeygenAlgorithm be the result of normalizing an algorithm with an operation name of generateKey and a supportedAlgorithms value specific to production of certificates for RTCPeerConnection.
If the above normalization step fails with an error, return a promise that is rejected with error.
If the normalizedKeygenAlgorithm parameter identifies an algorithm that the user agent cannot or will not use to generate a certificate for RTCPeerConnection, return a promise that is rejected with a DOMException of type NotSupportedError. In particular, normalizedKeygenAlgorithm MUST be an asymmetric algorithm that can be used to produce a signature used to authenticate DTLS connections.
Let p be a new promise.
Run the following steps in parallel:
1. Perform the generate key operation specified by normalizedKeygenAlgorithm using keygenAlgorithm.
2. Let generatedKeyingMaterial and generatedKeyCertificate be the private keying material and certificate generated by the above step.
3. Let certificate be a new RTCCertificate object.
4. Set certificate.[[Expires]] to the current time plus expires value.
5. Set certificate.[[Origin]] to the relevant settings object's origin.
6. Store the generatedKeyingMaterial in a secure module, and let handle be a reference identifier to it.
7. Set certificate.[[KeyingMaterialHandle]] to handle.
8. Set certificate.[[Certificate]] to generatedCertificate.
9. Queue a global task on the networking task source given the current realm's global object as global to resolve p with certificate.
Return p.

RTCCertificateExpiration is used to set an expiration date on certificates generated by generateCertificate.

dictionary RTCCertificateExpiration {
  [EnforceRange] unsigned long long expires;
};

expires, of type unsigned long long: An optional expires attribute MAY be added to the definition of the algorithm that is passed to generateCertificate. If this parameter is present it indicates the maximum time in milliseconds that the RTCCertificate is valid for, measured from the time the certificate is created.

The RTCCertificate interface represents a certificate used to authenticate WebRTC communications. In addition to the visible properties, internal slots contain a handle to the generated private keying materal ([[KeyingMaterialHandle]]), a certificate ([[Certificate]]) that RTCPeerConnection uses to authenticate with a peer, and the origin ([[Origin]]) that created the object.

[Exposed=Window, Serializable]
interface RTCCertificate {
  readonly attribute EpochTimeStamp expires;
  sequence<RTCDtlsFingerprint> getFingerprints();
};

expires of type EpochTimeStamp, readonly

The expires attribute indicates the date and time in milliseconds relative to 1970-01-01T00:00:00Z after which the certificate will be considered invalid by the browser. After this time, attempts to construct an RTCPeerConnection using this certificate fail.

Note that this value might not be reflected in a notAfter parameter in the certificate itself.

getFingerprints: Returns the list of certificate fingerprints, one of which is computed with the digest algorithm used in the certificate signature.

For the purposes of this API, the [[Certificate]] slot contains unstructured binary data. No mechanism is provided for applications to access the [[KeyingMaterialHandle]] internal slot or the keying material it references. Implementations MUST support applications storing and retrieving RTCCertificate objects from persistent storage, in a manner that also preserves the keying material referenced by [[KeyingMaterialHandle]]. Implementations SHOULD store the sensitive keying material in a secure module safe from same-process memory attacks. This allows the private key to be stored and used, but not easily read using a memory attack.

RTCCertificate objects are serializable objects [HTML]. Their serialization steps, given value and serialized, are:

Set serialized.[[Expires]] to the value of value.expires attribute.
Set serialized.[[Certificate]] to a copy of the unstructured binary data in value.[[Certificate]].
Set serialized.[[Origin]] to a copy of the unstructured binary data in value.[[Origin]].
Set serialized.[[KeyingMaterialHandle]] to a serialization of the handle in value.[[KeyingMaterialHandle]] (not the private keying material itself).

Their deserialization steps, given serialized and value, are:

Initialize value.expires attribute to contain serialized.[[Expires]].
Set value.[[Certificate]] to a copy of serialized.[[Certificate]]
Set value.[[Origin]] to a copy of serialized.[[Origin]]
Set value.[[KeyingMaterialHandle]] to the private keying material handle resulting from deserializing serialized.[[KeyingMaterialHandle]]

Note

Supporting structured cloning in this manner allows RTCCertificate instances to be persisted to stores. It also allows instances to be passed to other origins using APIs like postMessage(message, options) [html]. However, the object cannot be used by any other origin than the one that originally created it.

The RTP media API lets a web application send and receive MediaStreamTracks over a peer-to-peer connection. Tracks, when added to an RTCPeerConnection, result in signaling; when this signaling is forwarded to a remote peer, it causes corresponding tracks to be created on the remote side.

Note

There is not an exact 1:1 correspondence between tracks sent by one RTCPeerConnection and received by the other. For one, IDs of tracks sent have no mapping to the IDs of tracks received. Also, replaceTrack changes the track sent by an RTCRtpSender without creating a new track on the receiver side; the corresponding RTCRtpReceiver will only have a single track, potentially representing multiple sources of media stitched together. Both addTransceiver and replaceTrack can be used to cause the same track to be sent multiple times, which will be observed on the receiver side as multiple receivers each with its own separate track. Thus it's more accurate to think of a 1:1 relationship between an RTCRtpSender on one side and an RTCRtpReceiver's track on the other side, matching senders and receivers using the RTCRtpTransceiver's mid if necessary.

When sending media, the sender may need to rescale or resample the media to meet various requirements, including the envelope negotiated by SDP, alignment restrictions of the encoder, or even CPU overuse detection or bandwidth estimation.

Following the rules in [RFC9429] (section 3.6.), the video MAY be downscaled. The media MUST NOT be upscaled to create fake data that did not occur in the input source, the media MUST NOT be cropped except as needed to satisfy constraints on pixel counts, and the aspect ratio MUST NOT be changed.

Note

The WebRTC Working Group is seeking implementation feedback on the need and timeline for a more complex handling of this situation. Some possible designs have been discussed in GitHub issue 1283.

Whenever video is rescaled as a result of scaleResolutionDownBy, situations when the resulting width or height is not an integer may occur. The user agent MUST NOT transmit video larger than the integer part of the scaled width and height from scaleResolutionDownBy, except to respect an encoder's minimum resolution. What to transmit if the integer part of the scaled width or height is zero is implementation-defined.

The actual encoding and transmission of MediaStreamTracks is managed through objects called RTCRtpSenders. Similarly, the reception and decoding of MediaStreamTracks is managed through objects called RTCRtpReceivers. Each RTCRtpSender is associated with at most one track, and each track to be received is associated with exactly one RTCRtpReceiver.

The encoding and transmission of each MediaStreamTrack SHOULD be made such that its characteristics (width, height and frameRate for video tracks; sampleSize, sampleRate and channelCount for audio tracks) are to a reasonable degree retained by the track created on the remote side. There are situations when this does not apply, there may for example be resource constraints at either endpoint or in the network or there may be RTCRtpSender settings applied that instruct the implementation to act differently.

An RTCPeerConnection object contains a set of RTCRtpTransceivers, representing the paired senders and receivers with some shared state. This set is initialized to the empty set when the RTCPeerConnection object is created. RTCRtpSenders and RTCRtpReceivers are always created at the same time as an RTCRtpTransceiver, which they will remain attached to for their lifetime. RTCRtpTransceivers are created implicitly when the application attaches a MediaStreamTrack to an RTCPeerConnection via the addTrack() method, or explicitly when the application uses the addTransceiver method. They are also created when a remote description is applied that includes a new media description. Additionally, when a remote description is applied that indicates the remote endpoint has media to send, the relevant MediaStreamTrack and RTCRtpReceiver are surfaced to the application via the track event.

In order for an RTCRtpTransceiver to send and/or receive media with another endpoint this must be negotiated with SDP such that both endpoints have an RTCRtpTransceiver object that is associated with the same media description.

When creating an offer, enough media descriptions will be generated to cover all transceivers on that end. When this offer is set as the local description, any disassociated transceivers get associated with media descriptions in the offer.

When an offer is set as the remote description, any media descriptions in it not yet associated with a transceiver get associated with a new or existing transceiver. In this case, only disassociated transceivers that were created via the addTrack() method may be associated. Disassociated transceivers created via the addTransceiver() method, however, won't get associated even if media descriptions are available in the remote offer. Instead, new transceivers will be created and associated if there aren't enough addTrack()-created transceivers. This sets addTrack()-created and addTransceiver()-created transceivers apart in a critical way that is not observable from inspecting their attributes.

When creating an answer, only media descriptions that were present in the offer may be listed in the answer. As a consequence, any transceivers that were not associated when setting the remote offer remain disassociated after setting the local answer. This can be remedied by the answerer creating a follow-up offer, initiating another offer/answer exchange, or in the case of using addTrack()-created transceivers, making sure that enough media descriptions are offered in the initial exchange.

The RTP media API extends the RTCPeerConnection interface as described below.

          partial interface RTCPeerConnection {
  sequence<RTCRtpSender> getSenders();
  sequence<RTCRtpReceiver> getReceivers();
  sequence<RTCRtpTransceiver> getTransceivers();
  RTCRtpSender addTrack(MediaStreamTrack track, MediaStream... streams);
  undefined removeTrack(RTCRtpSender sender);
  RTCRtpTransceiver addTransceiver((MediaStreamTrack or DOMString) trackOrKind,
                                   optional RTCRtpTransceiverInit init = {});
  attribute EventHandler ontrack;
};

ontrack of type EventHandler: The event type of this event handler is track.

getSenders

Returns a sequence of RTCRtpSender objects representing the RTP senders that belong to non-stopped RTCRtpTransceiver objects currently attached to this RTCPeerConnection object.

When the getSenders method is invoked, the user agent MUST return the result of executing the CollectSenders algorithm.

We define the CollectSenders algorithm as follows:

Let transceivers be the result of executing the CollectTransceivers algorithm.
Let senders be a new empty sequence.
For each transceiver in transceivers,
1. If transceiver.[[Stopped]] is false, add transceiver.[[Sender]] to senders.
Return senders.

getReceivers

Returns a sequence of RTCRtpReceiver objects representing the RTP receivers that belong to non-stopped RTCRtpTransceiver objects currently attached to this RTCPeerConnection object.

When the getReceivers method is invoked, the user agent MUST run the following steps:

Let transceivers be the result of executing the CollectTransceivers algorithm.
Let receivers be a new empty sequence.
For each transceiver in transceivers,
1. If transceiver.[[Stopped]] is false, add transceiver.[[Receiver]] to receivers.
Return receivers.

getTransceivers

Returns a sequence of RTCRtpTransceiver objects representing the RTP transceivers that are currently attached to this RTCPeerConnection object.

The getTransceivers method MUST return the result of executing the CollectTransceivers algorithm.

We define the CollectTransceivers algorithm as follows:

Let transceivers be a new sequence consisting of all RTCRtpTransceiver objects in this RTCPeerConnection object's set of transceivers, in insertion order.
Return transceivers.

addTrack

Adds a new track to the RTCPeerConnection, and indicates that it is contained in the specified MediaStreams.

When the addTrack method is invoked, the user agent MUST run the following steps:

Let connection be the RTCPeerConnection object on which this method was invoked.
Let track be the MediaStreamTrack object indicated by the method's first argument.
Let kind be track.kind.
Let streams be a list of MediaStream objects constructed from the method's remaining arguments, or an empty list if the method was called with a single argument.
If connection.[[IsClosed]] is true, throw an InvalidStateError.
Let senders be the result of executing the CollectSenders algorithm. If an RTCRtpSender for track already exists in senders, throw an InvalidAccessError.
The steps below describe how to determine if an existing sender can be reused. Doing so will cause future calls to createOffer and createAnswer to mark the corresponding media description as sendrecv or sendonly and add the MSID of the sender's streams, as defined in [RFC9429] (section 5.2.2. and section 5.3.2.).

If any RTCRtpSender object in senders matches all the following criteria, let sender be that object, or null otherwise:
- The sender's track is null.
- The transceiver kind of the RTCRtpTransceiver, associated with the sender, matches kind.
- The [[Stopping]] slot of the RTCRtpTransceiver associated with the sender is false.
- The sender has never been used to send. More precisely, the [[CurrentDirection]] slot of the RTCRtpTransceiver associated with the sender has never had a value of "sendrecv" or "sendonly".
If sender is not null, run the following steps to use that sender:
1. Set sender.[[SenderTrack]] to track.
2. Set sender.[[AssociatedMediaStreamIds]] to an empty set.
3. For each stream in streams, add stream.id to [[AssociatedMediaStreamIds]] if it's not already there.
4. Let transceiver be the RTCRtpTransceiver associated with sender.
5. If transceiver.[[Direction]] is "recvonly", set transceiver.[[Direction]] to "sendrecv".
6. If transceiver.[[Direction]] is "inactive", set transceiver.[[Direction]] to "sendonly".
If sender is null, run the following steps:
1. Create an RTCRtpSender with track, kind and streams, and let sender be the result.
2. Create an RTCRtpReceiver with kind, and let receiver be the result.
3. Create an RTCRtpTransceiver with sender, receiver and an RTCRtpTransceiverDirection value of "sendrecv", and let transceiver be the result.
4. Add transceiver to connection's set of transceivers.
A track could have contents that are inaccessible to the application. This can be due to anything that would make a track CORS cross-origin. These tracks can be supplied to the addTrack() method, and have an RTCRtpSender created for them, but content MUST NOT be transmitted. Silence (audio), black frames (video) or equivalently absent content is sent in place of track content.

Note that this property can change over time.
Update the negotiation-needed flag for connection.
Return sender.

removeTrack

Stops sending media from sender. The RTCRtpSender will still appear in getSenders. Doing so will cause future calls to createOffer to mark the media description for the corresponding transceiver as "recvonly" or "inactive", as defined in [RFC9429] (section 5.2.2.).

When the other peer stops sending a track in this manner, the track is removed from any remote MediaStreams that were initially revealed in the track event, and if the MediaStreamTrack is not already muted, a mute event is fired at the track.

Note

The same effect as removeTrack() can be achieved by setting the RTCRtpTransceiver.direction attribute of the corresponding transceiver and invoking RTCRtpSender.replaceTrack(null) on the sender. One minor difference is that replaceTrack() is asynchronous and removeTrack() is synchronous.

When the removeTrack method is invoked, the user agent MUST run the following steps:

Let sender be the argument to removeTrack.
Let connection be the RTCPeerConnection object on which the method was invoked.
If connection.[[IsClosed]] is true, throw an InvalidStateError.
If sender was not created by connection, throw an InvalidAccessError.
Let transceiver be the RTCRtpTransceiver object corresponding to sender.
If transceiver.[[Stopping]] is true, abort these steps.
Let senders be the result of executing the CollectSenders algorithm.
If sender is not in senders (which indicates its transceiver was stopped or removed due to setting a session description of type "rollback"), then abort these steps.
If sender.[[SenderTrack]] is null, abort these steps.
Set sender.[[SenderTrack]] to null.
If transceiver.[[Direction]] is "sendrecv", set transceiver.[[Direction]] to "recvonly".
If transceiver.[[Direction]] is "sendonly", set transceiver.[[Direction]] to "inactive".
Update the negotiation-needed flag for connection.

addTransceiver

Create a new RTCRtpTransceiver and add it to the set of transceivers.

Adding a transceiver will cause future calls to createOffer to add a media description for the corresponding transceiver, as defined in [RFC9429] (section 5.2.2.).

The initial value of mid is null. Setting a session description may later change it to a non-null value.

The sendEncodings argument can be used to specify the number of offered simulcast encodings, and optionally their RIDs and encoding parameters.

When this method is invoked, the user agent MUST run the following steps:

Let init be the second argument.
Let streams be init.streams.
Let sendEncodings be init.sendEncodings.
Let direction be init.direction.
If the first argument is a string, let kind be the first argument and run the following steps:
1. If kind is neither "audio" nor "video", throw a TypeError.
2. Let track be null.
If the first argument is a MediaStreamTrack, let track be the first argument and let kind be track.kind.
If connection.[[IsClosed]] is true, throw an InvalidStateError.
Validate sendEncodings by running the following addTransceiver sendEncodings validation steps, where each RTCRtpEncodingParameters dictionary in it is an "encoding":
1. Candidate Correction 18:TypeError unless all or none of encodings have rids and on duplicate rids (PR #2774, PR #2775)
2. Verify that each rid value in sendEncodings conforms to the grammar specified in Section 10 of [RFC8851]. If one of the RIDs does not meet these requirements, throw a TypeError.
  If any of the following conditions are met, throw a TypeError:
  - Any encoding contains a rid member whose value does not conform to the grammar requirements specified in Section 10 of [RFC8851].
  - Some but not all encodings contain a rid member.
  - Any encoding contains a rid member whose value is the same as that of a rid contained in another encoding in sendEncodings.
3. If any encoding contains a read-only parameter other than rid, throw an InvalidAccessError.
4. Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)
  
  If any encoding contains a codec member whose value does not match any codec in RTCRtpSender.getCapabilities(kind).codecs, throw an OperationError.
5. Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)
  
  If the user agent does not support changing codecs without negotiation or does not support setting codecs for individual encodings, return a promise rejected with a newly created OperationError.
6. If kind is "audio", remove the scaleResolutionDownBy and maxFramerate members from all encodings that contain any of them.
7. If any encoding contains a scaleResolutionDownBy member whose value is less than 1.0, throw a RangeError.
8. Verify that the value of each maxFramerate member in sendEncodings that is defined is greater than 0.0. If one of the maxFramerate values does not meet this requirement, throw a RangeError.
9. Let maxN be the maximum number of total simultaneous encodings the user agent may support for this kind, at minimum 1.This should be an optimistic number since the codec to be used is not known yet.
10. If any encoding contains a scaleResolutionDownBy member, then for each encoding without one, add a scaleResolutionDownBy member with the value 1.0.
11. If the number of encodings stored in sendEncodings exceeds maxN, then trim sendEncodings from the tail until its length is maxN.
12. If kind is "video" and none of the encodings contain a scaleResolutionDownBy member, then for each encoding, add a scaleResolutionDownBy member with the value 2^(length of sendEncodings - encoding index - 1). This results in smaller-to-larger resolutions where the last encoding has no scaling applied to it, e.g. 4:2:1 if the length is 3.
13. If the number of encodings now stored in sendEncodings is 1, then remove any rid member from the lone entry.
  
  Note
  Providing a single, default RTCRtpEncodingParameters in sendEncodings allows the application to subsequently set encoding parameters using setParameters, even when simulcast isn't used.
Create an RTCRtpSender with track, kind, streams and sendEncodings and let sender be the result.

If sendEncodings is set, then subsequent calls to createOffer will be configured to send multiple RTP encodings as defined in [RFC9429] (section 5.2.2. and section 5.2.1.). When setRemoteDescription is called with a corresponding remote description that is able to receive multiple RTP encodings as defined in [RFC9429] (section 3.7.), the RTCRtpSender may send multiple RTP encodings and the parameters retrieved via the transceiver's sender.getParameters() will reflect the encodings negotiated.
Create an RTCRtpReceiver with kind and let receiver be the result.
Create an RTCRtpTransceiver with sender, receiver and direction, and let transceiver be the result.
Add transceiver to connection's set of transceivers.
Update the negotiation-needed flag for connection.
Return transceiver.

dictionary RTCRtpTransceiverInit {
  RTCRtpTransceiverDirection direction = "sendrecv";
  sequence<MediaStream> streams = [];
  sequence<RTCRtpEncodingParameters> sendEncodings = [];
};

direction of type RTCRtpTransceiverDirection, defaulting to "sendrecv": The direction of the RTCRtpTransceiver.
streams of type sequence<MediaStream>: When the remote RTCPeerConnection's track event fires corresponding to the RTCRtpReceiver being added, these are the streams that will be put in the event.
sendEncodings of type sequence<RTCRtpEncodingParameters>: A sequence containing parameters for sending RTP encodings of media.

enum RTCRtpTransceiverDirection {
  "sendrecv",
  "sendonly",
  "recvonly",
  "inactive",
  "stopped"
};

`RTCRtpTransceiverDirection` Enumeration description
Enum value	Description
`sendrecv`	The `RTCRtpTransceiver`'s `RTCRtpSender` `sender` will offer to send RTP, and will send RTP if the remote peer accepts and `sender`.`getParameters()`.`encodings`[`i`].`active` is `true` for any value of `i`. The `RTCRtpTransceiver`'s `RTCRtpReceiver` will offer to receive RTP, and will receive RTP if the remote peer accepts.
`sendonly`	The `RTCRtpTransceiver`'s `RTCRtpSender` `sender` will offer to send RTP, and will send RTP if the remote peer accepts and `sender`.`getParameters()`.`encodings`[`i`].`active` is `true` for any value of `i`. The `RTCRtpTransceiver`'s `RTCRtpReceiver` will not offer to receive RTP, and will not receive RTP.
`recvonly`	The `RTCRtpTransceiver`'s `RTCRtpSender` will not offer to send RTP, and will not send RTP. The `RTCRtpTransceiver`'s `RTCRtpReceiver` will offer to receive RTP, and will receive RTP if the remote peer accepts.
`inactive`	The `RTCRtpTransceiver`'s `RTCRtpSender` will not offer to send RTP, and will not send RTP. The `RTCRtpTransceiver`'s `RTCRtpReceiver` will not offer to receive RTP, and will not receive RTP.
`stopped`	The `RTCRtpTransceiver` will neither send nor receive RTP. It will generate a zero port in the offer. In answers, its `RTCRtpSender` will not offer to send RTP, and its `RTCRtpReceiver` will not offer to receive RTP. This is a terminal state.

An application can reject incoming media descriptions by setting the transceiver's direction to either "inactive" to turn off both directions temporarily, or to "sendonly" to reject only the incoming side. To permanently reject an m-line in a manner that makes it available for reuse, the application would need to call RTCRtpTransceiver.stop() and subsequently initiate negotiation from its end.

To process remote tracks given an RTCRtpTransceiver transceiver, direction, msids, addList, removeList, and trackEventInits, run the following steps:

Set the associated remote streams with transceiver.[[Receiver]], msids, addList, and removeList.
If direction is "sendrecv" or "recvonly" and transceiver.[[FiredDirection]] is neither "sendrecv" nor "recvonly", or the previous step increased the length of addList, process the addition of a remote track with transceiver and trackEventInits.
If direction is "sendonly" or "inactive", set transceiver.[[Receptive]] to false.
If direction is "sendonly" or "inactive", and transceiver.[[FiredDirection]] is either "sendrecv" or "recvonly", process the removal of a remote track for the media description, with transceiver and muteTracks.
Set transceiver.[[FiredDirection]] to direction.

To process the addition of a remote track given an RTCRtpTransceiver transceiver and trackEventInits, run the following steps:

Let receiver be transceiver.[[Receiver]].
Let track be receiver.[[ReceiverTrack]].
Let streams be receiver.[[AssociatedRemoteMediaStreams]].
Create a new RTCTrackEventInit dictionary with receiver, track, streams and transceiver as members and add it to trackEventInits.

To process the removal of a remote track with an RTCRtpTransceiver transceiver and muteTracks, run the following steps:

Let receiver be transceiver.[[Receiver]].
Let track be receiver.[[ReceiverTrack]].
If track.muted is false, add track to muteTracks.

To set the associated remote streams given RTCRtpReceiver receiver, msids, addList, and removeList, run the following steps:

Let connection be the RTCPeerConnection object associated with receiver.
For each MSID in msids, unless a MediaStream object has previously been created with that id for this connection, create a MediaStream object with that id.
Let streams be a list of the MediaStream objects created for this connection with the ids corresponding to msids.
Let track be receiver.[[ReceiverTrack]].
For each stream in receiver.[[AssociatedRemoteMediaStreams]] that is not present in streams, add stream and track as a pair to removeList.
For each stream in streams that is not present in receiver.[[AssociatedRemoteMediaStreams]], add stream and track as a pair to addList.
Set receiver.[[AssociatedRemoteMediaStreams]] to streams.

The RTCRtpSender interface allows an application to control how a given MediaStreamTrack is encoded and transmitted to a remote peer. When setParameters is called on an RTCRtpSender object, the encoding is changed appropriately.

To create an RTCRtpSender with a MediaStreamTrack, track, a string, kind, a list of MediaStream objects, streams, and optionally a list of RTCRtpEncodingParameters objects, sendEncodings, run the following steps:

Let sender be a new RTCRtpSender object.
Let sender have a [[SenderTrack]] internal slot initialized to track.
Let sender have a [[SenderTransport]] internal slot initialized to null.
Let sender have a [[LastStableStateSenderTransport]] internal slot initialized to null.
Let sender have a [[Dtmf]] internal slot initialized to null.
If kind is "audio" then create an RTCDTMFSender dtmf and set the [[Dtmf]] internal slot to dtmf.
Let sender have an [[AssociatedMediaStreamIds]] internal slot, representing a list of Ids of MediaStream objects that this sender is to be associated with. The [[AssociatedMediaStreamIds]] slot is used when sender is represented in SDP as described in [RFC9429] (section 5.2.1.).
Set sender.[[AssociatedMediaStreamIds]] to an empty set.
For each stream in streams, add stream.id to [[AssociatedMediaStreamIds]] if it's not already there.
Let sender have a [[SendEncodings]] internal slot, representing a list of RTCRtpEncodingParameters dictionaries.
If sendEncodings is given as input to this algorithm, and is non-empty, set the [[SendEncodings]] slot to sendEncodings. Otherwise, set it to a list containing a single new RTCRtpEncodingParameters dictionary, and if kind is "video", add a scaleResolutionDownBy member with the value 1.0 to that dictionary.

Note
RTCRtpEncodingParameters dictionaries contain active members whose values are true by default.
Candidate Correction 13:Rollback restores ridless encoding trounced by sRD(simulcastOffer). (PR #2797)

Let sender have a [[LastStableRidlessSendEncodings]] internal slot initialized to null.
Let sender have a [[SendCodecs]] internal slot, representing a list of RTCRtpCodecParameters dictionaries, and initialized to an empty list.
Let sender have a [[LastReturnedParameters]] internal slot, which will be used to match getParameters and setParameters transactions.
Return sender.

[Exposed=Window]
interface RTCRtpSender {
  readonly attribute MediaStreamTrack? track;
  readonly attribute RTCDtlsTransport? transport;
  static RTCRtpCapabilities? getCapabilities(DOMString kind);
  Promise<undefined> setParameters(RTCRtpSendParameters parameters,
      optional RTCSetParameterOptions setParameterOptions = {});
  RTCRtpSendParameters getParameters();
  Promise<undefined> replaceTrack(MediaStreamTrack? withTrack);
  undefined setStreams(MediaStream... streams);
  Promise<RTCStatsReport> getStats();
};

track of type MediaStreamTrack, readonly, nullable

The track attribute is the track that is associated with this RTCRtpSender object. If track is ended, or if the track's output is disabled, i.e. the track is disabled and/or muted, the RTCRtpSender MUST send black frames (video) and MUST NOT send (audio). In the case of video, the RTCRtpSender SHOULD send one black frame per second. If track is null then the RTCRtpSender does not send. On getting, the attribute MUST return the value of the [[SenderTrack]] slot.

transport of type RTCDtlsTransport, readonly, nullable

The transport attribute is the transport over which media from track is sent in the form of RTP packets. Prior to construction of the RTCDtlsTransport object, the transport attribute will be null. When bundling is used, multiple RTCRtpSender objects will share one transport and will all send RTP and RTCP over the same transport.

On getting, the attribute MUST return the value of the [[SenderTransport]] slot.

getCapabilities, static

The static RTCRtpSender.getCapabilities() method provides a way to discover the types of capabilities the user agent supports for sending media of the given kind, without reserving any resources, ports, or other state.

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

Let kind be the method's first argument.
If kind is neither "video" nor "audio" return null.
Return a new RTCRtpCapabilities dictionary, with its codecs member initialized to the list of implemented send codecs for kind, and its headerExtensions member initialized to the list of implemented header extensions for sending with kind.

The list of implemented send codecs, given kind, is an implementation-defined list of RTCRtpCodec dictionaries representing the most optimistic view of the codecs the user agent supports for sending media of the given kind (video or audio).

The , given kind, is an implementation-defined list of RTCRtpHeaderExtensionCapability dictionaries representing the most optimistic view of the header extensions the user agent supports for sending media of the given kind (video or audio).

These capabilities provide generally persistent cross-origin information on the device and thus increases the fingerprinting surface of the application. In privacy-sensitive contexts, user agents MAY consider mitigations such as reporting only a common subset of the capabilities.

Note

The codec capabilities returned affect the setCodecPreferences() algorithm and what inputs it throws InvalidModificationError on, and should also be consistent with information revealed by createOffer() and createAnswer() about codecs negotiated for sending, to ensure any privacy mitigations are effective.

setParameters

The setParameters method updates how track is encoded and transmitted to a remote peer.

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

Let parameters be the method's first argument.
Let sender be the RTCRtpSender object on which setParameters is invoked.
Let transceiver be the RTCRtpTransceiver object associated with sender (i.e. sender is transceiver.[[Sender]]).
If transceiver.[[Stopping]] is true, return a promise rejected with a newly created InvalidStateError.
If sender.[[LastReturnedParameters]] is null, return a promise rejected with a newly created InvalidStateError.
Validate parameters by running the following setParameters validation steps:
1. Let encodings be parameters.encodings.
2. Let codecs be parameters.codecs.
3. Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)
  
  Let choosableCodecs be codecs.
4. Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)
  
  If choosableCodecs is an empty list, set choosableCodecs to transceiver.[[PreferredCodecs]] and exclude any codecs not included in the list of implemented send codecs.
5. Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)
  
  If choosableCodecs is still an empty list, set choosableCodecs to the list of implemented send codecs for transceiver's kind.
6. Let N be the number of RTCRtpEncodingParameters stored in sender.[[SendEncodings]].
7. If any of the following conditions are met, return a promise rejected with a newly created InvalidModificationError:
  - encodings.length is different from N.
  - encodings has been re-ordered.
  - Any parameter in parameters is marked as a Read-only parameter (such as RID) and has a value that is different from the corresponding parameter value in sender.[[LastReturnedParameters]]. Note that this also applies to transactionId.
  - Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)
    Any encoding in encodings contains a codec not found in choosableCodecs, using the codec dictionary match algorithm with ignoreLevels set to true.
8. If transceiver kind is "audio", remove the scaleResolutionDownBy and maxFramerate members from all encodings that contain any of them.
9. If transceiver kind is "video", then for each encoding in encodings that doesn't contain a scaleResolutionDownBy member, add a scaleResolutionDownBy member with the value 1.0.
10. If transceiver kind is "video", and any encoding in encodings contains a scaleResolutionDownBy member whose value is less than 1.0, return a promise rejected with a newly created RangeError.
11. Verify that each encoding in encodings has a maxFramerate member whose value is greater than or equal to 0.0. If one of the maxFramerate values does not meet this requirement, return a promise rejected with a newly created RangeError.
12. Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)
  
  If the user agent does not support setting the codec for any encoding or mixing different codec values on the different encodings, return a promise rejected with a newly created OperationError.
Let p be a new promise.
In parallel, configure the media stack to use parameters to transmit sender.[[SenderTrack]].
1. If the media stack is successfully configured with parameters, queue a task to run the following steps:
  1. Set sender.[[LastReturnedParameters]] to null.
  2. Set sender.[[SendEncodings]] to parameters.encodings.
  3. Resolve p with undefined.
2. If any error occurred while configuring the media stack, queue a task to run the following steps:
  1. If an error occurred due to hardware resources not being available, reject p with a newly created RTCError whose errorDetail is set to "hardware-encoder-not-available" and abort these steps.
  2. If an error occurred due to a hardware encoder not supporting parameters, reject p with a newly created RTCError whose errorDetail is set to "hardware-encoder-error" and abort these steps.
  3. For all other errors, reject p with a newly created OperationError.
Return p.

setParameters does not cause SDP renegotiation and can only be used to change what the media stack is sending or receiving within the envelope negotiated by Offer/Answer. The attributes in the RTCRtpSendParameters dictionary are designed to not enable this, so attributes like cname that cannot be changed are read-only. Other things, like bitrate, are controlled using limits such as maxBitrate, where the user agent needs to ensure it does not exceed the maximum bitrate specified by maxBitrate, while at the same time making sure it satisfies constraints on bitrate specified in other places such as the SDP.

getParameters

The getParameters() method returns the RTCRtpSender object's current parameters for how track is encoded and transmitted to a remote RTCRtpReceiver.

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

Let sender be the RTCRtpSender object on which the getter was invoked.
If sender.[[LastReturnedParameters]] is not null, return sender.[[LastReturnedParameters]], and abort these steps.
Let result be a new RTCRtpSendParameters dictionary constructed as follows:
- transactionId is set to a new unique identifier.
- encodings is set to the value of the [[SendEncodings]] internal slot.
- The headerExtensions sequence is populated based on the header extensions that have been negotiated for sending.
- codecs is set to the value of the [[SendCodecs]] internal slot.
- rtcp.cname is set to the CNAME of the associated RTCPeerConnection. rtcp.reducedSize is set to true if reduced-size RTCP has been negotiated for sending, and false otherwise.
Set sender.[[LastReturnedParameters]] to result.
Queue a task that sets sender.[[LastReturnedParameters]] to null.
Return result.

getParameters may be used with setParameters to change the parameters in the following way:

Example 2

async function updateParameters() {
  try {
    const params = sender.getParameters();
    // ... make changes to parameters
    params.encodings[0].active = false;
    await sender.setParameters(params);
  } catch (err) {
    console.error(err);
  }
}

After a completed call to setParameters, subsequent calls to getParameters will return the modified set of parameters.

replaceTrack

Attempts to replace the RTCRtpSender's current track with another track provided (or with a null track), without renegotiation.

When the replaceTrack method is invoked, the user agent MUST run the following steps:

Let sender be the RTCRtpSender object on which replaceTrack is invoked.
Let transceiver be the RTCRtpTransceiver object associated with sender.
Let connection be the RTCPeerConnection object associated with sender.
Let withTrack be the argument to this method.
If withTrack is non-null and withTrack.kind differs from the transceiver kind of transceiver, return a promise rejected with a newly created TypeError.
Return the result of chaining the following steps to connection's operations chain:
1. If transceiver.[[Stopping]] is true, return a promise rejected with a newly created InvalidStateError.
2. Let p be a new promise.
3. Let sending be true if transceiver.[[CurrentDirection]] is "sendrecv" or "sendonly", and false otherwise.
4. Run the following steps in parallel:
  1. If sending is true, and withTrack is null, have the sender stop sending.
  2. If sending is true, and withTrack is not null, determine if withTrack can be sent immediately by the sender without violating the sender's already-negotiated envelope, and if it cannot, then:
    1. Queue a global task on the networking task source given the current realm's global object as global to reject p with a newly created InvalidModificationError.
    2. Abort these steps.
  3. If sending is true, and withTrack is not null, have the sender switch seamlessly to transmitting withTrack instead of the sender's existing track.
  4. Queue a task that runs the following steps:
    1. If connection.[[IsClosed]] is true, abort these steps.
    2. Set sender.[[SenderTrack]] to withTrack.
    3. Queue a global task on the networking task source given the current realm's global object as global to resolve p with undefined.
5. Return p.

Note

Changing dimensions and/or frame rates might not require negotiation. Cases that may require negotiation include:

Changing a resolution to a value outside of the negotiated imageattr bounds, as described in [RFC6236].
Changing a frame rate to a value that causes the block rate for the codec to be exceeded.
A video track differing in raw vs. pre-encoded format.
An audio track having a different number of channels.
Sources that also encode (typically hardware encoders) might be unable to produce the negotiated codec; similarly, software sources might not implement the codec that was negotiated for an encoding source.

setStreams

Sets the MediaStreams to be associated with this sender's track.

When the setStreams method is invoked, the user agent MUST run the following steps:

Let sender be the RTCRtpSender object on which this method was invoked.
Let connection be the RTCPeerConnection object on which this method was invoked.
If connection.[[IsClosed]] is true, throw an InvalidStateError.
Let streams be a list of MediaStream objects constructed from the method's arguments, or an empty list if the method was called without arguments.
Set sender.[[AssociatedMediaStreamIds]] to an empty set.
For each stream in streams, add stream.id to [[AssociatedMediaStreamIds]] if it's not already there.
Update the negotiation-needed flag for connection.

getStats

Gathers stats for this sender only and reports the result asynchronously.

When the getStats() method is invoked, the user agent MUST run the following steps:

Let selector be the RTCRtpSender object on which the method was invoked.
Let p be a new promise, and run the following steps in parallel:
1. Gather the stats indicated by selector according to the stats selection algorithm.
2. Queue a global task on the networking task source given the current realm's global object as global to resolve p with the resulting RTCStatsReport object, containing the gathered stats.
Return p.

dictionary RTCRtpParameters {
  required sequence<RTCRtpHeaderExtensionParameters> headerExtensions;
  required RTCRtcpParameters rtcp;
  required sequence<RTCRtpCodecParameters> codecs;
};

of type sequence<RTCRtpHeaderExtensionParameters>, required: A sequence containing parameters for RTP header extensions. Read-only parameter.
rtcp of type RTCRtcpParameters, required: Parameters used for RTCP. Read-only parameter.
codecs of type sequence<RTCRtpCodecParameters>, required: A sequence containing the media codecs that an RTCRtpSender will choose from, as well as entries for RTX, RED and FEC mechanisms. Corresponding to each media codec where retransmission via RTX is enabled, there will be an entry in codecs with a mimeType attribute indicating retransmission via audio/rtx or video/rtx, and an sdpFmtpLine attribute (providing the "apt" and "rtx-time" parameters). Read-only parameter.

dictionary RTCRtpSendParameters : RTCRtpParameters {
  required DOMString transactionId;
  required sequence<RTCRtpEncodingParameters> encodings;
};

transactionId of type DOMString, required: A unique identifier for the last set of parameters applied. Ensures that setParameters can only be called based on a previous getParameters, and that there are no intervening changes. Read-only parameter.
encodings of type sequence<RTCRtpEncodingParameters>, required: A sequence containing parameters for RTP encodings of media.

dictionary RTCRtpCodingParameters {
  DOMString rid;
};

rid of type DOMString: If set, this RTP encoding will be sent with the RID header extension as defined by [RFC9429] (section 5.2.1.). The RID is not modifiable via setParameters. It can only be set or modified in addTransceiver on the sending side. Read-only parameter.

dictionary RTCRtpEncodingParameters : RTCRtpCodingParameters {
  boolean active = true;
  RTCRtpCodec codec;
  unsigned long maxBitrate;
  double maxFramerate;
  double scaleResolutionDownBy;
};

active of type boolean, defaulting to true

Indicates that this encoding is actively being sent. Setting it to false causes this encoding to no longer be sent. Setting it to true causes this encoding to be sent. Since setting the value to false does not cause the SSRC to be removed, an RTCP BYE is not sent.

Candidate Addition 49:Add codec to RTCRtpEncodingParameters (PR #2985)

codec of type RTCRtpCodec

Optional value selecting which codec is used for this encoding's RTP stream. If absent, the user agent can choose to use any codec negotiated for sending.

When codec is set and [[SendCodecs]] have been negotiated, the user agent SHOULD use the first [[SendCodecs]] matching codec for sending, according to the codec dictionary match algorithm with ignoreLevels set to true.

maxBitrate of type unsigned long

When present, indicates the maximum bitrate that can be used to send this encoding. The user agent is free to allocate bandwidth between the encodings, as long as the maxBitrate value is not exceeded. The encoding may also be further constrained by other limits (such as per-transport or per-session bandwidth limits) below the maximum specified here. maxBitrate is computed the same way as the Transport Independent Application Specific Maximum (TIAS) bandwidth defined in [RFC3890] Section 6.2.2, which is the maximum bandwidth needed without counting IP or other transport layers like TCP or UDP. The unit of maxBitrate is bits per second.

Note

How the bitrate is achieved is media and encoding dependent. For video, a frame will always be sent as fast as possible, but frames may be dropped until bitrate is low enough. Thus, even a bitrate of zero will allow sending one frame. For audio, it might be necessary to stop playing if the bitrate does not allow the chosen encoding enough bandwidth to be sent.

maxFramerate of type double

This member can only be present if the sender's kind is "video". When present, indicates the maximum frame rate that can be used to send this encoding, in frames per second. The user agent is free to allocate bandwidth between the encodings, as long as the maxFramerate value is not exceeded.

If changed with setParameters(), the new frame rate takes effect after the current picture is completed; setting the max frame rate to zero thus has the effect of freezing the video on the next frame.

scaleResolutionDownBy of type double

This member is only present if the sender's kind is "video". The video's resolution will be scaled down in each dimension by the given value before sending. For example, if the value is 2.0, the video will be scaled down by a factor of 2 in each dimension, resulting in sending a video of one quarter the size. If the value is 1.0, the video will not be affected. The value must be greater than or equal to 1.0. By default, scaling is applied in reverse order by a factor of two, to produce an order of smaller to higher resolutions, e.g. 4:2:1. If there is only one layer, the sender will by default not apply any scaling, (i.e. scaleResolutionDownBy will be 1.0).

dictionary RTCRtcpParameters {
  DOMString cname;
  boolean reducedSize;
};

cname of type DOMString: The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages). Read-only parameter.
reducedSize of type boolean: Whether reduced size RTCP [RFC5506] is configured (if true) or compound RTCP as specified in [RFC3550] (if false). Read-only parameter.

dictionary RTCRtpCodec {
  required DOMString mimeType;
  required unsigned long clockRate;
  unsigned short channels;
  DOMString sdpFmtpLine;
};

The RTCRtpCodec dictionary provides information about codec objects.

mimeType of type DOMString, required: The codec MIME media type/subtype. Valid media types and subtypes are listed in [IANA-RTP-2].
clockRate of type unsigned long, required: The codec clock rate expressed in Hertz.
channels of type unsigned short: If present, indicates the maximum number of channels (mono=1, stereo=2).
sdpFmtpLine of type DOMString: The "format specific parameters" field from the a=fmtp line in the SDP corresponding to the codec, if one exists, as defined by [RFC9429] (section 5.8.).

dictionary RTCRtpCodecParameters : RTCRtpCodec {
  required octet payloadType;
};

The RTCRtpCodecParameters dictionary provides information about the negotiated codecs. The fields inherited from RTCRtpCodec MUST all be Read-only parameters.

For an RTCRtpSender, the sdpFmtpLine parameters come from the [[CurrentRemoteDescription]], and for an RTCRtpReceiver, they come from the local description (which is [[PendingLocalDescription]] if not null, and [[CurrentLocalDescription]] otherwise).

payloadType of type octet, required: The RTP payload type used to identify this codec. Read-only parameter.

dictionary RTCRtpCapabilities {
  required sequence<RTCRtpCodec> codecs;
  required sequence<RTCRtpHeaderExtensionCapability> headerExtensions;
};

codecs of type sequence<RTCRtpCodec>, required

Supported media codecs as well as entries for RTX, RED and FEC mechanisms. Only combinations that would utilize distinct payload types in a generated SDP offer are to be provided. For example:

Two H.264/AVC codecs, one for each of two supported packetization-mode values.
Two CN codecs with different clock rates.

There MUST only be a single entry in codecs for retransmission via RTX, with sdpFmtpLine not present.

of type sequence<RTCRtpHeaderExtensionCapability>, required

Supported RTP header extensions.

dictionary RTCSetParameterOptions {
};

RTCSetParameterOptions is defined as an empty dictionary to allow for extensibility.

The RTCRtpReceiver interface allows an application to inspect the receipt of a MediaStreamTrack.

To create an RTCRtpReceiver with a string, kind, run the following steps:

Let receiver be a new RTCRtpReceiver object.
Let track be a new MediaStreamTrack object [GETUSERMEDIA]. The source of track is a remote source provided by receiver. Note that the track.id is generated by the user agent and does not map to any track IDs on the remote side.
Initialize track.kind to kind.
Initialize track.label to the result of concatenating the string "remote " with kind.
Initialize track.readyState to live.
Initialize track.muted to true. See the MediaStreamTrack section about how the muted attribute reflects if a MediaStreamTrack is receiving media data or not.
Let receiver have a [[ReceiverTrack]] internal slot initialized to track.
Let receiver have a [[ReceiverTransport]] internal slot initialized to null.
Let receiver have a [[LastStableStateReceiverTransport]] internal slot initialized to null.
Let receiver have an [[AssociatedRemoteMediaStreams]] internal slot, representing a list of MediaStream objects that the MediaStreamTrack object of this receiver is associated with, and initialized to an empty list.
Let receiver have a [[LastStableStateAssociatedRemoteMediaStreams]] internal slot and initialize it to an empty list.
Let receiver have a [[ReceiveCodecs]] internal slot, representing a list of RTCRtpCodecParameters dictionaries, and initialized to an empty list.
Let receiver have a [[LastStableStateReceiveCodecs]] internal slot and initialize it to an empty list.
Let receiver have a [[JitterBufferTarget]] internal slot initialized to null.
Return receiver.

[Exposed=Window]
interface RTCRtpReceiver {
  readonly attribute MediaStreamTrack track;
  readonly attribute RTCDtlsTransport? transport;
  static RTCRtpCapabilities? getCapabilities(DOMString kind);
  RTCRtpReceiveParameters getParameters();
  sequence<RTCRtpContributingSource> getContributingSources();
  sequence<RTCRtpSynchronizationSource> getSynchronizationSources();
  Promise<RTCStatsReport> getStats();
  attribute DOMHighResTimeStamp? jitterBufferTarget;
};

track of type MediaStreamTrack, readonly

The track attribute is the track that is associated with this RTCRtpReceiver object receiver.

Note that track.stop() is final, although clones are not affected. Since receiver.track.stop() does not implicitly stop receiver, Receiver Reports continue to be sent. On getting, the attribute MUST return the value of the [[ReceiverTrack]] slot.

transport of type RTCDtlsTransport, readonly, nullable

The transport attribute is the transport over which media for the receiver's track is received in the form of RTP packets. Prior to construction of the RTCDtlsTransport object, the transport attribute will be null. When bundling is used, multiple RTCRtpReceiver objects will share one transport and will all receive RTP and RTCP over the same transport.

On getting, the attribute MUST return the value of the [[ReceiverTransport]] slot.

jitterBufferTarget of type DOMHighResTimeStamp, nullable

This attribute allows the application to specify a target duration of time in milliseconds of media for the RTCRtpReceiver's jitter buffer to hold. This influences the amount of buffering done by the user agent, which in turn affects retransmissions and packet loss recovery. Altering the target value allows applications to control the tradeoff between playout delay and the risk of running out of audio or video frames due to network jitter.

The user agent MUST have a minimum allowed target and a maximum allowed target reflecting what the user agent is able or willing to provide based on network conditions and memory constraints, which can change at any time.

Note

This is a target value. The resulting change in delay can be gradually observed over time. The receiver's average jitter buffer delay can be measured as the delta jitterBufferDelay divided by the delta jitterBufferEmittedCount.

An average delay is expected even if DTX is used. For example, if DTX is used and packets start flowing after silence, larger targets can influence the user agent to buffer these packets rather than playing them out.

On getting, this attribute MUST return the value of the [[JitterBufferTarget]] internal slot.

On setting, the user agent MUST run the following steps:

Let receiver be the RTCRtpReceiver object on which the setter is invoked.
Let target be the argument to the setter.
If target is negative or larger than 4000 milliseconds, then throw a RangeError.
Set receiver's [[JitterBufferTarget]] to target.
Let track be receiver's [[ReceiverTrack]].
in parallel, begin executing the following steps:
1. Update the underlying system about the new target, or that there is no application preference if target is null.
  
  If track is synchronized with another RTCRtpReceiver's track for audio/video synchronization, then the user agent SHOULD use the larger of the two receivers' [[JitterBufferTarget]] for both receivers.
  
  When the underlying system is applying a jitter buffer target, it will continuously make sure that the actual jitter buffer target is clamped within the minimum allowed target and maximum allowed target.
  Note
  If the user agent ends up using a target different from the requested one (e.g. due to network conditions or physical memory constraints), this is not reflected in the [[JitterBufferTarget]] internal slot.
2. Modifying the jitter buffer target of the underlying system SHOULD affect the internal audio or video buffering gradually in order not to hurt user experience. Audio samples or video frames SHOULD be accelerated or decelerated before playout, similarly to how it is done for audio/video synchronization or in response to congestion control.
  
  The acceleration or deceleration rate may vary depending on network conditions or the type of audio received (e.g. speech or background noise). It MAY take several seconds to achieve 1 second of buffering but SHOULD not take more than 30 seconds assuming packets are being received. The speed MAY be different for audio and video.

getCapabilities, static

The static RTCRtpReceiver.getCapabilities() method provides a way to discover the types of capabilities the user agent supports for receiving media of the given kind, without reserving any resources, ports, or other state.

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

Let kind be the method's first argument.
If kind is neither "video" nor "audio" return null.
Return a new RTCRtpCapabilities dictionary, with its codecs member initialized to the list of implemented receive codecs for kind, and its headerExtensions member initialized to the list of implemented header extensions for receiving for kind.

The list of implemented receive codecs, given kind, is an implementation-defined list of RTCRtpCodec dictionaries representing the most optimistic view of the codecs the user agent supports for receiving media of the given kind (video or audio).

The , given kind, is an implementation-defined list of RTCRtpHeaderExtensionCapability dictionaries representing an optimistic view of the header extensions the user agent supports for receiving media of the given kind (video or audio).

Note

The codec capabilities returned affect the setCodecPreferences() algorithm and what inputs it throws InvalidModificationError on, and should also be consistent with information revealed by createOffer() and createAnswer() about codecs negotiated for reception, to ensure any privacy mitigations are effective.

getParameters

The getParameters() method returns the RTCRtpReceiver object's current parameters for how track is decoded.

When getParameters is called, the RTCRtpReceiveParameters dictionary is constructed as follows:

The headerExtensions sequence is populated based on the header extensions that the receiver is currently prepared to receive.
codecs is set to the value of the [[ReceiveCodecs]] internal slot.

Note
Both the local and remote description may affect this list of codecs. For example, if three codecs are offered, the receiver will be prepared to receive each of them and will return them all from getParameters. But if the remote endpoint only answers with two, the absent codec will no longer be returned by getParameters as the receiver no longer needs to be prepared to receive it.
rtcp.reducedSize is set to true if the receiver is currently prepared to receive reduced-size RTCP packets, and false otherwise. rtcp.cname is left out.

getContributingSources

Returns an RTCRtpContributingSource for each unique CSRC identifier received by this RTCRtpReceiver in the last 10 seconds, in descending timestamp order.

getSynchronizationSources

Returns an RTCRtpSynchronizationSource for each unique SSRC identifier received by this RTCRtpReceiver in the last 10 seconds, in descending timestamp order.

getStats

Gathers stats for this receiver only and reports the result asynchronously.

When the getStats() method is invoked, the user agent MUST run the following steps:

Let selector be the RTCRtpReceiver object on which the method was invoked.
Let p be a new promise, and run the following steps in parallel:
1. Gather the stats indicated by selector according to the stats selection algorithm.
2. Queue a global task on the networking task source given the current realm's global object as global to resolve p with the resulting RTCStatsReport object, containing the gathered stats.
Return p.

The RTCRtpContributingSource and RTCRtpSynchronizationSource dictionaries contain information about a given contributing source (CSRC) or synchronization source (SSRC) respectively. When an audio or video frame from one or more RTP packets is delivered to the RTCRtpReceiver's MediaStreamTrack, the user agent MUST queue a task to update the relevant information for the RTCRtpContributingSource and RTCRtpSynchronizationSource dictionaries based on the content of those packets. The information relevant to the RTCRtpSynchronizationSource dictionary corresponding to the SSRC identifier, is updated each time, and if an RTP packet contains CSRC identifiers, then the information relevant to the RTCRtpContributingSource dictionaries corresponding to those CSRC identifiers is also updated. The user agent MUST process RTP packets in order of ascending RTP timestamps. The user agent MUST keep information from RTP packets delivered to the RTCRtpReceiver's MediaStreamTrack in the previous 10 seconds.

Note

Even if the MediaStreamTrack is not attached to any sink for playout, getSynchronizationSources and getContributingSources returns up-to-date information as long as the track is not ended; sinks are not a prerequisite for decoding RTP packets.

Note

As stated in the conformance section, requirements phrased as algorithms may be implemented in any manner so long as the end result is equivalent. So, an implementation does not need to literally queue a task for every frame, as long as the end result is that within a single event loop task execution, all returned RTCRtpSynchronizationSource and RTCRtpContributingSource dictionaries for a particular RTCRtpReceiver contain information from a single point in the RTP stream.

dictionary RTCRtpContributingSource {
  required DOMHighResTimeStamp timestamp;
  required unsigned long source;
  double audioLevel;
  required unsigned long rtpTimestamp;
};

timestamp of type DOMHighResTimeStamp, required

The timestamp indicating the most recent time a frame from an RTP packet, originating from this source, was delivered to the RTCRtpReceiver's MediaStreamTrack. The timestamp is defined as Performance.timeOrigin + Performance.now() at that time.

source of type unsigned long, required

The CSRC or SSRC identifier of the contributing or synchronization source.

audioLevel of type double

Only present for audio receivers. This is a value between 0..1 (linear), where 1.0 represents 0 dBov, 0 represents silence, and 0.5 represents approximately 6 dBSPL change in the sound pressure level from 0 dBov.

For CSRCs, this MUST be converted from the level value defined in [RFC6465] if the RFC 6465 header extension is present, otherwise this member MUST be absent.

For SSRCs, this MUST be converted from the level value defined in [RFC6464]. If the RFC 6464 header extension is not present in the received packets (such as if the other endpoint is not a user agent or is a legacy endpoint), this value SHOULD be absent.

Both RFCs define the level as an integral value from 0 to 127 representing the audio level in negative decibels relative to the loudest signal that the system could possibly encode. Thus, 0 represents the loudest signal the system could possibly encode, and 127 represents silence.

To convert these values to the linear 0..1 range, a value of 127 is converted to 0, and all other values are converted using the equation: 10^(-rfc_level/20).

rtpTimestamp of type unsigned long, required

The RTP timestamp, as defined in [RFC3550] Section 5.1, of the media played out at timestamp.

dictionary RTCRtpSynchronizationSource : RTCRtpContributingSource {};

The RTCRtpSynchronizationSource dictionary is expected to serve as an extension point for the specification to surface data only available in SSRCs.

The RTCRtpTransceiver interface represents a combination of an RTCRtpSender and an RTCRtpReceiver that share a common media stream "identification-tag". As defined in [RFC9429] (section 3.4.1.), an RTCRtpTransceiver is said to be associated with a media description if its "mid" property is non-null and matches a media stream "identification-tag" in the media description; otherwise it is said to be disassociated with that media description.

Note

A RTCRtpTransceiver may become associated with a new pending description in RFC9429 while still being disassociated with the current description. This may happen in check if negotiation is needed.

The transceiver kind of an RTCRtpTransceiver is defined by the kind of the associated RTCRtpReceiver's MediaStreamTrack object.

To create an RTCRtpTransceiver with an RTCRtpReceiver object, receiver, RTCRtpSender object, sender, and an RTCRtpTransceiverDirection value, direction, run the following steps:

Let transceiver be a new RTCRtpTransceiver object.
Let transceiver have a [[Sender]] internal slot, initialized to sender.
Let transceiver have a [[Receiver]] internal slot, initialized to receiver.
Let transceiver have a [[Stopping]] internal slot, initialized to false.
Let transceiver have a [[Stopped]] internal slot, initialized to false.
Let transceiver have a [[Direction]] internal slot, initialized to direction.
Let transceiver have a [[Receptive]] internal slot, initialized to false.
Let transceiver have a [[CurrentDirection]] internal slot, initialized to null.
Let transceiver have a [[FiredDirection]] internal slot, initialized to null.
Let transceiver have a [[PreferredCodecs]] internal slot, initialized to an empty list.
Let transceiver have a [[JsepMid]] internal slot, initialized to null. This is the "RtpTransceiver mid property" defined in [RFC9429] (section 5.2.1. and section 5.3.1.), and is only modified there.
Let transceiver have a [[Mid]] internal slot, initialized to null.
Return transceiver.

Note

Creating a transceiver does not create the underlying RTCDtlsTransport and RTCIceTransport objects. This will only occur as part of the process of setting a session description.

[Exposed=Window]
interface RTCRtpTransceiver {
  readonly attribute DOMString? mid;
  [SameObject] readonly attribute RTCRtpSender sender;
  [SameObject] readonly attribute RTCRtpReceiver receiver;
  attribute RTCRtpTransceiverDirection direction;
  readonly attribute RTCRtpTransceiverDirection? currentDirection;
  undefined stop();
  undefined setCodecPreferences(sequence<RTCRtpCodec> codecs);
};

mid of type DOMString, readonly, nullable

The mid attribute is the media stream "identification-tag" negotiated and present in the local and remote descriptions. On getting, the attribute MUST return the value of the [[Mid]] slot.

sender of type RTCRtpSender, readonly

The sender attribute exposes the RTCRtpSender corresponding to the RTP media that may be sent with mid = [[Mid]]. On getting, the attribute MUST return the value of the [[Sender]] slot.

receiver of type RTCRtpReceiver, readonly

The receiver attribute is the RTCRtpReceiver corresponding to the RTP media that may be received with mid = [[Mid]]. On getting the attribute MUST return the value of the [[Receiver]] slot.

direction of type RTCRtpTransceiverDirection

As defined in [RFC9429] (section 4.2.4.), the direction attribute indicates the preferred direction of this transceiver, which will be used in calls to createOffer and createAnswer. An update of directionality does not take effect immediately. Instead, future calls to createOffer and createAnswer mark the corresponding media description as sendrecv, sendonly, recvonly or inactive as defined in [RFC9429] (section 5.2.2. and section 5.3.2.)

On getting, the user agent MUST run the following steps:

Let transceiver be the RTCRtpTransceiver object on which the getter is invoked.
If transceiver.[[Stopping]] is true, return "stopped".
Otherwise, return the value of the [[Direction]] slot.

On setting, the user agent MUST run the following steps:

Let transceiver be the RTCRtpTransceiver object on which the setter is invoked.
Let connection be the RTCPeerConnection object associated with transceiver.
If transceiver.[[Stopping]] is true, throw an InvalidStateError.
Let newDirection be the argument to the setter.
If newDirection is equal to transceiver.[[Direction]], abort these steps.
If newDirection is equal to "stopped", throw a TypeError.
Set transceiver.[[Direction]] to newDirection.
Update the negotiation-needed flag for connection.

currentDirection of type RTCRtpTransceiverDirection, readonly, nullable

As defined in [RFC9429] (section 4.2.5.), the currentDirection attribute indicates the current direction negotiated for this transceiver. The value of currentDirection is independent of the value of RTCRtpEncodingParameters.active since one cannot be deduced from the other. If this transceiver has never been represented in an offer/answer exchange, the value is null. If the transceiver is stopped, the value is "stopped".

On getting, the user agent MUST run the following steps:

Let transceiver be the RTCRtpTransceiver object on which the getter is invoked.
If transceiver.[[Stopped]] is true, return "stopped".
Otherwise, return the value of the [[CurrentDirection]] slot.

stop

Irreversibly marks the transceiver as stopping, unless it is already stopped. This will immediately cause the transceiver's sender to no longer send, and its receiver to no longer receive. Calling stop() also updates the negotiation-needed flag for the RTCRtpTransceiver's associated RTCPeerConnection.

A stopping transceiver will cause future calls to createOffer to generate a zero port in the media description for the corresponding transceiver, as defined in [RFC9429] (section 4.2.1.) (The user agent MUST treat a stopping transceiver as stopped for the purposes of RFC9429 only in this case). However, to avoid problems with [RFC8843], a transceiver that is stopping, but not stopped, will not affect createAnswer.

A stopped transceiver will cause future calls to createOffer or createAnswer to generate a zero port in the media description for the corresponding transceiver, as defined in [RFC9429] (section 4.2.1.).

The transceiver will remain in the stopping state, unless it becomes stopped by setRemoteDescription processing a rejected m-line in a remote offer or answer.

Note

A transceiver that is stopping but not stopped will always need negotiation. In practice, this means that calling stop() on a transceiver will cause the transceiver to become stopped eventually, provided negotiation is allowed to complete on both ends.

When the stop method is invoked, the user agent MUST run the following steps:

Let transceiver be the RTCRtpTransceiver object on which the method is invoked.
Let connection be the RTCPeerConnection object associated with transceiver.
If connection.[[IsClosed]] is true, throw an InvalidStateError.
If transceiver.[[Stopping]] is true, abort these steps.
Stop sending and receiving with transceiver.
Update the negotiation-needed flag for connection.

The stop sending and receiving algorithm given a transceiver and, optionally, a disappear boolean defaulting to false, is as follows:

Let sender be transceiver.[[Sender]].
Let receiver be transceiver.[[Receiver]].
In parallel, stop sending media with sender, and send an RTCP BYE for each RTP stream that was being sent by sender, as specified in [RFC3550].
In parallel, stop receiving media with receiver.
If disappear is false, execute the steps for receiver.[[ReceiverTrack]] to be ended. This fires an event.
Set transceiver.[[Direction]] to "inactive".
Set transceiver.[[Stopping]] to true.

The stop the RTCRtpTransceiver algorithm given a transceiver and, optionally, a disappear boolean defaulting to false, is as follows:

If transceiver.[[Stopping]] is false, stop sending and receiving with transceiver and disappear.
Set transceiver.[[Stopped]] to true.
Set transceiver.[[Receptive]] to false.
Set transceiver.[[CurrentDirection]] to null.

setCodecPreferences

Candidate Addition 51:setCodecPreferences supports both send and receive codecs (filtered by direction) (PR #3018)

The setCodecPreferences method overrides the default codec preferences used by the user agent as input to negotiation. ~~When~~ When generating a session description ~~using either~~ using either createOffer or createAnswer, the user agent ~~MUST~~ ~~use~~ MUST filter the ~~indicated codecs~~preferred codecs on direction and, if this results in ~~the order~~ a non-empty list, it MUST use the specified codecs in ~~the~~ the order of the codecs argument, for the media ~~section~~ section corresponding to ~~this~~ this RTCRtpTransceiver.

This method allows applications to disable the negotiation of specific codecs (including RTX/RED/FEC). ~~It also allows an application to cause a remote peer to prefer the codec that appears first in the list~~ by listing all codecs except for ~~sending~~the ones to disable.

Note

If the m= section is used for receiving, the order of the codecs in the SDP (both the offer and the answer) tells the remote endpoint which codec the local endpoint prefers to receive. Even if the m= section is not used for receiving, an answerer that does not have any codec preferences of their own defaults to using the same order for its SDP answer.

Note

An RTCRtpSender defaults to sending what the remote endpoint indicated that it prefers to receive, but the application can change which codec to send amongst negotiated codecs by calling setParameters and specifying which codec to send. An RTCRtpReceiver is prepared to receive any negotiated codec.

Codec preferences remain in effect for all calls to createOffer and createAnswer that include this RTCRtpTransceiver until this method is called again. Setting codecs to an empty ~~sequence resets codec preferences to any~~ sequence, or one that becomes empty after direction filtering, results in default ~~value~~codec preferences.

Note

Codecs have their payload types listed under each m= section in the SDP, defining the mapping between payload types and codecs. These payload types are referenced by the m=video or m=audio lines in the order of preference, and codecs that are not negotiated do not appear in this list as defined in section 5.2.1 of [~~RFC8829~~RFC9429]. A previously negotiated codec that is subsequently removed disappears from the m=video or m=audio line, and while its codec payload type is not to be reused in future offers or answers, its payload type may also be removed from the mapping of payload types in the SDP.

~~The codecs sequence passed into~~ setCodecPreferences ~~can only contain~~ will reject attempts to set codecs not matching codecs ~~that are returned by~~ found in either RTCRtpSender.getCapabilities(kind) or RTCRtpReceiver.getCapabilities(kind), where kind is the kind of the RTCRtpTransceiver on which the method is called. Additionally, the RTCRtpCodecCapability dictionary members cannot be modified. If codecs does not fulfill these requirements, the user agent MUST throw an InvalidModificationError.

Note
Due to a recommendation in [SDP], calls to createAnswer SHOULD use only the common subset of the codec preferences and the codecs that appear in the offer. For example, if codec preferences are "C, B, A", but only codecs "A, B" were offered, the answer should only contain codecs "B, A". However, [RFC8829] (section 5.3.1.) allows adding codecs that were not in the offer, so implementations can behave differently.

When setCodecPreferences() in is invoked, the user agent MUST run the following steps:

Let transceiver be the RTCRtpTransceiver object this method was invoked on.
Let codecs be the first argument.
If codecs is an empty list, set transceiver.[[PreferredCodecs]] to codecs and abort these steps.
Remove any duplicate values in codecs. Start at the back of the list such that the priority of the codecs is maintained; the index of the first occurrence of a codec within the list is the same before and after this step.
Remove any duplicate values in codecs, ensuring that the first occurrence of each value remains in place.
Let kind be the transceiver's transceiver kind.
If the intersection between codecs and RTCRtpSender.getCapabilities(kind).codecs or the intersection between codecs and RTCRtpReceiver.getCapabilities(kind).codecs only contains RTX, RED or FEC codecs or is an empty set, throw InvalidModificationError. This ensures that we always have something to offer, regardless of transceiver.direction.
Let codecCapabilities be the union of RTCRtpSender.getCapabilities(kind).codecs and RTCRtpReceiver.getCapabilities(kind).codecs.
For each codec in codecs,
If codec is not in codecCapabilities, throw InvalidModificationError.
For each codec in codecs,
1. If codec does not match any codec in codecCapabilities, throw InvalidModificationError.
If codecs only contains entries for RTX, RED, FEC or Comfort Noise or is an empty set, throw InvalidModificationError. This ensures that we always have something to offer, regardless of transceiver.direction.
Set transceiver.[[PreferredCodecs]] to codecs.

The codec dictionary match algorithm given two RTCRtpCodec dictionaries first and second, and an ignoreLevels boolean defaulting to false if not specified, is as follows:

If first.mimeType is not an ASCII case-insensitive match for second.mimeType, return false.
If first.clockRate is different from second.clockRate, return false.
If either (but not both) of first.channels and second.channels are missing, or if they both exist and first.channels is different from second.channels, return false.
If either (but not both) of first.sdpFmtpLine and second.sdpFmtpLine are missing, return false.
If both first.sdpFmtpLine and second.sdpFmtpLine exist, run the following steps:
1. If either of first.sdpFmtpLine and second.sdpFmtpLine is not in key-value format, return the result of performing an equals comparison between first.sdpFmtpLine and second.sdpFmtpLine.
2. Let firstMediaFormat be a key-value map of the media formats constructed from first.sdpFmtpLine and secondMediaFormat be a key-value map of the media formats constructed from second.sdpFmtpLine.
  
  Note
  Which FMTP parameters make up the media format is codec specific. In some cases a parameter can be omitted and still be inferred, in which case it is also a part of the media format of that codec.
3. If firstMediaFormat is not equal to secondMediaFormat, return false.
4. Candidate Correction 52:Two codecs are considered the same even if level-id is not (PR #3023)
  If ignoreLevels is false and the highest complying bitstream levels inferred from first.sdpFmtpLine and second.sdpFmtpLine are different, return false.
  Note
  Even if ignoreLevels is true, some codecs (such as H.264) include levels in the media format, so that ignoring the level requires codec-specific parsing.
Return true.

Note

If set, the offerer's receive codec preferences will decide the order of the codecs in the offer. If the answerer does not have any codec preferences then the same order will be used in the answer. However, if the answerer also has codec preferences, these preferences override the order in the answer. In this case, the offerer's preferences would affect which codecs were on offer but not the final order.

Simulcast sending functionality is enabled by the addTransceiver method via its sendEncodings argument, or the setRemoteDescription method with a remote offer to receive simulcast, which are both methods on the RTCPeerConnection object. Additionally, the setParameters method on each RTCRtpSender object can be used to inspect and modify the functionality.

An RTCRtpSender's simulcast envelope is established in the first successful negotiation that involves it sending simulcast instead of unicast, and includes the maximum number of simulcast streams that can be sent, as well as the ordering of its encodings. This simulcast envelope may be narrowed (reducing the number of layers) in subsequent renegotiation, but cannot be reexpanded. Characteristics of individual simulcast streams can be modified using the setParameters method, but the simulcast envelope itself cannot be changed by that method.

One way to configure simulcast is with the sendEncodings option to addTransceiver(). While the addTrack() method lacks the sendEncodings argument necessary to configure simulcast, senders can be promoted to simulcast when the user agent is the answerer. Upon calling the setRemoteDescription method with a remote offer to receive simulcast, a proposed envelope is configured on an RTCRtpSender to contain the layers described in the specified session description. As long as this description isn't rolled back, the proposed envelope becomes the RTCRtpSender's simulcast envelope when negotiation completes. As above, this simulcast envelope may be narrowed in subsequent renegotiation, but not reexpanded.

Candidate Correction 12:Mark RTP Pause/Resume as not supported (PR #2755)

While setParameters cannot modify the ~~simulcast~~ simulcast envelope,, it is still possible to control the number of streams that are sent and the characteristics of those streams. Using setParameters, simulcast streams can be made inactive by setting the active member to false, or can be reactivated by setting the active member to true. [RFC7728] (RTP Pause/Resume) is not supported, nor is signaling of pause/resume via SDP Offer/Answer. Using setParameters, stream characteristics can be changed by modifying attributes such as maxBitrate.

Note

Simulcast is frequently used to send multiple encodings to an SFU, which will then forward one of the simulcast streams to the end user. The user agent is therefore expected to allocate bandwidth between encodings in such a way that all simulcast streams are usable on their own; for instance, if two simulcast streams have the same maxBitrate, one would expect to see a similar bitrate on both streams. If bandwidth does not permit all simulcast streams to be sent in an usable form, the user agent is expected to stop sending some of the simulcast streams.

As defined in [RFC9429] (section 3.7.), an offer from a user-agent will only contain a "send" description and no "recv" description on the a=simulcast line. Alternatives and restrictions (described in [RFC8853]) are not supported.

This specification does not define how to configure reception of multiple RTP encodings using createOffer, createAnswer or addTransceiver. However when setRemoteDescription is called with a corresponding remote description that is able to send multiple RTP encodings as defined in [RFC9429], and the browser supports receiving multiple RTP encodings, the RTCRtpReceiver may receive multiple RTP encodings and the parameters retrieved via the transceiver's receiver.getParameters() will reflect the encodings negotiated.

Note

An RTCRtpReceiver can receive multiple RTP streams in a scenario where a Selective Forwarding Unit (SFU) switches between simulcast streams it receives from user agents. If the SFU does not rewrite RTP headers so as to arrange the switched streams into a single RTP stream prior to forwarding, the RTCRtpReceiver will receive packets from distinct RTP streams, each with their own SSRC and sequence number space. While the SFU may only forward a single RTP stream at any given time, packets from multiple RTP streams can become intermingled at the receiver due to reordering. An RTCRtpReceiver equipped to receive multiple RTP streams will therefore need to be able to correctly order the received packets, recognize potential loss events and react to them. Correct operation in this scenario is non-trivial and therefore is optional for implementations of this specification.

This section is non-normative.

Examples of simulcast scenarios implemented with encoding parameters:

Example 3

// Example of 3-layer spatial simulcast with all but the lowest resolution layer disabled
var encodings = [
  {rid: 'q', active: true, scaleResolutionDownBy: 4.0}
  {rid: 'h', active: false, scaleResolutionDownBy: 2.0},
  {rid: 'f', active: false},
];

This section is non-normative.

Together, the direction attribute and the replaceTrack method enable developers to implement "hold" scenarios.

To send music to a peer and cease rendering received audio (music-on-hold):

Example 4

async function playMusicOnHold() {
  try {
    // Assume we have an audio transceiver and a music track named musicTrack
    await audio.sender.replaceTrack(musicTrack);
    // Mute received audio
    audio.receiver.track.enabled = false;
    // Set the direction to send-only (requires negotiation)
    audio.direction = 'sendonly';
  } catch (err) {
    console.error(err);
  }
}

To respond to a remote peer's "sendonly" offer:

Example 5

async function handleSendonlyOffer() {
  try {
    // Apply the sendonly offer first,
    // to ensure the receiver is ready for ICE candidates.
    await pc.setRemoteDescription(sendonlyOffer);
    // Stop sending audio
    await audio.sender.replaceTrack(null);
    // Align our direction to avoid further negotiation
    audio.direction = 'recvonly';
    // Call createAnswer and send a recvonly answer
    await doAnswer();
  } catch (err) {
    // handle signaling error
  }
}

To stop sending music and send audio captured from a microphone, as well to render received audio:

Example 6

async function stopOnHoldMusic() {
  // Assume we have an audio transceiver and a microphone track named micTrack
  await audio.sender.replaceTrack(micTrack);
  // Unmute received audio
  audio.receiver.track.enabled = true;
  // Set the direction to sendrecv (requires negotiation)
  audio.direction = 'sendrecv';
}

To respond to being taken off hold by a remote peer:

Example 7

async function onOffHold() {
  try {
    // Apply the sendrecv offer first, to ensure receiver is ready for ICE candidates.
    await pc.setRemoteDescription(sendrecvOffer);
    // Start sending audio
    await audio.sender.replaceTrack(micTrack);
    // Set the direction sendrecv (just in time for the answer)
    audio.direction = 'sendrecv';
    // Call createAnswer and send a sendrecv answer
    await doAnswer();
  } catch (err) {
    // handle signaling error
  }
}

The RTCDtlsTransport interface allows an application access to information about the Datagram Transport Layer Security (DTLS) transport over which RTP and RTCP packets are sent and received by RTCRtpSender and RTCRtpReceiver objects, as well other data such as SCTP packets sent and received by data channels. In particular, DTLS adds security to an underlying transport, and the RTCDtlsTransport interface allows access to information about the underlying transport and the security added. RTCDtlsTransport objects are constructed as a result of calls to setLocalDescription() and setRemoteDescription(). Each RTCDtlsTransport object represents the DTLS transport layer for the RTP or RTCP component of a specific RTCRtpTransceiver, or a group of RTCRtpTransceivers if such a group has been negotiated via [RFC8843].

Note

A new DTLS association for an existing RTCRtpTransceiver will be represented by an existing RTCDtlsTransport object, whose state will be updated accordingly, as opposed to being represented by a new object.

An RTCDtlsTransport has a [[DtlsTransportState]] internal slot initialized to "new" and a [[RemoteCertificates]] slot initialized to an empty list.

When the underlying DTLS transport experiences an error, such as a certificate validation failure, or a fatal alert (see [RFC5246] section 7.2), the user agent MUST queue a task that runs the following steps:

Let transport be the RTCDtlsTransport object to receive the state update and error notification.
If the state of transport is already "failed", abort these steps.
Set transport.[[DtlsTransportState]] to "failed".
Fire an event named error using the RTCErrorEvent interface with its errorDetail attribute set to either "dtls-failure" or "fingerprint-failure", as appropriate, and other fields set as described under the RTCErrorDetailType enum description, at transport.
Fire an event named statechange at transport.

When the underlying DTLS transport needs to update the state of the corresponding RTCDtlsTransport object for any other reason, the user agent MUST queue a task that runs the following steps:

Let transport be the RTCDtlsTransport object to receive the state update.
Let newState be the new state.
Set transport.[[DtlsTransportState]] to newState.
If newState is connected then let newRemoteCertificates be the certificate chain in use by the remote side, with each certificate encoded in binary Distinguished Encoding Rules (DER) [X690], and set transport.[[RemoteCertificates]] to newRemoteCertificates.
Fire an event named statechange at transport.

[Exposed=Window]
interface RTCDtlsTransport : EventTarget {
  [SameObject] readonly attribute RTCIceTransport iceTransport;
  readonly attribute RTCDtlsTransportState state;
  sequence<ArrayBuffer> getRemoteCertificates();
  attribute EventHandler onstatechange;
  attribute EventHandler onerror;
};

iceTransport of type RTCIceTransport, readonly: The iceTransport attribute is the underlying transport that is used to send and receive packets. The underlying transport may not be shared between multiple active RTCDtlsTransport objects.
state of type RTCDtlsTransportState, readonly: The state attribute MUST, on getting, return the value of the [[DtlsTransportState]] slot.
onstatechange of type EventHandler: The event type of this event handler is statechange.
onerror of type EventHandler: The event type of this event handler is error.

getRemoteCertificates: Returns the value of [[RemoteCertificates]].

enum RTCDtlsTransportState {
  "new",
  "connecting",
  "connected",
  "closed",
  "failed"
};

`RTCDtlsTransportState` Enumeration description
Enum value	Description
`new`	DTLS has not started negotiating yet.
`connecting`	DTLS is in the process of negotiating a secure connection and verifying the remote fingerprint.
`connected`	DTLS has completed negotiation of a secure connection and verified the remote fingerprint.
`closed`	The transport has been closed intentionally as the result of receipt of a close_notify alert, or calling `close()`.
`failed`	The transport has failed as the result of an error (such as receipt of an error alert or failure to validate the remote fingerprint).

The RTCDtlsFingerprint dictionary includes the hash function algorithm and certificate fingerprint as described in [RFC4572].

dictionary RTCDtlsFingerprint {
  DOMString algorithm;
  DOMString value;
};

algorithm of type DOMString: One of the the hash function algorithms defined in the 'Hash function Textual Names' registry [IANA-HASH-FUNCTION].
value of type DOMString: The value of the certificate fingerprint in lowercase hex string as expressed utilizing the syntax of 'fingerprint' in [RFC4572] Section 5.

The RTCIceTransport interface allows an application access to information about the ICE transport over which packets are sent and received. In particular, ICE manages peer-to-peer connections which involve state which the application may want to access. RTCIceTransport objects are constructed as a result of calls to setLocalDescription() and setRemoteDescription(). The underlying ICE state is managed by the ICE agent; as such, the state of an RTCIceTransport changes when the ICE Agent provides indications to the user agent as described below. Each RTCIceTransport object represents the ICE transport layer for the RTP or RTCP component of a specific RTCRtpTransceiver, or a group of RTCRtpTransceivers if such a group has been negotiated via [RFC8843].

Note

An ICE restart for an existing RTCRtpTransceiver will be represented by an existing RTCIceTransport object, whose state will be updated accordingly, as opposed to being represented by a new object.

Candidate Correction 24:Queue two tasks upon finishing ICE gathering, and fire gatheringstatechange & icegatheringstatechange in same task (PR #2894)

When the ICE Agent indicates that it began gathering a generation of candidates for an RTCIceTransport transport associated with an RTCPeerConnection connection, the user agent MUST queue a task that runs the following steps:

~~Let connection be the RTCPeerConnection object associated with this ICE Agent.~~
If connection.[[IsClosed]] is true, abort these steps.
~~Let transport be the RTCIceTransport for which candidate gathering began.~~
Set transport.[[IceGathererState]] to gathering.
.
Set connection.[[IceGatheringState]] to the value of deriving a new state value as described by the RTCIceGatheringState enum.
Let connectionIceGatheringStateChanged be true if connection.[[IceGatheringState]] changed in the previous step, otherwise false.
Do not read or modify state beyond this point.
Fire an event named gatheringstatechange at transport.
~~Update the ICE gathering state of connection.~~
If connectionIceGatheringStateChanged is true, fire an event named icegatheringstatechange at connection.

When the ICE Agent is finished gathering a generation of candidates for an RTCIceTransport transport associated with an RTCPeerConnection connection, and those candidates have been surfaced to the application, the user agent MUST queue a task ~~that runs~~ to run the ~~following~~ following steps:

~~Let connection be the RTCPeerConnection object associated with this ICE Agent.~~
If connection.[[IsClosed]] is true, abort these steps.
~~Let transport be the RTCIceTransport for which candidate gathering finished.~~
If connection.[[PendingLocalDescription]] is not null, and represents the ICE generation for which gathering finished, add a=end-of-candidates to connection.[[PendingLocalDescription]].sdp.
If connection.[[CurrentLocalDescription]] is not null, and represents the ICE generation for which gathering finished, add a=end-of-candidates to connection.[[CurrentLocalDescription]].sdp.
Let ~~newCandidate~~endOfGatheringCandidate be the result of creating an an RTCIceCandidate with a new dictionary whose sdpMid and sdpMLineIndex are set to the values associated with this RTCIceTransport, usernameFragment is is set to the username fragment of the generation of candidates for which gathering finished, and candidate is ~~set~~ set to ~~an empty string~~"".
Fire an event named icecandidate using the RTCPeerConnectionIceEvent interface with the candidate attribute set to ~~newCandidate~~endOfGatheringCandidate at connection.

If another generation of candidates is still being gathered, abort these steps.
Note
This may occur if an ICE restart is initiated while the ICE agent is still gathering the previous generation of candidates.
Set transport.[[IceGathererState]] to complete.
Fire an event named gatheringstatechange at transport.
Update the ICE gathering state of connection.

When the ICE Agent has queued the above task, and no other generations of candidates is being gathered, the user agent MUST also queue a second task to run the following steps:

Note

Other generations of candidates might still be gathering if an ICE restart was initiated while the ICE agent is still gathering the previous generation of candidates.

If connection.[[IsClosed]] is true, abort these steps.
Set transport.[[IceGathererState]] to complete.
Set connection.[[IceGatheringState]] to the value of deriving a new state value as described by the RTCIceGatheringState enum.
Let connectionIceGatheringStateChanged be true if connection.[[IceGatheringState]] changed in the previous step, otherwise false.
Do not read or modify state beyond this point.
Fire an event named gatheringstatechange at transport.
If connectionIceGatheringStateChanged is true, fire an event named icegatheringstatechange at connection.
Fire an event named icecandidate using the RTCPeerConnectionIceEvent interface with the candidate attribute set to null at connection.
Note
The null candidate event is fired to ensure legacy compatibility. New code should monitor the gathering state of RTCIceTransport and/or RTCPeerConnection.

When the ICE Agent indicates that a new ICE candidate is available for an RTCIceTransport, either by taking one from the ICE candidate pool or gathering it from scratch, the user agent MUST queue a task that runs the following steps:

Let candidate be the available ICE candidate.
Let connection be the RTCPeerConnection object associated with this ICE Agent.
If connection.[[IsClosed]] is true, abort these steps.
If either connection.[[PendingLocalDescription]] or connection.[[CurrentLocalDescription]] are not null, and represent the ICE generation for which candidate was gathered, surface the candidate with candidate and connection, and abort these steps.
Otherwise, append candidate to connection.[[EarlyCandidates]].

When the ICE Agent signals that the ICE role has changed due to an ICE binding request with a role collision per [RFC8445] section 7.3.1.1, the UA will queue a task to set the value of [[IceRole]] to the new value.

To release early candidates of a connection, run the following steps:

For each candidate, candidate, in connection.[[EarlyCandidates]], queue a task to surface the candidate with candidate and connection.
Set connection.[[EarlyCandidates]] to an empty list.

To surface a candidate with candidate and connection, run the following steps:

If connection.[[IsClosed]] is true, abort these steps.
Let transport be the RTCIceTransport for which candidate is being made available.
If connection.[[PendingLocalDescription]] is not null, and represents the ICE generation for which candidate was gathered, add candidate to connection.[[PendingLocalDescription]].sdp.
If connection.[[CurrentLocalDescription]] is not null, and represents the ICE generation for which candidate was gathered, add candidate to connection.[[CurrentLocalDescription]].sdp.
Let newCandidate be the result of creating an RTCIceCandidate with a new dictionary whose sdpMid and sdpMLineIndex are set to the values associated with this RTCIceTransport, usernameFragment is set to the username fragment of the candidate, and candidate is set to a string encoded using the candidate-attribute grammar to represent candidate.
Add newCandidate to transport's set of local candidates.
Fire an event named icecandidate using the RTCPeerConnectionIceEvent interface with the candidate attribute set to newCandidate at connection.

The RTCIceTransportState of an RTCIceTransport may change because a candidate pair with a usable connection was found and selected or it may change without the selected candidate pair changing. The selected pair and RTCIceTransportState are related and are handled in the same task.

When the ICE Agent indicates that an RTCIceTransport has changed either the selected candidate pair, the RTCIceTransportState or both, the user agent MUST queue a task that runs the steps to change the selected candidate pair and state:

Let connection be the RTCPeerConnection object associated with this ICE Agent.
If connection.[[IsClosed]] is true, abort these steps.
Let transport be the RTCIceTransport whose state is changing.
Let selectedCandidatePairChanged be false.
Let transportIceConnectionStateChanged be false.
Let connectionIceConnectionStateChanged be false.
Let connectionStateChanged be false.
If transport's selected candidate pair was changed, run the following steps:
1. Let newCandidatePair be the result of creating an RTCIceCandidatePair with local and remote, representing the local and remote candidates of the indicated pair if one is selected, and null otherwise.
2. Set transport.[[SelectedCandidatePair]] to newCandidatePair.
3. Set selectedCandidatePairChanged to true.
If transport's RTCIceTransportState was changed, run the following steps:
1. Set transport.[[IceTransportState]] to the new indicated RTCIceTransportState.
2. Set transportIceConnectionStateChanged to true.
3. Set connection.[[IceConnectionState]] to the value of deriving a new state value as described by the RTCIceConnectionState enum.
4. If connection.[[IceConnectionState]] changed in the previous step, set connectionIceConnectionStateChanged to true.
5. Set connection.[[ConnectionState]] to the value of deriving a new state value as described by the RTCPeerConnectionState enum.
6. If connection.[[ConnectionState]] changed in the previous step, set connectionStateChanged to true.
If selectedCandidatePairChanged is true, fire an event named selectedcandidatepairchange at transport.
If transportIceConnectionStateChanged is true, fire an event named statechange at transport.
If connectionIceConnectionStateChanged is true, fire an event named iceconnectionstatechange at connection.
If connectionStateChanged is true, fire an event named connectionstatechange at connection.

An RTCIceTransport object has the following internal slots:

[[IceTransportState]] initialized to "new"
[[IceGathererState]] initialized to "new"
[[SelectedCandidatePair]] initialized to null
[[IceRole]] initialized to "unknown"

[Exposed=Window]
interface RTCIceTransport : EventTarget {
  readonly attribute RTCIceRole role;
  readonly attribute RTCIceComponent component;
  readonly attribute RTCIceTransportState state;
  readonly attribute RTCIceGathererState gatheringState;
  sequence<RTCIceCandidate> getLocalCandidates();
  sequence<RTCIceCandidate> getRemoteCandidates();
  RTCIceCandidatePair? getSelectedCandidatePair();
  RTCIceParameters? getLocalParameters();
  RTCIceParameters? getRemoteParameters();
  attribute EventHandler onstatechange;
  attribute EventHandler ongatheringstatechange;
  attribute EventHandler onselectedcandidatepairchange;
};

role of type RTCIceRole, readonly: The role attribute MUST, on getting, return the value of the [[IceRole]] internal slot.
component of type RTCIceComponent, readonly: The component attribute MUST return the ICE component of the transport. When RTCP mux is used, a single RTCIceTransport transports both RTP and RTCP and component is set to "rtp".
state of type RTCIceTransportState, readonly: The state attribute MUST, on getting, return the value of the [[IceTransportState]] slot.
gatheringState of type RTCIceGathererState, readonly: The gatheringState attribute MUST, on getting, return the value of the [[IceGathererState]] slot.
onstatechange of type EventHandler: This event handler, of event handler event type statechange, MUST be fired any time the RTCIceTransport state changes.
ongatheringstatechange of type EventHandler: This event handler, of event handler event type gatheringstatechange, MUST be fired any time the RTCIceTransport's [[IceGathererState]] changes.
onselectedcandidatepairchange of type EventHandler: This event handler, of event handler event type selectedcandidatepairchange, MUST be fired any time the RTCIceTransport's selected candidate pair changes.

getLocalCandidates: Returns a sequence describing the local ICE candidates gathered for this RTCIceTransport and sent in onicecandidate.
getRemoteCandidates: Returns a sequence describing the remote ICE candidates received by this RTCIceTransport via addIceCandidate().

Note
getRemoteCandidates will not expose peer reflexive candidates since they are not received via addIceCandidate().
getSelectedCandidatePair: Returns the selected candidate pair on which packets are sent. This method MUST return the value of the [[SelectedCandidatePair]] slot. When RTCIceTransport.state is "new" or "closed" getSelectedCandidatePair returns null.
getLocalParameters: Returns the local ICE parameters received by this RTCIceTransport via setLocalDescription, or null if the parameters have not yet been received.
getRemoteParameters: Returns the remote ICE parameters received by this RTCIceTransport via setRemoteDescription or null if the parameters have not yet been received.

dictionary RTCIceParameters {
  DOMString usernameFragment;
  DOMString password;
};

usernameFragment of type DOMString: The ICE username fragment as defined in [RFC5245], Section 7.1.2.3.
password of type DOMString: The ICE password as defined in [RFC5245], Section 7.1.2.3.

Candidate Addition 45:Convert RTCIceCandidatePair dictionary to an interface (PR #2961)

~~5.6.2 RTCIceCandidatePair Dictionary~~

This interface represents an ICE candidate pair, described in Section 4 in [RFC8445]. An RTCIceCandidatePair is a pairing of a local and a remote RTCIceCandidate.

To create an RTCIceCandidatePair with RTCIceCandidate objects, local and remote, run the following steps:

Let candidatePair be a newly created RTCIceCandidatePair object.
Let candidatePair have a [[Local]] internal slot, initialized to local.
Let candidatePair have a [[Remote]] internal slot, initialized to remote.
Return candidatePair.

dictionary [Exposed=Window]
interface RTCIceCandidatePair {
  [SameObject] readonly attribute RTCIceCandidate local;
  [SameObject] readonly attribute RTCIceCandidate remote;
};

~~Dictionary RTCIceCandidatePair Members~~

local of type RTCIceCandidate , readonly: ~~The local ICE candidate.~~; The local attribute MUST, on getting, return the value of the [[Local]] internal slot.
remote of type RTCIceCandidate , readonly: ~~The remote ICE candidate.~~; The remote attribute MUST, on getting, return the value of the [[Remote]] internal slot.

enum RTCIceGathererState {
  "new",
  "gathering",
  "complete"
};

`RTCIceGathererState` Enumeration description
Enum value	Description
`new`	The `RTCIceTransport` was just created, and has not started gathering candidates yet.
`gathering`	The `RTCIceTransport` is in the process of gathering candidates.
`complete`	The `RTCIceTransport` has completed gathering and the end-of-candidates indication for this transport has been sent. It will not gather candidates again until an ICE restart causes it to restart.

enum RTCIceTransportState {
  "closed",
  "failed",
  "disconnected",
  "new",
  "checking",
  "completed",
  "connected"
};

`RTCIceTransportState` Enumeration description
Enum value	Description
`closed`	The `RTCIceTransport` has shut down and is no longer responding to STUN requests.
`failed`	The `RTCIceTransport` has finished gathering, received an indication that there are no more remote candidates, finished checking all candidate pairs, and all pairs have either failed connectivity checks or lost consent, and either zero local candidates were gathered or the PAC timer has expired [RFC8863]. This is a terminal state until ICE is restarted. Since an ICE restart may cause connectivity to resume, entering the "`failed`" state does not cause DTLS transports, SCTP associations or the data channels that run over them to close, or tracks to mute.
`disconnected`	The ICE Agent has determined that connectivity is currently lost for this `RTCIceTransport`. This is a transient state that may trigger intermittently (and resolve itself without action) on a flaky network. The way this state is determined is implementation dependent. Examples include: Losing the network interface for the connection in use. Repeatedly failing to receive a response to STUN requests. Alternatively, the `RTCIceTransport` has finished checking all existing candidates pairs and not found a connection (or consent checks [RFC7675] once successful, have now failed), but it is still gathering and/or waiting for additional remote candidates.
`new`	The `RTCIceTransport` is gathering candidates and/or waiting for remote candidates to be supplied, and has not yet started checking.
`checking`	The `RTCIceTransport` has received at least one remote candidate (by means of `addIceCandidate()` or discovered as a peer-reflexive candidate when receiving a STUN binding request) and is checking candidate pairs and has either not yet found a connection or consent checks [RFC7675] have failed on all previously successful candidate pairs. In addition to checking, it may also still be gathering.
`completed`	The `RTCIceTransport` has finished gathering, received an indication that there are no more remote candidates, finished checking all candidate pairs and found a connection. If consent checks [RFC7675] subsequently fail on all successful candidate pairs, the state transitions to "`failed`".
`connected`	The `RTCIceTransport` has found a usable connection, but is still checking other candidate pairs to see if there is a better connection. It may also still be gathering and/or waiting for additional remote candidates. If consent checks [RFC7675] fail on the connection in use, and there are no other successful candidate pairs available, then the state transitions to "`checking`" (if there are candidate pairs remaining to be checked) or "`disconnected`" (if there are no candidate pairs to check, but the peer is still gathering and/or waiting for additional remote candidates).

Note

The most common transitions for a successful call will be new -> checking -> connected -> completed, but under specific circumstances (only the last checked candidate succeeds, and gathering and the no-more candidates indication both occur prior to success), the state can transition directly from "checking" to "completed".

An ICE restart causes candidate gathering and connectivity checks to begin anew, causing a transition to "connected" if begun in the "completed" state. If begun in the transient "disconnected" state, it causes a transition to "checking", effectively forgetting that connectivity was previously lost.

The "failed" and "completed" states require an indication that there are no additional remote candidates. This can be indicated by calling addIceCandidate with a candidate value whose candidate property is set to an empty string or by canTrickleIceCandidates being set to false.

Some example state transitions are:

(RTCIceTransport first created, as a result of setLocalDescription or setRemoteDescription): "new"
("new", remote candidates received): "checking"
("checking", found usable connection): "connected"
("checking", checks fail but gathering still in progress): "disconnected"
("checking", gave up): "failed"
("disconnected", new local candidates): "checking"
("connected", finished all checks): "completed"
("completed", lost connectivity): "disconnected"
("disconnected" or "failed", ICE restart occurs): "checking"
("completed", ICE restart occurs): "connected"
RTCPeerConnection.close(): "closed"

Figure 2 Non-normative ICE transport state transition diagram

enum RTCIceRole {
  "unknown",
  "controlling",
  "controlled"
};

`RTCIceRole` Enumeration description
Enum value	Description
`unknown`	An agent whose role as defined by [RFC5245], Section 3, has not yet been determined.
`controlling`	A controlling agent as defined by [RFC5245], Section 3.
`controlled`	A controlled agent as defined by [RFC5245], Section 3.

enum RTCIceComponent {
  "rtp",
  "rtcp"
};

`RTCIceComponent` Enumeration description
Enum value	Description
`rtp`	The ICE Transport is used for RTP (or RTCP multiplexing), as defined in [RFC5245], Section 4.1.1.1. Protocols multiplexed with RTP (e.g. data channel) share its component ID. This represents the `component-id` value `1` when encoded in `candidate-attribute`.
`rtcp`	The ICE Transport is used for RTCP as defined by [RFC5245], Section 4.1.1.1. This represents the `component-id` value `2` when encoded in `candidate-attribute`.

The track event uses the RTCTrackEvent interface.

The Peer-to-peer Data API lets a web application send and receive generic application data peer-to-peer. The API for sending and receiving data models the behavior of Web Sockets.

The Peer-to-peer data API extends the RTCPeerConnection interface as described below.

          partial interface RTCPeerConnection {
  readonly attribute RTCSctpTransport? sctp;
  RTCDataChannel createDataChannel(USVString label,
                                   optional RTCDataChannelInit dataChannelDict = {});
  attribute EventHandler ondatachannel;
};

sctp of type RTCSctpTransport, readonly, nullable: The SCTP transport over which SCTP data is sent and received. If SCTP has not been negotiated, the value is null. This attribute MUST return the RTCSctpTransport object stored in the [[SctpTransport]] internal slot.
ondatachannel of type EventHandler: The event type of this event handler is datachannel.

createDataChannel

Creates a new RTCDataChannel object with the given label. The RTCDataChannelInit dictionary can be used to configure properties of the underlying channel such as data reliability.

When the createDataChannel method is invoked, the user agent MUST run the following steps.

Let connection be the RTCPeerConnection object on which the method is invoked.
If connection.[[IsClosed]] is true, throw an InvalidStateError.
Create an RTCDataChannel, channel.
Initialize channel.[[DataChannelLabel]] to the value of the first argument.
If the UTF-8 representation of [[DataChannelLabel]] is longer than 65535 bytes, throw a TypeError.
Let options be the second argument.
Initialize channel.[[MaxPacketLifeTime]] to option.maxPacketLifeTime, if present, otherwise null.
Initialize channel.[[MaxRetransmits]] to option.maxRetransmits, if present, otherwise null.
Initialize channel.[[Ordered]] to option.ordered.
Initialize channel.[[DataChannelProtocol]] to option.protocol.
If the UTF-8 representation of [[DataChannelProtocol]] is longer than 65535 bytes, throw a TypeError.
Initialize channel.[[Negotiated]] to option.negotiated.
Initialize channel.[[DataChannelId]] to the value of option.id, if it is present and [[Negotiated]] is true, otherwise null.

Note
This means the id member will be ignored if the data channel is negotiated in-band; this is intentional. Data channels negotiated in-band should have IDs selected based on the DTLS role, as specified in [RFC8832].
If [[Negotiated]] is true and [[DataChannelId]] is null, throw a TypeError.
If both [[MaxPacketLifeTime]] and [[MaxRetransmits]] attributes are set (not null), throw a TypeError.
If a setting, either [[MaxPacketLifeTime]] or [[MaxRetransmits]], has been set to indicate unreliable mode, and that value exceeds the maximum value supported by the user agent, the value MUST be set to the user agents maximum value.
If [[DataChannelId]] is equal to 65535, which is greater than the maximum allowed ID of 65534 but still qualifies as an unsigned short, throw a TypeError.
If the [[DataChannelId]] slot is null (due to no ID being passed into createDataChannel, or [[Negotiated]] being false), and the DTLS role of the SCTP transport has already been negotiated, then initialize [[DataChannelId]] to a value generated by the user agent, according to [RFC8832], and skip to the next step. If no available ID could be generated, or if the value of the [[DataChannelId]] slot is being used by an existing RTCDataChannel, throw an OperationError exception.

Note
If the [[DataChannelId]] slot is null after this step, it will be populated during the RTCSctpTransport connected procedure.
Let transport be connection.[[SctpTransport]].

If the [[DataChannelId]] slot is not null, transport is in the "connected" state and [[DataChannelId]] is greater or equal to transport.[[MaxChannels]], throw an OperationError.
If channel is the first RTCDataChannel created on connection, update the negotiation-needed flag for connection.
Append channel to connection.[[DataChannels]].
Return channel and continue the following steps in parallel.
Create channel's associated underlying data transport and configure it according to the relevant properties of channel.

The RTCSctpTransport interface allows an application access to information about the SCTP data channels tied to a particular SCTP association.

To create an RTCSctpTransport with an initial state, initialState, run the following steps:

Let transport be a new RTCSctpTransport object.
Let transport have a [[SctpTransportState]] internal slot initialized to initialState.
Let transport have a [[MaxMessageSize]] internal slot and run the steps labeled update the data max message size to initialize it.
Let transport have a [[MaxChannels]] internal slot initialized to null.
Return transport.

To update the data max message size of an RTCSctpTransport run the following steps:

Let transport be the RTCSctpTransport object to be updated.
Let remoteMaxMessageSize be the value of the max-message-size SDP attribute read from the remote description, as described in [RFC8841] (section 6), or 65536 if the attribute is missing.
Let canSendSize be the number of bytes that this client can send (i.e. the size of the local send buffer) or 0 if the implementation can handle messages of any size.
If both remoteMaxMessageSize and canSendSize are 0, set [[MaxMessageSize]] to the positive Infinity value.
Else, if either remoteMaxMessageSize or canSendSize is 0, set [[MaxMessageSize]] to the larger of the two.
Else, set [[MaxMessageSize]] to the smaller of remoteMaxMessageSize or canSendSize.

Once an SCTP transport is connected, meaning the SCTP association of an RTCSctpTransport has been established, the user agent MUST queue a task that runs the following steps:

Let transport be the RTCSctpTransport object.
Let connection be the RTCPeerConnection object associated with transport.
Set [[MaxChannels]] to the minimum of the negotiated amount of incoming and outgoing SCTP streams.
For each of connection's RTCDataChannel:
1. Let channel be the RTCDataChannel object.
2. If channel.[[DataChannelId]] is null, initialize [[DataChannelId]] to the value generated by the underlying sctp data channel, according to [RFC8832].
3. If channel.[[DataChannelId]] is greater or equal to transport.[[MaxChannels]], or the previous step failed to assign an id, close the channel due to a failure. Otherwise, announce the channel as open.
Fire an event named statechange at transport.

Note
This event is fired before the open events fired by announcing the channel as open; the open events are fired from a separate queued task.

[Exposed=Window]
interface RTCSctpTransport : EventTarget {
  readonly attribute RTCDtlsTransport transport;
  readonly attribute RTCSctpTransportState state;
  readonly attribute unrestricted double maxMessageSize;
  readonly attribute unsigned short? maxChannels;
  attribute EventHandler onstatechange;
};

transport of type RTCDtlsTransport, readonly: The transport over which all SCTP packets for data channels will be sent and received.
state of type RTCSctpTransportState, readonly: The current state of the SCTP transport. On getting, this attribute MUST return the value of the [[SctpTransportState]] slot.
maxMessageSize of type unrestricted double, readonly: The maximum size of data that can be passed to RTCDataChannel's send() method. The attribute MUST, on getting, return the value of the [[MaxMessageSize]] slot.
maxChannels of type unsigned short , readonly, nullable: The maximum amount of RTCDataChannel's that can be used simultaneously. The attribute MUST, on getting, return the value of the [[MaxChannels]] slot.

Note
This attribute's value will be null until the SCTP transport goes into the "connected" state.
onstatechange of type EventHandler: The event type of this event handler is statechange.

RTCSctpTransportState indicates the state of the SCTP transport.

enum RTCSctpTransportState {
  "connecting",
  "connected",
  "closed"
};

`RTCSctpTransportState` Enumeration description
Enum value	Description
`connecting`	The `RTCSctpTransport` is in the process of negotiating an association. This is the initial state of the [[SctpTransportState]] slot when an `RTCSctpTransport` is created.
`connected`	When the negotiation of an association is completed, a task is queued to update the [[SctpTransportState]] slot to "`connected`".
`closed`	A task is queued to update the [[SctpTransportState]] slot to "`closed`" when: a SHUTDOWN or ABORT chunk is received. the SCTP association has been closed intentionally, such as by closing the peer connection or applying a remote description that rejects data or changes the SCTP port. the underlying DTLS association has transitioned to "`closed`" state. Note that the last transition is logical due to the fact that an SCTP association requires an established DTLS connection - [RFC8261] section 6.1 specifies that SCTP over DTLS is single-homed - and that no way of of switching to an alternate transport is defined in this API.

The RTCDataChannel interface represents a bi-directional data channel between two peers. An RTCDataChannel is created via a factory method on an RTCPeerConnection object. The messages sent between the browsers are described in [RFC8831] and [RFC8832].

There are two ways to establish a connection with RTCDataChannel. The first way is to simply create an RTCDataChannel at one of the peers with the negotiated RTCDataChannelInit dictionary member unset or set to its default value false. This will announce the new channel in-band and trigger an RTCDataChannelEvent with the corresponding RTCDataChannel object at the other peer. The second way is to let the application negotiate the RTCDataChannel. To do this, create an RTCDataChannel object with the negotiated RTCDataChannelInit dictionary member set to true, and signal out-of-band (e.g. via a web server) to the other side that it SHOULD create a corresponding RTCDataChannel with the negotiated RTCDataChannelInit dictionary member set to true and the same id. This will connect the two separately created RTCDataChannel objects. The second way makes it possible to create channels with asymmetric properties and to create channels in a declarative way by specifying matching ids.

Each RTCDataChannel has an associated underlying data transport that is used to transport actual data to the other peer. In the case of SCTP data channels utilizing an RTCSctpTransport (which represents the state of the SCTP association), the underlying data transport is the SCTP stream pair. The transport properties of the underlying data transport, such as in order delivery settings and reliability mode, are configured by the peer as the channel is created. The properties of a channel cannot change after the channel has been created. The actual wire protocol between the peers is specified by the WebRTC DataChannel Protocol specification [RFC8831].

An RTCDataChannel can be configured to operate in different reliability modes. A reliable channel ensures that the data is delivered at the other peer through retransmissions. An unreliable channel is configured to either limit the number of retransmissions ( maxRetransmits ) or set a time during which transmissions (including retransmissions) are allowed ( maxPacketLifeTime ). These properties can not be used simultaneously and an attempt to do so will result in an error. Not setting any of these properties results in a reliable channel.

An RTCDataChannel, created with createDataChannel or dispatched via an RTCDataChannelEvent, MUST initially be in the "connecting" state. When the RTCDataChannel object's underlying data transport is ready, the user agent MUST announce the RTCDataChannel as open.

To create an RTCDataChannel, run the following steps:

Let channel be a newly created RTCDataChannel object.
Let channel have a [[ReadyState]] internal slot initialized to "connecting".
Let channel have a [[BufferedAmount]] internal slot initialized to 0.
Let channel have internal slots named [[DataChannelLabel]], [[Ordered]], [[MaxPacketLifeTime]], [[MaxRetransmits]], [[DataChannelProtocol]], [[Negotiated]], and [[DataChannelId]].
Let channel have a [[IsTransferable]] internal slot initialized to true.
Queue a task to run the following step:
1. Set channel.[[IsTransferable]] to false.
This task needs to run before any task enqueued by the receiving messages on a data channel algorithm for channel. This ensures that no message is lost during the transfer of a RTCDataChannel.
Return channel.

When the user agent is to announce an RTCDataChannel as open, the user agent MUST queue a task to run the following steps:

If the associated RTCPeerConnection object's [[IsClosed]] slot is true, abort these steps.
Let channel be the RTCDataChannel object to be announced.
If channel.[[ReadyState]] is "closing" or "closed", abort these steps.
Set channel.[[ReadyState]] to "open".
Fire an event named open at channel.

When an underlying data transport is to be announced (the other peer created a channel with negotiated unset or set to false), the user agent of the peer that did not initiate the creation process MUST queue a task to run the following steps:

Let connection be the RTCPeerConnection object associated with the underlying data transport.
If connection.[[IsClosed]] is true, abort these steps.
Create an RTCDataChannel, channel.
Let configuration be an information bundle received from the other peer as a part of the process to establish the underlying data transport described by the WebRTC DataChannel Protocol specification [RFC8832].
Initialize channel.[[DataChannelLabel]], [[Ordered]], [[MaxPacketLifeTime]], [[MaxRetransmits]], [[DataChannelProtocol]], and [[DataChannelId]] internal slots to the corresponding values in configuration.
Initialize channel.[[Negotiated]] to false.
Append channel to connection.[[DataChannels]].
Set channel.[[ReadyState]] to "open" (but do not fire the open event, yet).

Note
This allows to start sending messages inside of the datachannel event handler prior to the open event being fired.
Fire an event named datachannel using the RTCDataChannelEvent interface with the channel attribute set to channel at connection.
Announce the data channel as open.

Candidate Correction 38:Prevent GC of non-closed RTCDataChannels (PR #2902)

~~6.2.4 Closing procedure~~

An RTCDataChannel object's underlying data transport may be torn down in a non-abrupt manner by running the closing procedure. When that happens the user agent MUST queue a task to run the following steps:

Let channel be the RTCDataChannel object whose underlying data transport was closed.
Let connection be the RTCPeerConnection object associated with channel.
Remove channel from connection.[[DataChannels]].
Unless the procedure was initiated by channel.close, set channel.[[ReadyState]] to "closing" and fire an event named closing at channel.
Run the following steps ~~in parallel~~in parallel:
1. Finish sending all currently pending messages of the channel.
2. Follow the closing procedure defined for the channel's underlying data transport :
  1. In the case of an SCTP-based transport, follow [RFC8831], section 6.7.
3. ~~Render~~ Close the channel's data transport ~~closed~~ by following the associated procedure.

When an RTCDataChannel object's underlying data transport has been closed, the user agent MUST queue a task to run the following steps:

Let channel be the RTCDataChannel object whose underlying data transport was closed.
If channel.[[ReadyState]] is "closed", abort these steps.
Set channel.[[ReadyState]] to "closed".
Remove channel from connection.[[DataChannels]] if it is still there.
If the transport was closed with an error, fire an event named error using the RTCErrorEvent interface with its errorDetail attribute set to "sctp-failure" at channel.
Fire an event named close at channel.

The RTCDataChannel transfer steps, given value and dataHolder, are:

If value.[[IsTransferable]] is false, throw a DataCloneError DOMException.
Set dataHolder.[[ReadyState]] to value.[[ReadyState]].
Set dataHolder.[[DataChannelLabel]] to value.[[DataChannelLabel]].
Set dataHolder.[[Ordered]] to value.[[Ordered]].
Set dataHolder.[[MaxPacketLifeTime]] to value..[[MaxPacketLifeTime]]
Set dataHolder.[[MaxRetransmits]] to value.[[MaxRetransmits]].
Set dataHolder.[[DataChannelProtocol]] to value.[[DataChannelProtocol]].
Set dataHolder.[[Negotiated]] to value.[[Negotiated]].
Set dataHolder.[[DataChannelId]] to value.[[DataChannelId]].
Set dataHolder’s underlying data transport to value underlying data transport.
Set value.[[IsTransferable]] to false.
Set value.[[ReadyState]] to "closed".

The RTCDataChannel transfer-receiving steps, given dataHolder and channel, are:

Initialize channel.[[ReadyState]] to dataHolder.[[ReadyState]].
Initialize channel.[[DataChannelLabel]] to dataHolder.[[DataChannelLabel]].
Initialize channel.[[Ordered]] to dataHolder.[[Ordered]].
Initialize channel.[[MaxPacketLifeTime]] to dataHolder.[[MaxPacketLifeTime]].
Initialize channel.[[MaxRetransmits]] to dataHolder.[[MaxRetransmits]].
Initialize channel.[[DataChannelProtocol]] to dataHolder.[[DataChannelProtocol]].
Initialize channel.[[Negotiated]] to dataHolder.[[Negotiated]].
Initialize channel.[[DataChannelId]] to dataHolder.[[DataChannelId]].
Initialize channel’s underlying data transport to dataHolder’s underlying data transport.

The above steps do not need to transfer [[BufferedAmount]] as its value will always be equal to 0. The reason is an RTCDataChannel can be transferred only if its send() algorithm was not called prior the transfer.

If the underlying data transport is closed at the time of the transfer-receiving steps, the RTCDataChannel object will be closed by running the announcing a data channel as closed algorithm immediately after the transfer-receiving steps.

In some cases, the user agent may be unable to create an RTCDataChannel 's underlying data transport. For example, the data channel's id may be outside the range negotiated by the [RFC8831] implementations in the SCTP handshake. When the user agent determines that an RTCDataChannel's underlying data transport cannot be created, the user agent MUST queue a task to run the following steps:

Let channel be the RTCDataChannel object for which the user agent could not create an underlying data transport.
Set channel.[[ReadyState]] to "closed".
Fire an event named error using the RTCErrorEvent interface with the errorDetail attribute set to "data-channel-failure" at channel.
Fire an event named close at channel.

When an RTCDataChannel message has been received via the underlying data transport with type type and data rawData, the user agent MUST queue a task to run the following steps:

Let channel be the RTCDataChannel object for which the user agent has received a message.
Let connection be the RTCPeerConnection object associated with channel.
If channel.[[ReadyState]] is not "open", abort these steps and discard rawData.
Execute the sub step by switching on type and channel.binaryType:
- If type indicates that rawData is a string:
  
  Let data be a DOMString that represents the result of decoding rawData as UTF-8.
- If type indicates that rawData is binary and binaryType is "blob":
  
  Let data be a new Blob object containing rawData as its raw data source.
- If type indicates that rawData is binary and binaryType is "arraybuffer":
  
  Let data be a new ArrayBuffer object containing rawData as its raw data source.
Fire an event named message using the MessageEvent interface with its origin attribute initialized to the serialization of an origin of connection.[[DocumentOrigin]], and the data attribute initialized to data at channel.

[Exposed=(Window,DedicatedWorker), Transferable]
interface RTCDataChannel : EventTarget {
  readonly attribute USVString label;
  readonly attribute boolean ordered;
  readonly attribute unsigned short? maxPacketLifeTime;
  readonly attribute unsigned short? maxRetransmits;
  readonly attribute USVString protocol;
  readonly attribute boolean negotiated;
  readonly attribute unsigned short? id;
  readonly attribute RTCDataChannelState readyState;
  readonly attribute unsigned long bufferedAmount;
  [EnforceRange] attribute unsigned long bufferedAmountLowThreshold;
  attribute EventHandler onopen;
  attribute EventHandler onbufferedamountlow;
  attribute EventHandler onerror;
  attribute EventHandler onclosing;
  attribute EventHandler onclose;
  undefined close();
  attribute EventHandler onmessage;
  attribute BinaryType binaryType;
  undefined send(USVString data);
  undefined send(Blob data);
  undefined send(ArrayBuffer data);
  undefined send(ArrayBufferView data);
};

label of type USVString, readonly

The label attribute represents a label that can be used to distinguish this RTCDataChannel object from other RTCDataChannel objects. Scripts are allowed to create multiple RTCDataChannel objects with the same label. On getting, the attribute MUST return the value of the [[DataChannelLabel]] slot.

ordered of type boolean, readonly

The ordered attribute returns true if the RTCDataChannel is ordered, and false if out of order delivery is allowed. On getting, the attribute MUST return the value of the [[Ordered]] slot.

maxPacketLifeTime of type unsigned short, readonly, nullable

The maxPacketLifeTime attribute returns the length of the time window (in milliseconds) during which transmissions and retransmissions may occur in unreliable mode. On getting, the attribute MUST return the value of the [[MaxPacketLifeTime]] slot.

maxRetransmits of type unsigned short, readonly, nullable

The maxRetransmits attribute returns the maximum number of retransmissions that are attempted in unreliable mode. On getting, the attribute MUST return the value of the [[MaxRetransmits]] slot.

protocol of type USVString, readonly

The protocol attribute returns the name of the sub-protocol used with this RTCDataChannel. On getting, the attribute MUST return the value of the [[DataChannelProtocol]] slot.

negotiated of type boolean, readonly

The negotiated attribute returns true if this RTCDataChannel was negotiated by the application, or false otherwise. On getting, the attribute MUST return the value of the [[Negotiated]] slot.

id of type unsigned short, readonly, nullable

The id attribute returns the ID for this RTCDataChannel. The value is initially null, which is what will be returned if the ID was not provided at channel creation time, and the DTLS role of the SCTP transport has not yet been negotiated. Otherwise, it will return the ID that was either selected by the script or generated by the user agent according to [RFC8832]. After the ID is set to a non-null value, it will not change. On getting, the attribute MUST return the value of the [[DataChannelId]] slot.

readyState of type RTCDataChannelState, readonly

The readyState attribute represents the state of the RTCDataChannel object. On getting, the attribute MUST return the value of the [[ReadyState]] slot.

bufferedAmount of type unsigned long, readonly

The bufferedAmount attribute MUST, on getting, return the value of the [[BufferedAmount]] slot. The attribute exposes the number of bytes of application data (UTF-8 text and binary data) that have been queued using send(). Even though the data transmission can occur in parallel, the returned value MUST NOT be decreased before the current task yielded back to the event loop to prevent race conditions. The value does not include framing overhead incurred by the protocol, or buffering done by the operating system or network hardware. The value of the [[BufferedAmount]] slot will only increase with each call to the send() method as long as the [[ReadyState]] slot is "open"; however, the slot does not reset to zero once the channel closes. When the underlying data transport sends data from its queue, the user agent MUST queue a task that reduces [[BufferedAmount]] with the number of bytes that was sent.

bufferedAmountLowThreshold of type unsigned long

The bufferedAmountLowThreshold attribute sets the threshold at which the bufferedAmount is considered to be low. When the bufferedAmount decreases from above this threshold to equal or below it, the bufferedamountlow event fires. The bufferedAmountLowThreshold is initially zero on each new RTCDataChannel, but the application may change its value at any time.

onopen of type EventHandler

The event type of this event handler is open.

onbufferedamountlow of type EventHandler

The event type of this event handler is bufferedamountlow.

onerror of type EventHandler

The event type of this event handler is RTCErrorEvent. errorDetail contains "sctp-failure", sctpCauseCode contains the SCTP Cause Code value, and message contains the SCTP Cause-Specific-Information, possibly with additional text.

onclosing of type EventHandler

The event type of this event handler is closing.

onclose of type EventHandler

The event type of this event handler is close.

onmessage of type EventHandler

The event type of this event handler is message.

binaryType of type BinaryType

The binaryType attribute returns the value to which it was last set. When an RTCDataChannel object is created, the binaryType attribute MUST be initialized to the string "arraybuffer".

This attribute controls how binary data is exposed to scripts. See Web Socket's binaryType.

close()

Closes the RTCDataChannel. It may be called regardless of whether the RTCDataChannel object was created by this peer or the remote peer.

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

Let channel be the RTCDataChannel object which is about to be closed.
If channel.[[ReadyState]] is "closing" or "closed", then abort these steps.
Set channel.[[ReadyState]] to "closing".
If the closing procedure has not started yet, start it.

send

Run the steps described by the send() algorithm with argument type string object.

send

Run the steps described by the send() algorithm with argument type Blob object.

send

Run the steps described by the send() algorithm with argument type ArrayBuffer object.

send

Run the steps described by the send() algorithm with argument type ArrayBufferView object.

The send() method is overloaded to handle different data argument types. When any version of the method is called, the user agent MUST run the following steps:

Let channel be the RTCDataChannel object on which data is to be sent.
Set channel.[[IsTransferable]] to false.
If channel.[[ReadyState]] is not "open", throw an InvalidStateError.
Execute the sub step that corresponds to the type of the methods argument:
- string object:
  
  Let data be a byte buffer that represents the result of encoding the method's argument as UTF-8.
- Blob object:
  
  Let data be the raw data represented by the Blob object.
  
  Note
  Although the actual retrieval of data from a Blob object can happen asynchronously, the user agent will make sure to queue the data on the channel's underlying data transport in the same order as the send method is called. The byte size of data needs to be known synchronously.
- ArrayBuffer object:
  
  Let data be the data stored in the buffer described by the ArrayBuffer object.
- ArrayBufferView object:
  
  Let data be the data stored in the section of the buffer described by the ArrayBuffer object that the ArrayBufferView object references.
Note
Any data argument type this method has not been overloaded with will result in a TypeError. This includes null and undefined.
If the byte size of data exceeds the value of maxMessageSize on channel's associated RTCSctpTransport, throw a TypeError.
Queue data for transmission on channel's underlying data transport. If queuing data is not possible because not enough buffer space is available, throw an OperationError.

Note
The actual transmission of data occurs in parallel. If sending data leads to an SCTP-level error, the application will be notified asynchronously through onerror.
Increase the value of the [[BufferedAmount]] slot by the byte size of data.

dictionary RTCDataChannelInit {
  boolean ordered = true;
  [EnforceRange] unsigned short maxPacketLifeTime;
  [EnforceRange] unsigned short maxRetransmits;
  USVString protocol = "";
  boolean negotiated = false;
  [EnforceRange] unsigned short id;
};

ordered of type boolean, defaulting to true: If set to false, data is allowed to be delivered out of order. The default value of true, guarantees that data will be delivered in order.
maxPacketLifeTime of type unsigned short: Limits the time (in milliseconds) during which the channel will transmit or retransmit data if not acknowledged. This value may be clamped if it exceeds the maximum value supported by the user agent.
maxRetransmits of type unsigned short: Limits the number of times a channel will retransmit data if not successfully delivered. This value may be clamped if it exceeds the maximum value supported by the user agent.
protocol of type USVString, defaulting to "": Subprotocol name used for this channel.
negotiated of type boolean, defaulting to false: The default value of false tells the user agent to announce the channel in-band and instruct the other peer to dispatch a corresponding RTCDataChannel object. If set to true, it is up to the application to negotiate the channel and create an RTCDataChannel object with the same id at the other peer.

Note
If set to true, the application must also take care to not send a message until the other peer has created a data channel to receive it. Receiving a message on an SCTP stream with no associated data channel is undefined behavior, and it may be silently dropped. This will not be possible as long as both endpoints create their data channel before the first offer/answer exchange is complete.
id of type unsigned short: Sets the channel ID when negotiated is true. Ignored when negotiated is false.

enum RTCDataChannelState {
  "connecting",
  "open",
  "closing",
  "closed"
};

`RTCDataChannelState` Enumeration description
Enum value	Description
`connecting`	The user agent is attempting to establish the underlying data transport. This is the initial state of an `RTCDataChannel` object, whether created with `createDataChannel`, or dispatched as a part of an `RTCDataChannelEvent`.
`open`	The underlying data transport is established and communication is possible.
`closing`	The procedure to close down the underlying data transport has started.
`closed`	The underlying data transport has been `closed` or could not be established.

The datachannel event uses the RTCDataChannelEvent interface.

[Exposed=Window]
interface RTCDataChannelEvent : Event {
  constructor(DOMString type, RTCDataChannelEventInit eventInitDict);
  readonly attribute RTCDataChannel channel;
};

RTCDataChannelEvent.constructor()

channel of type RTCDataChannel, readonly: The channel attribute represents the RTCDataChannel object associated with the event.

dictionary RTCDataChannelEventInit : EventInit {
  required RTCDataChannel channel;
};

channel of type RTCDataChannel, required: The RTCDataChannel object to be announced by the event.

An RTCDataChannel object MUST not be garbage collected if its

[[ReadyState]] slot is "connecting" and at least one event listener is registered for open events, message events, error events, closing events, or close events.
[[ReadyState]] slot is "open" and at least one event listener is registered for message events, error events, closing events, or close events.
[[ReadyState]] slot is "closing" and at least one event listener is registered for error events, or close events.
underlying data transport is established and data is queued to be transmitted.

This section describes an interface on RTCRtpSender to send DTMF (phone keypad) values across an RTCPeerConnection. Details of how DTMF is sent to the other peer are described in [RFC7874].

The Peer-to-peer DTMF API extends the RTCRtpSender interface as described below.

          partial interface RTCRtpSender {
  readonly attribute RTCDTMFSender? dtmf;
};

dtmf of type RTCDTMFSender, readonly, nullable: On getting, the dtmf attribute returns the value of the [[Dtmf]] internal slot, which represents a RTCDTMFSender which can be used to send DTMF, or null if unset. The [[Dtmf]] internal slot is set when the kind of an RTCRtpSender's [[SenderTrack]] is "audio".

To create an RTCDTMFSender, the user agent MUST run the following steps:

Let dtmf be a newly created RTCDTMFSender object.
Let dtmf have a [[Duration]] internal slot.
Let dtmf have a [[InterToneGap]] internal slot.
Let dtmf have a [[ToneBuffer]] internal slot.

[Exposed=Window]
interface RTCDTMFSender : EventTarget {
  undefined insertDTMF(DOMString tones, optional unsigned long duration = 100, optional unsigned long interToneGap = 70);
  attribute EventHandler ontonechange;
  readonly attribute boolean canInsertDTMF;
  readonly attribute DOMString toneBuffer;
};

ontonechange of type EventHandler: The event type of this event handler is tonechange.
canInsertDTMF of type boolean, readonly: Whether the RTCDTMFSender dtmfSender is capable of sending DTMF. On getting, the user agent MUST return the result of running determine if DTMF can be sent for dtmfSender.
toneBuffer of type DOMString, readonly: The toneBuffer attribute MUST return a list of the tones remaining to be played out. For the syntax, content, and interpretation of this list, see insertDTMF.

insertDTMF

An RTCDTMFSender object's insertDTMF method is used to send DTMF tones.

The tones parameter is treated as a series of characters. The characters 0 through 9, A through D, #, and * generate the associated DTMF tones. The characters a to d MUST be normalized to uppercase on entry and are equivalent to A to D. As noted in [RTCWEB-AUDIO] Section 3, support for the characters 0 through 9, A through D, #, and * are required. The character ',' MUST be supported, and indicates a delay of 2 seconds before processing the next character in the tones parameter. All other characters (and only those other characters) MUST be considered unrecognized.

The duration parameter indicates the duration in ms to use for each character passed in the tones parameters. The duration cannot be more than 6000 ms or less than 40 ms. The default duration is 100 ms for each tone.

The interToneGap parameter indicates the gap between tones in ms. The user agent clamps it to at least 30 ms and at most 6000 ms. The default value is 70 ms.

The browser MAY increase the duration and interToneGap times to cause the times that DTMF start and stop to align with the boundaries of RTP packets but it MUST not increase either of them by more than the duration of a single RTP audio packet.

When the insertDTMF() method is invoked, the user agent MUST run the following steps:

Let sender be the RTCRtpSender used to send DTMF.
Let transceiver be the RTCRtpTransceiver object associated with sender.
Let dtmf be the RTCDTMFSender associated with sender.
If determine if DTMF can be sent for dtmf returns false, throw an InvalidStateError.
Let tones be the method's first argument.
Let duration be the method's second argument.
Let interToneGap be the method's third argument.
If tones contains any unrecognized characters, throw an InvalidCharacterError.
Set the object's [[ToneBuffer]] slot to tones.
Set dtmf.[[Duration]] to the value of duration.
Set dtmf.[[InterToneGap]] to the value of interToneGap.
If the value of duration is less than 40 ms, set dtmf.[[Duration]] to 40 ms.
If the value of duration parameter is greater than 6000 ms, set dtmf.[[Duration]] to 6000 ms.
If the value of interToneGap is less than 30 ms, set dtmf.[[InterToneGap]] to 30 ms.
If the value of interToneGap is greater than 6000 ms, set dtmf.[[InterToneGap]] to 6000 ms.
If [[ToneBuffer]] slot is an empty string, abort these steps.
If a task to run the DTMF playout task steps is scheduled to be run, abort these steps; otherwise queue a task that runs the following DTMF playout task steps:
1. Candidate Correction 33:Determine if DTMF can be sent inside queued playout task (PR #2861)
2. ~~If transceiver.[[CurrentDirection]]~~If determine if DTMF can be sent ~~is neither "sendrecv" nor "sendonly"~~for dtmf returns false, abort these steps.
3. If the [[ToneBuffer]] slot contains the empty string, fire an event named tonechange using the RTCDTMFToneChangeEvent interface with the tone attribute set to an empty string at the RTCDTMFSender object and abort these steps.
4. Remove the first character from the [[ToneBuffer]] slot and let that character be tone.
5. If tone is "," delay sending tones for 2000 ms on the associated RTP media stream, and queue a task to be executed in 2000 ms from now that runs the DTMF playout task steps.
6. If tone is not "," start playout of tone for [[Duration]] ms on the associated RTP media stream, using the appropriate codec, then queue a task to be executed in [[Duration]] + [[InterToneGap]] ms from now that runs the DTMF playout task steps.
7. Fire an event named tonechange using the RTCDTMFToneChangeEvent interface with the tone attribute set to tone at the RTCDTMFSender object.

Since insertDTMF replaces the tone buffer, in order to add to the DTMF tones being played, it is necessary to call insertDTMF with a string containing both the remaining tones (stored in the [[ToneBuffer]] slot) and the new tones appended together. Calling insertDTMF with an empty tones parameter can be used to cancel all tones queued to play after the currently playing tone.

To determine if DTMF can be sent for an RTCDTMFSender instance dtmfSender, the user agent MUST run the following steps:

Let sender be the RTCRtpSender associated with dtmfSender.
Let transceiver be the RTCRtpTransceiver associated with sender.
Let connection be the RTCPeerConnection associated with transceiver.
If connection's RTCPeerConnectionState is not "connected" return false.
If transceiver.[[Stopping]] is true return false.
If sender.[[SenderTrack]] is null return false.
If transceiver.[[CurrentDirection]] is neither "sendrecv" nor "sendonly" return false.
If sender.[[SendEncodings]][0].active is false return false.
If no codec with mimetype "audio/telephone-event" has been negotiated for sending with this sender, return false.
Otherwise, return true.

The tonechange event uses the RTCDTMFToneChangeEvent interface.

[Exposed=Window]
interface RTCDTMFToneChangeEvent : Event {
  constructor(DOMString type, optional RTCDTMFToneChangeEventInit eventInitDict = {});
  readonly attribute DOMString tone;
};

RTCDTMFToneChangeEvent.constructor()

tone of type DOMString, readonly: The tone attribute contains the character for the tone (including ",") that has just begun playout (see insertDTMF ). If the value is the empty string, it indicates that the [[ToneBuffer]] slot is an empty string and that the previous tones have completed playback.

          dictionary RTCDTMFToneChangeEventInit : EventInit {
  DOMString tone = "";
};

tone of type DOMString, defaulting to "": The tone attribute contains the character for the tone (including ",") that has just begun playout (see insertDTMF ). If the value is the empty string, it indicates that the [[ToneBuffer]] slot is an empty string and that the previous tones have completed playback.

The basic statistics model is that the browser maintains a set of statistics for monitored objects, in the form of stats objects.

A group of related objects may be referenced by a selector. The selector may, for example, be a MediaStreamTrack. For a track to be a valid selector, it MUST be a MediaStreamTrack that is sent or received by the RTCPeerConnection object on which the stats request was issued. The calling Web application provides the selector to the getStats() method and the browser emits (in the JavaScript) a set of statistics that are relevant to the selector, according to the stats selection algorithm. Note that that algorithm takes the sender or receiver of a selector.

The statistics returned in stats objects are designed in such a way that repeated queries can be linked by the RTCStats id dictionary member. Thus, a Web application can make measurements over a given time period by requesting measurements at the beginning and end of that period.

With a few exceptions, monitored objects, once created, exist for the duration of their associated RTCPeerConnection. This ensures statistics from them are available in the result from getStats() even past the associated peer connection being closed.

Only a few monitored objects have shorter lifetimes. Statistics from these objects are no longer available in subsequent getStats() results. The object descriptions in [WEBRTC-STATS] describe when these monitored objects are deleted.

The Statistics API extends the RTCPeerConnection interface as described below.

          partial interface RTCPeerConnection {
  Promise<RTCStatsReport> getStats(optional MediaStreamTrack? selector = null);
};

getStats

Gathers stats for the given selector and reports the result asynchronously.

When the getStats() method is invoked, the user agent MUST run the following steps:

Let selectorArg be the method's first argument.
Let connection be the RTCPeerConnection object on which the method was invoked.
If selectorArg is null, let selector be null.
If selectorArg is a MediaStreamTrack let selector be an RTCRtpSender or RTCRtpReceiver on connection which track attribute matches selectorArg. If no such sender or receiver exists, or if more than one sender or receiver fit this criteria, return a promise rejected with a newly created InvalidAccessError.
Let p be a new promise.
Run the following steps in parallel:
1. Gather the stats indicated by selector according to the stats selection algorithm.
2. Queue a global task on the networking task source given the current realm's global object as global to resolve p with the resulting RTCStatsReport object, containing the gathered stats.
Return p.

The getStats() method delivers a successful result in the form of an RTCStatsReport object. An RTCStatsReport object is a map between strings that identify the inspected objects (id attribute in RTCStats instances), and their corresponding RTCStats-derived dictionaries.

An RTCStatsReport may be composed of several RTCStats-derived dictionaries, each reporting stats for one underlying object that the implementation thinks is relevant for the selector. One achieves the total for the selector by summing over all the stats of a certain type; for instance, if an RTCRtpSender uses multiple SSRCs to carry its track over the network, the RTCStatsReport may contain one RTCStats-derived dictionary per SSRC (which can be distinguished by the value of the ssrc stats attribute).

[Exposed=Window]
interface RTCStatsReport {
  readonly maplike<DOMString, object>;
};

Use these to retrieve the various dictionaries descended from RTCStats that this stats report is composed of. The set of supported property names [WEBIDL] is defined as the ids of all the RTCStats-derived dictionaries that have been generated for this stats report.

An RTCStats dictionary represents the stats object constructed by inspecting a specific monitored object. The RTCStats dictionary is a base type that specifies as set of default attributes, such as timestamp and type. Specific stats are added by extending the RTCStats dictionary.

Note that while stats names are standardized, any given implementation may be using experimental values or values not yet known to the Web application. Thus, applications MUST be prepared to deal with unknown stats.

Statistics need to be synchronized with each other in order to yield reasonable values in computation; for instance, if bytesSent and packetsSent are both reported, they both need to be reported over the same interval, so that "average packet size" can be computed as "bytes / packets" - if the intervals are different, this will yield errors. Thus implementations MUST return synchronized values for all stats in an RTCStats-derived dictionary.

dictionary RTCStats {
  required DOMHighResTimeStamp timestamp;
  required RTCStatsType type;
  required DOMString id;
};

timestamp of type DOMHighResTimeStamp

Candidate Correction 50:Use Performance.timeOrigin + Performance.now() for stats timestamps (PR #3005)

~~The timestamp, of type~~ Timestamps are expressed with DOMHighResTimeStamp, ~~associated with this object. The time is relative to the UNIX epoch (Jan 1~~[HIGHRES-TIME], ~~1970,~~and are defined as Performance.timeOrigin ~~UTC)~~+ Performance.now() at the time the information is collected. For statistics that came from a remote source (e.g., from received RTCP packets), timestamp represents the time at which the information arrived at the local endpoint. The remote timestamp can be found in an additional field in an RTCStats-derived dictionary, if applicable.

type of type RTCStatsType

The type of this object.

The type attribute MUST be initialized to the name of the most specific type this RTCStats dictionary represents.

id of type DOMString

A unique id that is associated with the object that was inspected to produce this RTCStats object. Two RTCStats objects, extracted from two different RTCStatsReport objects, MUST have the same id if they were produced by inspecting the same underlying object.

Stats ids MUST NOT be predictable by an application. This prevents applications from depending on a particular user agent's way of generating ids, since this prevents an application from getting stats objects by their id unless they have already read the id of that specific stats object.

User agents are free to pick any format for the id as long as it meets the requirements above.

Note

A user agent can turn a predictably generated string into an unpredictable string using a hash function, as long as it uses a salt that is unique to the peer connection. This allows an implementation to have predictable ids internally, which may make it easier to guarantee that stats objects have stable ids across getStats() calls.

The set of valid values for RTCStatsType, and the dictionaries derived from RTCStats that they indicate, are documented in [WEBRTC-STATS].

The stats selection algorithm is as follows:

Let result be an empty RTCStatsReport.
If selector is null, gather stats for the whole connection, add them to result, return result, and abort these steps.
If selector is an RTCRtpSender, gather stats for and add the following objects to result:
- All RTCOutboundRtpStreamStats objects representing RTP streams being sent by selector.
- All stats objects referenced directly or indirectly by the RTCOutboundRtpStreamStats objects added.
If selector is an RTCRtpReceiver, gather stats for and add the following objects to result:
- All RTCInboundRtpStreamStats objects representing RTP streams being received by selector.
- All stats objects referenced directly or indirectly by the RTCInboundRtpStreamStats added.
Return result.

The stats listed in [WEBRTC-STATS] are intended to cover a wide range of use cases. Not all of them have to be implemented by every WebRTC implementation.

An implementation MUST support generating statistics of the following types when the corresponding objects exist on a RTCPeerConnection, with the fields that are listed when they are valid for that object in addition to the generic fields defined in the RTCStats dictionary:

`RTCStatsType`	Dictionary	Fields
"`codec`"	`RTCCodecStats`	`payloadType`, `mimeType`, `clockRate`, `channels`, `sdpFmtpLine`
"`inbound-rtp`"	`RTCRtpStreamStats`	`ssrc`, `kind`, `transportId`, `codecId`
	`RTCReceivedRtpStreamStats`	`packetsReceived`, `packetsLost`, `jitter`,
	`RTCInboundRtpStreamStats`	`trackIdentifier`, `remoteId`, `framesDecoded`, `framesDropped` `nackCount`, `framesReceived`, `bytesReceived`, `totalAudioEnergy`, `totalSamplesDuration` `packetsDiscarded`,
"`outbound-rtp`"	`RTCRtpStreamStats`	`ssrc`, `kind`, `transportId`, `codecId`
	`RTCSentRtpStreamStats`	`packetsSent`, `bytesSent`
	`RTCOutboundRtpStreamStats`	`remoteId`, `framesEncoded`, `nackCount`, `framesSent`
"`remote-inbound-rtp`"	`RTCRtpStreamStats`	`ssrc`, `kind`, `transportId`, `codecId`
	`RTCReceivedRtpStreamStats`	`packetsReceived`, `packetsLost`, `jitter`
	`RTCRemoteInboundRtpStreamStats`	`localId`, `roundTripTime`
"`remote-outbound-rtp`"	`RTCRtpStreamStats`	`ssrc`, `kind`, `transportId`, `codecId`
	`RTCSentRtpStreamStats`	`packetsSent`, `bytesSent`
	`RTCRemoteOutboundRtpStreamStats`	`localId`, `remoteTimestamp`
"`media-source`"	`RTCMediaSourceStats`	`trackIdentifier`, `kind`
	`RTCAudioSourceStats`	`totalAudioEnergy`, `totalSamplesDuration` (for audio tracks attached to senders)
	`RTCVideoSourceStats`	`width`, `height`, `framesPerSecond` (for video tracks attached to senders)
"`peer-connection`"	`RTCPeerConnectionStats`	`dataChannelsOpened`, `dataChannelsClosed`
"`data-channel`"	`RTCDataChannelStats`	`label` , `protocol`, `dataChannelIdentifier`, `state`, `messagesSent`, `bytesSent`, `messagesReceived`, `bytesReceived`
"`transport`"	`RTCTransportStats`	`bytesSent`, `bytesReceived`, `selectedCandidatePairId`, `localCertificateId`, `remoteCertificateId`
"`candidate-pair`"	`RTCIceCandidatePairStats`	`transportId`, `localCandidateId`, `remoteCandidateId`, `state`, `nominated`, `bytesSent`, `bytesReceived`, `totalRoundTripTime`, `responsesReceived`, `currentRoundTripTime`
"`local-candidate`"	`RTCIceCandidateStats`	`address`, `port`, `protocol`, `candidateType`, `url`
"`remote-candidate`"	`RTCIceCandidateStats`	`address`, `port`, `protocol`, `candidateType`, `url`
"`certificate`"	`RTCCertificateStats`	`fingerprint`, `fingerprintAlgorithm`, `base64Certificate`, `issuerCertificateId`

An implementation MAY support generating any other statistic defined in [WEBRTC-STATS], and MAY generate statistics that are not documented.

Consider the case where the user is experiencing bad sound and the application wants to determine if the cause of it is packet loss. The following example code might be used:

Example 8

async function gatherStats(pc) {
  try {
    const [sender] = pc.getSenders();
    const baselineReport = await sender.getStats();
    await new Promise(resolve => setTimeout(resolve, aBit)); // wait a bit
    const currentReport = await sender.getStats();

    // compare the elements from the current report with the baseline
    for (const now of currentReport.values()) {
      if (now.type != 'outbound-rtp') continue;

      // get the corresponding stats from the baseline report
      const base = baselineReport.get(now.id);
      if (!base) continue;

      const remoteNow = currentReport.get(now.remoteId);
      const remoteBase = baselineReport.get(base.remoteId);

      const packetsSent = now.packetsSent - base.packetsSent;
      const packetsReceived = remoteNow.packetsReceived -
                              remoteBase.packetsReceived;

      const fractionLost = (packetsSent - packetsReceived) / packetsSent;
      if (fractionLost > 0.3) {
        // if fractionLost is > 0.3, we have probably found the culprit
      }
    }
  } catch (err) {
    console.error(err);
  }
}

The MediaStreamTrack interface, as defined in the [GETUSERMEDIA] specification, typically represents a stream of data of audio or video. One or more MediaStreamTracks can be collected in a MediaStream (strictly speaking, a MediaStream as defined in [GETUSERMEDIA] may contain zero or more MediaStreamTrack objects).

A MediaStreamTrack may be extended to represent a media flow that either comes from or is sent to a remote peer (and not just the local camera, for instance). The extensions required to enable this capability on the MediaStreamTrack object will be described in this section. How the media is transmitted to the peer is described in [RFC8834], [RFC7874], and [RFC8835].

A MediaStreamTrack sent to another peer will appear as one and only one MediaStreamTrack to the recipient. A peer is defined as a user agent that supports this specification. In addition, the sending side application can indicate what MediaStream object(s) the MediaStreamTrack is a member of. The corresponding MediaStream object(s) on the receiver side will be created (if not already present) and populated accordingly.

As also described earlier in this document, the objects RTCRtpSender and RTCRtpReceiver can be used by the application to get more fine grained control over the transmission and reception of MediaStreamTracks.

Channels are the smallest unit considered in the Media Capture and Streams specification. Channels are intended to be encoded together for transmission as, for instance, an RTP payload type. All of the channels that a codec needs to encode jointly MUST be in the same MediaStreamTrack and the codecs SHOULD be able to encode, or discard, all the channels in the track.

The concepts of an input and output to a given MediaStreamTrack apply in the case of MediaStreamTrack objects transmitted over the network as well. A MediaStreamTrack created by an RTCPeerConnection object (as described previously in this document) will take as input the data received from a remote peer. Similarly, a MediaStreamTrack from a local source, for instance a camera via [GETUSERMEDIA], will have an output that represents what is transmitted to a remote peer if the object is used with an RTCPeerConnection object.

The concept of duplicating MediaStream and MediaStreamTrack objects as described in [GETUSERMEDIA] is also applicable here. This feature can be used, for instance, in a video-conferencing scenario to display the local video from the user's camera and microphone in a local monitor, while only transmitting the audio to the remote peer (e.g. in response to the user using a "video mute" feature). Combining different MediaStreamTrack objects into new MediaStream objects is useful in certain situations.

Note

In this document, we only specify aspects of the following objects that are relevant when used along with an RTCPeerConnection. Please refer to the original definitions of the objects in the [GETUSERMEDIA] document for general information on using MediaStream and MediaStreamTrack.

The id attribute specified in MediaStream returns an id that is unique to this stream, so that streams can be recognized at the remote end of the RTCPeerConnection API.

When a MediaStream is created to represent a stream obtained from a remote peer, the id attribute is initialized from information provided by the remote source.

Note

The id of a MediaStream object is unique to the source of the stream, but that does not mean it is not possible to end up with duplicates. For example, the tracks of a locally generated stream could be sent from one user agent to a remote peer using RTCPeerConnection and then sent back to the original user agent in the same manner, in which case the original user agent will have multiple streams with the same id (the locally-generated one and the one received from the remote peer).

A MediaStreamTrack object's reference to its MediaStream in the non-local media source case (an RTP source, as is the case for each MediaStreamTrack associated with an RTCRtpReceiver) is always strong.

Whenever an RTCRtpReceiver receives data on an RTP source whose corresponding MediaStreamTrack is muted, but not ended, and the [[Receptive]] slot of the RTCRtpTransceiver object the RTCRtpReceiver is a member of is true, it MUST queue a task to set the muted state of the corresponding MediaStreamTrack to false.

When one of the SSRCs for RTP source media streams received by an RTCRtpReceiver is removed either due to reception of a BYE or via timeout, it MUST queue a task to set the muted state of the corresponding MediaStreamTrack to true. Note that setRemoteDescription can also lead to the setting of the muted state of the track to the value true.

The procedures add a track, remove a track and set a track's muted state are specified in [GETUSERMEDIA].

When a MediaStreamTrack track produced by an RTCRtpReceiver receiver has ended [GETUSERMEDIA] (such as via a call to receiver.track.stop), the user agent MAY choose to free resources allocated for the incoming stream, by for instance turning off the decoder of receiver.

The concept of constraints and constrainable properties, including MediaTrackConstraints (MediaStreamTrack.getConstraints(), MediaStreamTrack.applyConstraints()), and MediaTrackSettings (MediaStreamTrack.getSettings()) are outlined in [GETUSERMEDIA]. However, the constrainable properties of tracks sourced from a peer connection are different than those sourced by getUserMedia(); the constraints and settings applicable to MediaStreamTracks sourced from a remote source are defined here. The settings of a remote track represent the latest frame received.

MediaStreamTrack.getCapabilities() MUST always return the empty set and MediaStreamTrack.applyConstraints() MUST always reject with OverconstrainedError on remote tracks for constraints defined here.

The following constrainable properties are defined to apply to video MediaStreamTracks sourced from a remote source:

Property Name	Values	Notes
width	`ConstrainULong`	As a setting, this is the width, in pixels, of the latest frame received.
height	`ConstrainULong`	As a setting, this is the height, in pixels, of the latest frame received.
frameRate	`ConstrainDouble`	As a setting, this is an estimate of the frame rate based on recently received frames.
aspectRatio	`ConstrainDouble`	As a setting, this is the aspect ratio of the latest frame; this is the width in pixels divided by height in pixels as a double rounded to the tenth decimal place.

This document does not define any constrainable properties to apply to audio MediaStreamTracks sourced from a remote source.

This section is non-normative.

When two peers decide they are going to set up a connection to each other, they both go through these steps. The STUN/TURN server configuration describes a server they can use to get things like their public IP address or to set up NAT traversal. They also have to send data for the signaling channel to each other using the same out-of-band mechanism they used to establish that they were going to communicate in the first place.

Example 9

const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
const pc = new RTCPeerConnection(configuration);

// send any ice candidates to the other peer
pc.onicecandidate = ({candidate}) => signaling.send({candidate});

// let the "negotiationneeded" event trigger offer generation
pc.onnegotiationneeded = async () => {
  try {
    await pc.setLocalDescription();
    // send the offer to the other peer
    signaling.send({description: pc.localDescription});
  } catch (err) {
    console.error(err);
  }
};

pc.ontrack = ({track, streams}) => {
  // once media for a remote track arrives, show it in the remote video element
  track.onunmute = () => {
    // don't set srcObject again if it is already set.
    if (remoteView.srcObject) return;
    remoteView.srcObject = streams[0];
  };
};

// call start() to initiate
function start() {
  addCameraMic();
}

// add camera and microphone to connection
async function addCameraMic() {
  try {
    // get a local stream, show it in a self-view and add it to be sent
    const stream = await navigator.mediaDevices.getUserMedia(constraints);
    for (const track of stream.getTracks()) {
      pc.addTrack(track, stream);
    }
    selfView.srcObject = stream;
  } catch (err) {
    console.error(err);
  }
}

signaling.onmessage = async ({data: {description, candidate}}) => {
  try {
    if (description) {
      await pc.setRemoteDescription(description);
      // if we got an offer, we need to reply with an answer
      if (description.type == 'offer') {
        if (!selfView.srcObject) {
          // blocks negotiation on permission (not recommended in production code)
          await addCameraMic();
        }
        await pc.setLocalDescription();
        signaling.send({description: pc.localDescription});
      }
    } else if (candidate) {
      await pc.addIceCandidate(candidate);
    }
  } catch (err) {
    console.error(err);
  }
};

When two peers decide they are going to set up a connection to each other and want to have the ICE, DTLS, and media connections "warmed up" such that they are ready to send and receive media immediately, they both go through these steps.

Example 10

const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
let pc;
let audio;
let video;
let started = false;

// Call warmup() before media is ready, to warm-up ICE, DTLS, and media.
async function warmup(isAnswerer) {
  pc = new RTCPeerConnection(configuration);
  if (!isAnswerer) {
    audio = pc.addTransceiver('audio');
    video = pc.addTransceiver('video');
  }

  // send any ice candidates to the other peer
  pc.onicecandidate = ({candidate}) => signaling.send({candidate});

  // let the "negotiationneeded" event trigger offer generation
  pc.onnegotiationneeded = async () => {
    try {
      await pc.setLocalDescription();
      // send the offer to the other peer
      signaling.send({description: pc.localDescription});
    } catch (err) {
      console.error(err);
    }
  };

  pc.ontrack = async ({track, transceiver}) => {
    try {
      // once media for the remote track arrives, show it in the video element
      event.track.onunmute = () => {
        // don't set srcObject again if it is already set.
        if (!remoteView.srcObject) {
          remoteView.srcObject = new MediaStream();
        }
        remoteView.srcObject.addTrack(track);
      }

      if (isAnswerer) {
        if (track.kind == 'audio') {
          audio = transceiver;
        } else if (track.kind == 'video') {
          video = transceiver;
        }
        if (started) await addCameraMicWarmedUp();
      }
    } catch (err) {
      console.error(err);
    }
  };

  try {
    // get a local stream, show it in a self-view and add it to be sent
    selfView.srcObject = await navigator.mediaDevices.getUserMedia(constraints);
    if (started) await addCameraMicWarmedUp();
  } catch (err) {
    console.error(err);
  }
}

// call start() after warmup() to begin transmitting media from both ends
function start() {
  signaling.send({start: true});
  signaling.onmessage({data: {start: true}});
}

// add camera and microphone to already warmed-up connection
async function addCameraMicWarmedUp() {
  const stream = selfView.srcObject;
  if (audio && video && stream) {
    await Promise.all([
      audio.sender.replaceTrack(stream.getAudioTracks()[0]),
      video.sender.replaceTrack(stream.getVideoTracks()[0]),
    ]);
  }
}

signaling.onmessage = async ({data: {start, description, candidate}}) => {
  if (!pc) warmup(true);

  try {
    if (start) {
      started = true;
      await addCameraMicWarmedUp();
    } else if (description) {
      await pc.setRemoteDescription(description);
      // if we got an offer, we need to reply with an answer
      if (description.type == 'offer') {
        await pc.setLocalDescription();
        signaling.send({description: pc.localDescription});
      }
    } else {
      await pc.addIceCandidate(candidate);
    }
  } catch (err) {
    console.error(err);
  }
};

A client wants to send multiple RTP encodings (simulcast) to a server.

Example 11

const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {'iceServers': [{'urls': 'stun:stun.example.org'}]};
let pc;

// call start() to initiate
async function start() {
  pc = new RTCPeerConnection(configuration);

  // let the "negotiationneeded" event trigger offer generation
  pc.onnegotiationneeded = async () => {
    try {
      await pc.setLocalDescription();
      // send the offer to the other peer
      signaling.send({description: pc.localDescription});
    } catch (err) {
      console.error(err);
    }
  };

  try {
    // get a local stream, show it in a self-view and add it to be sent
    const stream = await navigator.mediaDevices.getUserMedia(constraints);
    selfView.srcObject = stream;
    pc.addTransceiver(stream.getAudioTracks()[0], {direction: 'sendonly'});
    pc.addTransceiver(stream.getVideoTracks()[0], {
      direction: 'sendonly',
      sendEncodings: [
        {rid: 'q', scaleResolutionDownBy: 4.0}
        {rid: 'h', scaleResolutionDownBy: 2.0},
        {rid: 'f'},
      ]
    });
  } catch (err) {
    console.error(err);
  }
}

signaling.onmessage = async ({data: {description, candidate}}) => {
  try {
    if (description) {
      await pc.setRemoteDescription(description);
      // if we got an offer, we need to reply with an answer
      if (description.type == 'offer') {
        await pc.setLocalDescription();
        signaling.send({description: pc.localDescription});
      }
    } else if (candidate) {
      await pc.addIceCandidate(candidate);
    }
  } catch (err) {
    console.error(err);
  }
};

This example shows how to create an RTCDataChannel object and perform the offer/answer exchange required to connect the channel to the other peer. The RTCDataChannel is used in the context of a simple chat application using an input field for user input.

Example 12

const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
let pc, channel;

// call start() to initiate
function start() {
  pc = new RTCPeerConnection(configuration);

  // send any ice candidates to the other peer
  pc.onicecandidate = ({candidate}) => signaling.send({candidate});

  // let the "negotiationneeded" event trigger offer generation
  pc.onnegotiationneeded = async () => {
    try {
      await pc.setLocalDescription();
      // send the offer to the other peer
      signaling.send({description: pc.localDescription});
    } catch (err) {
      console.error(err);
    }
  };

  // create data channel and setup chat using "negotiated" pattern
  channel = pc.createDataChannel('chat', {negotiated: true, id: 0});
  channel.onopen = () => input.disabled = false;
  channel.onmessage = ({data}) => showChatMessage(data);

  input.onkeydown = ({key}) => {
    if (key != 'Enter') return;
    channel.send(input.value);
  }
}

signaling.onmessage = async ({data: {description, candidate}}) => {
  if (!pc) start();

  try {
    if (description) {
      await pc.setRemoteDescription(description);
      // if we got an offer, we need to reply with an answer
      if (description.type == 'offer') {
        await pc.setLocalDescription();
        signaling.send({description: pc.localDescription});
      }
    } else if (candidate) {
      await pc.addIceCandidate(candidate);
    }
  } catch (err) {
    console.error(err);
  }
};

This shows an example of one possible call flow between two browsers. This does not show the procedure to get access to local media or every callback that gets fired but instead tries to reduce it down to only show the key events and messages.

A message sequence chart detailing a call flow between two browsers

Examples assume that sender is an RTCRtpSender.

Sending the DTMF signal "1234" with 500 ms duration per tone:

Example 13

if (sender.dtmf.canInsertDTMF) {
  const duration = 500;
  sender.dtmf.insertDTMF('1234', duration);
} else {
  console.log('DTMF function not available');
}

Send the DTMF signal "123" and abort after sending "2".

Example 14

async function sendDTMF() {
  if (sender.dtmf.canInsertDTMF) {
    sender.dtmf.insertDTMF('123');
    await new Promise(r => sender.dtmf.ontonechange = e => e.tone == '2' && r());
    // empty the buffer to not play any tone after "2"
    sender.dtmf.insertDTMF('');
  } else {
    console.log('DTMF function not available');
  }
}

Send the DTMF signal "1234", and light up the active key using lightKey(key) while the tone is playing (assuming that lightKey("") will darken all the keys):

Example 15

const wait = ms => new Promise(resolve => setTimeout(resolve, ms));

if (sender.dtmf.canInsertDTMF) {
  const duration = 500; // ms
  sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '1234', duration);
  sender.dtmf.ontonechange = async ({tone}) => {
    if (!tone) return;
    lightKey(tone); // light up the key when playout starts
    await wait(duration);
    lightKey(''); // turn off the light after tone duration
  };
} else {
  console.log('DTMF function not available');
}

It is always safe to append to the tone buffer. This example appends before any tone playout has started as well as during playout.

Example 16

if (sender.dtmf.canInsertDTMF) {
  sender.dtmf.insertDTMF('123');
  // append more tones to the tone buffer before playout has begun
  sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '456');

  sender.dtmf.ontonechange = ({tone}) => {
    // append more tones when playout has begun
    if (tone != '1') return;
    sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '789');
  };
} else {
  console.log('DTMF function not available');
}

Send a 1-second "1" tone followed by a 2-second "2" tone:

Example 17

if (sender.dtmf.canInsertDTMF) {
  sender.dtmf.ontonechange = ({tone}) => {
    if (tone == '1') {
      sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '2', 2000);
    }
  };
  sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '1', 1000);
} else {
  console.log('DTMF function not available');
}

Perfect negotiation is a recommended pattern to manage negotiation transparently, abstracting this asymmetric task away from the rest of an application. This pattern has advantages over one side always being the offerer, as it lets applications operate on both peer connection objects simultaneously without risk of glare (an offer coming in outside of "stable" state). The rest of the application may use any and all modification methods and attributes, without worrying about signaling state races.

It designates different roles to the two peers, with behavior to resolve signaling collisions between them:

The polite peer uses rollback to avoid collision with an incoming offer.
The impolite peer ignores an incoming offer when this would collide with its own.

Together, they manage signaling for the rest of the application in a manner that doesn't deadlock. The example assumes a polite boolean variable indicating the designated role:

Example 18

const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
const pc = new RTCPeerConnection(configuration);

// call start() anytime on either end to add camera and microphone to connection
async function start() {
  try {
    const stream = await navigator.mediaDevices.getUserMedia(constraints);
    for (const track of stream.getTracks()) {
      pc.addTrack(track, stream);
    }
    selfView.srcObject = stream;
  } catch (err) {
    console.error(err);
  }
}

pc.ontrack = ({track, streams}) => {
  // once media for a remote track arrives, show it in the remote video element
  track.onunmute = () => {
    // don't set srcObject again if it is already set.
    if (remoteView.srcObject) return;
    remoteView.srcObject = streams[0];
  };
};

// - The perfect negotiation logic, separated from the rest of the application ---

// keep track of some negotiation state to prevent races and errors
let makingOffer = false;
let ignoreOffer = false;
let isSettingRemoteAnswerPending = false;

// send any ice candidates to the other peer
pc.onicecandidate = ({candidate}) => signaling.send({candidate});

// let the "negotiationneeded" event trigger offer generation
pc.onnegotiationneeded = async () => {
  try {
    makingOffer = true;
    await pc.setLocalDescription();
    signaling.send({description: pc.localDescription});
  } catch (err) {
     console.error(err);
  } finally {
    makingOffer = false;
  }
};

signaling.onmessage = async ({data: {description, candidate}}) => {
  try {
    if (description) {
      // An offer may come in while we are busy processing SRD(answer).
      // In this case, we will be in "stable" by the time the offer is processed
      // so it is safe to chain it on our Operations Chain now.
      const readyForOffer =
          !makingOffer &&
          (pc.signalingState == "stable" || isSettingRemoteAnswerPending);
      const offerCollision = description.type == "offer" && !readyForOffer;

      ignoreOffer = !polite && offerCollision;
      if (ignoreOffer) {
        return;
      }
      isSettingRemoteAnswerPending = description.type == "answer";
      await pc.setRemoteDescription(description); // SRD rolls back as needed
      isSettingRemoteAnswerPending = false;
      if (description.type == "offer") {
        await pc.setLocalDescription();
        signaling.send({description: pc.localDescription});
      }
    } else if (candidate) {
      try {
        await pc.addIceCandidate(candidate);
      } catch (err) {
        if (!ignoreOffer) throw err; // Suppress ignored offer's candidates
      }
    }
  } catch (err) {
    console.error(err);
  }
}

Note that this is timing sensitive, and deliberately uses versions of setLocalDescription (without arguments) and setRemoteDescription (with implicit rollback) to avoid races with other signaling messages being serviced.

The ignoreOffer variable is needed, because the RTCPeerConnection object on the impolite side is never told about ignored offers. We must therefore suppress errors from incoming candidates belonging to such offers.

Some operations throw or fire RTCError. This is an extension of DOMException that carries additional WebRTC-specific information.

constructor()

Run the following steps:

Let init be the constructor's first argument.
Let message be the constructor's second argument.
Let e be a new RTCError object.
Invoke the DOMException constructor of e with the message argument set to message and the name argument set to "OperationError".

Note
This name does not have a mapping to a legacy code so e.code will return 0.
Set all RTCError attributes of e to the value of the corresponding attribute in init if it is present, otherwise set it to null.
Return e.

errorDetail of type RTCErrorDetailType, readonly: The WebRTC-specific error code for the type of error that occurred.
sdpLineNumber of type long, readonly, nullable: If errorDetail is "sdp-syntax-error" this is the line number where the error was detected (the first line has line number 1).
sctpCauseCode of type long, readonly, nullable: If errorDetail is "sctp-failure" this is the SCTP cause code of the failed SCTP negotiation.
receivedAlert of type unsigned long, readonly, nullable: If errorDetail is "dtls-failure" and a fatal DTLS alert was received, this is the value of the DTLS alert received.
sentAlert of type unsigned long, readonly, nullable: If errorDetail is "dtls-failure" and a fatal DTLS alert was sent, this is the value of the DTLS alert sent.; (Feature at Risk) Issue 1
All attributes defined in RTCError are marked at risk due to lack of implementation (errorDetail, sdpLineNumber, sctpCauseCode, receivedAlert and sentAlert). This does not include attributes inherited from DOMException.

dictionary RTCErrorInit {
  required RTCErrorDetailType errorDetail;
  long sdpLineNumber;
  long sctpCauseCode;
  unsigned long receivedAlert;
  unsigned long sentAlert;
};

The errorDetail, sdpLineNumber, sctpCauseCode, receivedAlert and sentAlert members of RTCErrorInit have the same definitions as the attributes of the same name of RTCError.

`RTCErrorDetailType` Enumeration description
Enum value	Description
`data-channel-failure`	The data channel has failed.
`dtls-failure`	The DTLS negotiation has failed or the connection has been terminated with a fatal error. The `message` contains information relating to the nature of error. If a fatal DTLS alert was received, the `receivedAlert` attribute is set to the value of the DTLS alert received. If a fatal DTLS alert was sent, the `sentAlert` attribute is set to the value of the DTLS alert sent.
`fingerprint-failure`	The `RTCDtlsTransport`'s remote certificate did not match any of the fingerprints provided in the SDP. If the remote peer cannot match the local certificate against the provided fingerprints, this error is not generated. Instead a "bad_certificate" (42) DTLS alert might be received from the remote peer, resulting in a "`dtls-failure`".
`sctp-failure`	The SCTP negotiation has failed or the connection has been terminated with a fatal error. The `sctpCauseCode` attribute is set to the SCTP cause code.
`sdp-syntax-error`	The SDP syntax is not valid. The `sdpLineNumber` attribute is set to the line number in the SDP where the syntax error was detected.
`hardware-encoder-not-available`	The hardware encoder resources required for the requested operation are not available.
`hardware-encoder-error`	The hardware encoder does not support the provided parameters.

The RTCErrorEvent interface is defined for cases when an RTCError is raised as an event:

constructor(): Constructs a new RTCErrorEvent.

error of type RTCError, readonly: The RTCError describing the error that triggered the event.

error of type RTCError: The RTCError describing the error associated with the event (if any).

This section is non-normative.

The following events fire on RTCDataChannel objects:

Event name	Interface	Fired when...
open	`Event`	The `RTCDataChannel` object's underlying data transport has been established (or re-established).
message	`MessageEvent` [html]	A message was successfully received.
bufferedamountlow	`Event`	The `RTCDataChannel` object's `bufferedAmount` decreases from above its `bufferedAmountLowThreshold` to less than or equal to its `bufferedAmountLowThreshold`.
error	`RTCErrorEvent`	An error occurred on the data channel.
closing	`Event`	The `RTCDataChannel` object transitions to the "`closing`" state
close	`Event`	The `RTCDataChannel` object's underlying data transport has been closed.

The following events fire on RTCPeerConnection objects:

Event name	Interface	Fired when...
track	`RTCTrackEvent`	New incoming media has been negotiated for a specific `RTCRtpReceiver`, and that receiver's `track` has been added to any associated remote `MediaStream`s.
negotiationneeded	`Event`	The browser wishes to inform the application that session negotiation needs to be done (i.e. a createOffer call followed by setLocalDescription).
signalingstatechange	`Event`	The connection's `[[SignalingState]]` has changed. This state change is the result of either `setLocalDescription` or `setRemoteDescription` being invoked.
iceconnectionstatechange	`Event`	The `RTCPeerConnection`'s `[[IceConnectionState]]` has changed.
icegatheringstatechange	`Event`	The `RTCPeerConnection`'s `[[IceGatheringState]]` has changed.
icecandidate	`RTCPeerConnectionIceEvent`	A new `RTCIceCandidate` is made available to the script.
connectionstatechange	`Event`	The `RTCPeerConnection`.`connectionState` has changed.
icecandidateerror	`RTCPeerConnectionIceErrorEvent`	A failure occured when gathering ICE candidates.
datachannel	`RTCDataChannelEvent`	A new `RTCDataChannel` is dispatched to the script in response to the other peer creating a channel.

The following events fire on RTCDTMFSender objects:

Event name	Interface	Fired when...
tonechange	`RTCDTMFToneChangeEvent`	The `RTCDTMFSender` object has either just begun playout of a tone (returned as the `tone` attribute) or just ended the playout of tones in the `toneBuffer` (returned as an empty value in the `tone` attribute).

The following events fire on RTCIceTransport objects:

Event name	Interface	Fired when...
statechange	`Event`	The `RTCIceTransport` state changes.
gatheringstatechange	`Event`	The `RTCIceTransport` gathering state changes.
selectedcandidatepairchange	`Event`	The `RTCIceTransport`'s selected candidate pair changes.

The following events fire on RTCDtlsTransport objects:

Event name	Interface	Fired when...
statechange	`Event`	The `RTCDtlsTransport` state changes.
error	`RTCErrorEvent`	An error occurred on the `RTCDtlsTransport` (either "`dtls-failure`" or "`fingerprint-failure`").

The following events fire on RTCSctpTransport objects:

Event name	Interface	Fired when...
statechange	`Event`	The `RTCSctpTransport` state changes.

This section is non-normative.

This section is non-normative; it specifies no new behaviour, but instead summarizes information already present in other parts of the specification. The overall security considerations of the general set of APIs and protocols used in WebRTC are described in [RFC8827].

This document extends the Web platform with the ability to set up real-time, direct communication between browsers and other devices, including other browsers.

This means that data and media can be shared between applications running in different browsers, or between an application running in the same browser and something that is not a browser, something that is an extension to the usual barriers in the Web model against sending data between entities with different origins.

The WebRTC specification provides no user prompts or chrome indicators for communication; it assumes that once the Web page has been allowed to access media, it is free to share that media with other entities as it chooses. Peer-to-peer exchanges of data view WebRTC datachannels can thus occur without any user explicit consent or involvement, similarly as a server-mediated exchange (e.g. via Web Sockets) could occur without user involvement.

Even without WebRTC, the Web server providing a Web application will know the public IP address to which the application is delivered. Setting up communications exposes additional information about the browser’s network context to the web application, and may include the set of (possibly private) IP addresses available to the browser for WebRTC use. Some of this information has to be passed to the corresponding party to enable the establishment of a communication session.

Revealing IP addresses can leak location and means of connection; this can be sensitive. Depending on the network environment, it can also increase the fingerprinting surface and create persistent cross-origin state that cannot easily be cleared by the user.

A connection will always reveal the IP addresses proposed for communication to the corresponding party. The application can limit this exposure by choosing not to use certain addresses using the settings exposed by the RTCIceTransportPolicy dictionary, and by using relays (for instance TURN servers) rather than direct connections between participants. One will normally assume that the IP address of TURN servers is not sensitive information. These choices can for instance be made by the application based on whether the user has indicated consent to start a media connection with the other party.

Mitigating the exposure of IP addresses to the application itself requires limiting the IP addresses that can be used, which will impact the ability to communicate on the most direct path between endpoints. Browsers are encouraged to provide appropriate controls for deciding which IP addresses are made available to applications, based on the security posture desired by the user. The choice of which addresses to expose is controlled by local policy (see [RFC8828] for details).

Since the browser is an active platform executing in a trusted network environment (inside the firewall), it is important to limit the damage that the browser can do to other elements on the local network, and it is important to protect data from interception, manipulation and modification by untrusted participants.

Mitigations include:

A user agent will always request permission from the correspondent user agent to communicate using ICE. This ensures that the user agent can only send to partners who you have shared credentials with.
A user agent will always request ongoing permission to continue sending using ICE continued consent. This enables a receiver to withdraw consent to receive.
A user agent will always encrypt data, with strong per-session keying (DTLS-SRTP).
A user agent will always use congestion control. This ensures that WebRTC cannot be used to flood the network.

These measures are specified in the relevant IETF documents.

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.

Communication certificates may be opaquely shared using postMessage(message, options) in anticipation of future needs. User agents are strongly encouraged to isolate the private keying material these objects hold a handle to, from the processes that have access to the RTCCertificate objects, to reduce memory attack surface.

As described above, the list of IP addresses exposed by the WebRTC API can be used as a persistent cross-origin state.

Beyond IP addresses, the WebRTC API exposes information about the underlying media system via the RTCRtpSender.getCapabilities and RTCRtpReceiver.getCapabilities methods, including detailed and ordered information about the codecs that the system is able to produce and consume. A subset of that information is likely to be represented in the SDP session descriptions generated, exposed and transmitted during session negotiation. That information is in most cases persistent across time and origins, and increases the fingerprint surface of a given device.

When establishing DTLS connections, the WebRTC API can generate certificates that can be persisted by the application (e.g. in IndexedDB). These certificates are not shared across origins, and get cleared when persistent storage is cleared for the origin.

setRemoteDescription guards against malformed and invalid SDP by throwing exceptions, but makes no attempt to guard against SDP that might be unexpected by the application. Setting the remote description can cause significant resources to be allocated (including image buffers and network ports), media to start flowing (which may have privacy and bandwidth implications) among other things. An application that does not guard against malicious SDP could be at risk of resource deprivation, unintentionally allowing incoming media or at risk of not having certain events fire like ontrack if the other endpoint does not negotiate sending. Applications need to be on guard against malevolent SDP.

This section is non-normative.

The WebRTC 1.0 specification exposes an API to control protocols (defined within the IETF) necessary to establish real-time audio, video and data exchange.

The Telecommunications Device for the Deaf (TDD/TTY) enables individuals who are hearing or speech impaired (among others) to communicate over telephone lines. Real-Time Text, defined in [RFC4103], utilizes T.140 encapsulated in RTP to enable the transition from TDD/TTY devices to IP-based communications, including emergency communication with Public Safety Access Points (PSAP).

Since Real-Time Text requires the ability to send and receive data in near real time, it can be best supported via the WebRTC 1.0 data channel API. As defined by the IETF, the data channel protocol utilizes the SCTP/DTLS/UDP protocol stack, which supports both reliable and unreliable data channels. The IETF chose to standardize SCTP/DTLS/UDP over proposals for an RTP data channel which relied on SRTP key management and were focused on unreliable communications.

Since the IETF chose a different approach than the RTP data channel as part of the WebRTC suite of protocols, as of the time of this publication there is no standardized way for the WebRTC APIs to directly support Real-Time Text as defined at IETF and implemented in U.S. (FCC) regulations. The WebRTC working Group will evaluate whether the developing IETF protocols in this space warrant direct exposure in the browser APIs and is looking for input from the relevant user communities on this potential gap.

Within the IETF MMUSIC Working Group, work is ongoing to enable Real-time text to be sent over the WebRTC data channel, allowing gateways to be deployed to translate between the SCTP data channel protocol and RFC 4103 Real-Time Text. This work, once completed, is expected to enable a unified and interoperable approach for integrating real-time text in WebRTC user-agents (including browsers) - through a gateway or otherwise.

At the time of this publication, gateways that enable effective RTT support in WebRTC clients can be developed e.g. through a custom WebRTC data channel. This is deemed sufficient until such time as future standardized gateways are enabled via IETF protocols such as the SCTP data channel protocol and RFC 4103 Real-Time Text. This will need to be defined at IETF in conjunction with related work at W3C groups to effectively and consistently standardise RTT support internationally.

Since its publication as a W3C Recommendation in January 2021, the following candidate amendments have been integrated in this document.

Candidate Correction 5:
- Forbid ICE gathering and connectivity checks on administrative prohibited candidates - section Set the session description (PR #2708) - Changes to Web Platform Tests: #21025 #45339
- Forbid ICE gathering and connectivity checks on administrative prohibited candidates - section Set the session description (PR #2708) - Changes to Web Platform Tests: #21025 #45339
Candidate Correction 12:
- Mark RTP Pause/Resume as not supported - section 5.4.1 Simulcast functionality (PR #2755) - Changes to Web Platform Tests: #34912
- Remove interaction between encoding.active and simulcast ~rid - section 4.4.1.5 Set the session description (PR #2754) - Changes to Web Platform Tests: #34912
Candidate Correction 13:
- Rollback restores ridless encoding trounced by sRD(simulcastOffer). - section RTCRtpSender Interface (PR #2797) - Changes to Web Platform Tests: #37477
- Rollback restores ridless encoding trounced by sRD(simulcastOffer). - section Set the session description (PR #2797) - Changes to Web Platform Tests: #37477
- Rollback restores ridless encoding trounced by sRD(simulcastOffer). - section Set the session description (PR #2797) - Changes to Web Platform Tests: #37477
Candidate Correction 14:
- Make RTCTransceiver.direction reflects local preference in offers and answers - section 4.4.1.5 Set the session description (PR #2759)
Candidate Addition 16:
- Add RTCIceCandidate.relayProtocol - section 4.8.1 RTCIceCandidate Interface (PR #2763) - Changes to Web Platform Tests: #36157
- Add RTCIceCandidate.relayProtocol - section RTCIceServerTransportProtocol Enum (PR #2763) - Changes to Web Platform Tests: #36157
- Add RTCIceCandidate.relayProtocol - section Attributes (PR #2763) - Changes to Web Platform Tests: #36157
Candidate Correction 18:
- TypeError unless all or none of encodings have rids and on duplicate rids - section Methods (PR #2774, PR #2775) - Changes to Web Platform Tests: #37477
Candidate Correction 22:
- Allow remote offer rid pruning of encodings through the client answer. - section 4.4.1.5 Set the session description (PR #2758)
Candidate Addition 23:
- Add RTCIceCandidate.url - section 4.8.1 RTCIceCandidate Interface (PR #2773) - Changes to Web Platform Tests: #36572
- Mark RTCPeerConnectionIceEvent.url as deprecated - section Attributes (PR #2773) (not testable)
- Add RTCIceCandidate.url - section Attributes (PR #2773) - Changes to Web Platform Tests: #36572
Candidate Correction 24:
- Queue two tasks upon finishing ICE gathering, and fire gatheringstatechange & icegatheringstatechange in same task - section 5.6 RTCIceTransport Interface (PR #2894) - Changes to Web Platform Tests: #44687
Candidate Correction 25:
- Remove duplicate rids in proposedSendEncodings. - section 4.4.1.5 Set the session description (PR #2800) - Changes to Web Platform Tests: #37477
Candidate Correction 26:
- Prune createAnswer()'s encodings and SendEncodings in sLD(answer). - section 4.4.1.5 Set the session description (PR #2801)
- Prune createAnswer()'s encodings and SendEncodings in sLD(answer). - section Methods (PR #2801)
Candidate Correction 27:
- Ignore comma-separated rid alternatives. - section 4.4.1.5 Set the session description (PR #2813) - Changes to Web Platform Tests: #36155 #37477
- Ignore comma-separated rid alternatives. - section Methods (PR #2813) - Changes to Web Platform Tests: #36155 #37477
Candidate Correction 33:
- Use the url spec to parse ice server urls - section 4.4.1.6 Set the configuration (PR #2853, PR #2996, PR #2998) - Changes to Web Platform Tests: #47959
- Determine if DTMF can be sent inside queued playout task - section Methods (PR #2861)
Candidate Correction 37:
- Don't fail sRD(offer) over rid mismatch, just answer with unicast. - section 4.4.1.5 Set the session description (PR #2794)
- Don't fail sRD(offer) over rid mismatch, just answer with unicast. - section 4.4.1.5 Set the session description (PR #2794)
Candidate Correction 38:
- Prevent GC of non-closed RTCDataChannels - section 6.2.4 Closing procedure (PR #2902) - Changes to Web Platform Tests: #43369
Candidate Addition 45:
- Convert RTCIceCandidatePair dictionary to an interface - section 5.6.2 RTCIceCandidatePair Dictionary (PR #2961) - Changes to Web Platform Tests: #46647 #46655
Candidate Correction 46:
- Replace RFC8829 reference with RFC9429 - section B.1 Normative references (PR #2966) (no change needed in tests)
Candidate Addition 49:
- Add codec to RTCRtpEncodingParameters - section Methods (PR #2985) - Changes to Web Platform Tests: #47663
- Add codec to RTCRtpEncodingParameters - section Methods (PR #2985) - Changes to Web Platform Tests: #47663
- Add codec to RTCRtpEncodingParameters - section Methods (PR #2985) - Changes to Web Platform Tests: #47663
- Add codec to RTCRtpEncodingParameters - section Methods (PR #2985) - Changes to Web Platform Tests: #47663
- Add codec to RTCRtpEncodingParameters - section Methods (PR #2985) - Changes to Web Platform Tests: #47663
- Add codec to RTCRtpEncodingParameters - section Methods (PR #2985) - Changes to Web Platform Tests: #47663
- Add codec to RTCRtpEncodingParameters - section Methods (PR #2985) - Changes to Web Platform Tests: #47663
- Add codec to RTCRtpEncodingParameters - section Dictionary RTCRtpEncodingParameters Members (PR #2985) - Changes to Web Platform Tests: #47663
- Add codec to RTCRtpEncodingParameters - section Set the session description (PR #2985) - Changes to Web Platform Tests: #47663
Candidate Correction 50:
- Use Performance.timeOrigin + Performance.now() for stats timestamps - section Dictionary RTCStats Members (PR #3005) - Changes to Web Platform Tests: #48361
Candidate Addition 51:
- setCodecPreferences supports both send and receive codecs (filtered by direction) - section Methods (PR #3018) - Changes to Web Platform Tests: #49421
Candidate Correction 52:
- Two codecs are considered the same even if level-id is not - section Methods (PR #3023) - Changes to Web Platform Tests: #49990

The editors wish to thank the Working Group chairs and Team Contact, Harald Alvestrand, Stefan Håkansson, Erik Lagerway and Dominique Hazaël-Massieux, for their support. Substantial text in this specification was provided by many people including Martin Thomson, Harald Alvestrand, Justin Uberti, Eric Rescorla, Peter Thatcher, Jan-Ivar Bruaroey and Peter Saint-Andre. Dan Burnett would like to acknowledge the significant support received from Voxeo and Aspect during the development of this specification.

The RTCRtpSender and RTCRtpReceiver objects were initially described in the W3C ORTC CG, and have been adapted for use in this specification.

Candidate Correction 46:Replace RFC8829 reference with RFC9429 (PR #2966)

~~B.1 Normative references~~

[DOM]: DOM Standard. Anne van Kesteren. WHATWG. Living Standard. URL: https://dom.spec.whatwg.org/
[ECMASCRIPT-6.0]: ECMA-262 6th Edition, The ECMAScript 2015 Language Specification. Allen Wirfs-Brock. Ecma International. June 2015. Standard. URL: http://www.ecma-international.org/ecma-262/6.0/index.html
[Fetch]: Fetch Standard. Anne van Kesteren. WHATWG. Living Standard. URL: https://fetch.spec.whatwg.org/
[FILEAPI]: File API. Marijn Kruisselbrink~~; Arun Ranganathan~~. W3C. ~~11 September 2019~~4 December 2024. W3C Working Draft. URL: https://www.w3.org/TR/FileAPI/
[FIPS-180-4]: FIPS PUB ~~180-4~~ 180-4: Secure Hash ~~Standard~~Standard (SHS). U.S. Department of Commerce/National Institute of Standards and Technology. August 2015. National Standard. URL: https://nvlpubs.nist.gov/nistpubs/FIPS/NIST.FIPS.180-4.pdf
[GETUSERMEDIA]: Media Capture and Streams. Cullen Jennings; Bernard Aboba; Jan-Ivar Bruaroey; Henrik Boström; youenn fablet~~; Daniel Burnett; Adam Bergkvist; Anant Narayanan~~. W3C. ~~21 January 2021~~19 December 2024. ~~W3C Candidate Recommendation~~CRD. URL: https://www.w3.org/TR/mediacapture-streams/
[hr-time]: High Resolution ~~Time Level 2~~Time. ~~Ilya Grigorik~~Yoav Weiss. W3C. 21 7 November ~~2019~~2024. W3C ~~Recommendation~~Working Draft. URL: https://www.w3.~~org/TR/hr-time-2/~~org/TR/hr-time-3/
[HTML]: HTML Standard. Anne van Kesteren; Domenic Denicola; Dominic Farolino; Ian Hickson; Philip Jägenstedt; Simon Pieters. WHATWG. Living Standard. URL: https://html.spec.whatwg.org/multipage/
[IANA-HASH-FUNCTION]: Hash Function Textual Names. IANA. URL: https://www.iana.org/assignments/hash-function-text-names/hash-function-text-names.xml
[IANA-RTP-2]: RTP Payload Format media types. IANA. URL: https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
[INFRA]: Infra Standard. Anne van Kesteren; Domenic Denicola. WHATWG. Living Standard. URL: https://infra.spec.whatwg.org/
[RFC2119]: Key words for use in RFCs to Indicate Requirement Levels. S. Bradner. IETF. March 1997. Best Current Practice. URL: ~~https://tools.ietf.org/html/rfc2119~~https://www.rfc-editor.org/rfc/rfc2119
[RFC3550]: RTP: A Transport Protocol for Real-Time Applications. H. Schulzrinne; S. Casner; R. Frederick; V. Jacobson. IETF. July 2003. Internet Standard. URL: ~~https://tools.ietf.org/html/rfc3550~~https://www.rfc-editor.org/rfc/rfc3550
[RFC3890]: A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP). M. Westerlund. IETF. September 2004. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc3890~~https://www.rfc-editor.org/rfc/rfc3890
~~[RFC3986]~~: ~~Uniform Resource Identifier (URI): Generic Syntax. T. Berners-Lee; R. Fielding; L. Masinter. IETF. January 2005. Internet Standard. URL: https://tools.ietf.org/html/rfc3986~~
[RFC4566]: SDP: Session Description Protocol. M. Handley; V. Jacobson; C. Perkins. IETF. July 2006. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc4566~~https://www.rfc-editor.org/rfc/rfc4566
[RFC4572]: Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP). J. Lennox. IETF. July 2006. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc4572~~https://www.rfc-editor.org/rfc/rfc4572
[RFC5245]: Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols. J. Rosenberg. IETF. April 2010. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc5245~~https://www.rfc-editor.org/rfc/rfc5245
[RFC5246]: The Transport Layer Security (TLS) Protocol Version 1.2. T. Dierks; E. Rescorla. IETF. August 2008. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc5246~~https://www.rfc-editor.org/rfc/rfc5246
[RFC5285]: A General Mechanism for RTP Header Extensions. D. Singer; H. Desineni. IETF. July 2008. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc5285~~https://www.rfc-editor.org/rfc/rfc5285
[RFC5389]: Session Traversal Utilities for NAT (STUN). J. Rosenberg; R. Mahy; P. Matthews; D. Wing. IETF. October 2008. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc5389~~https://www.rfc-editor.org/rfc/rfc5389
[RFC5506]: Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences. I. Johansson; M. Westerlund. IETF. April 2009. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc5506~~https://www.rfc-editor.org/rfc/rfc5506
[RFC5888]: The Session Description Protocol (SDP) Grouping Framework. G. Camarillo; H. Schulzrinne. IETF. June 2010. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc5888~~https://www.rfc-editor.org/rfc/rfc5888
[RFC6464]: A Real-time Transport Protocol (RTP) Header Extension for Client-to-Mixer Audio Level Indication. J. Lennox, Ed.; E. Ivov; E. Marocco. IETF. December 2011. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc6464~~https://www.rfc-editor.org/rfc/rfc6464
[RFC6465]: A Real-time Transport Protocol (RTP) Header Extension for Mixer-to-Client Audio Level Indication. E. Ivov, Ed.; E. Marocco, Ed.; J. Lennox. IETF. December 2011. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc6465~~https://www.rfc-editor.org/rfc/rfc6465
[RFC6544]: TCP Candidates with Interactive Connectivity Establishment (ICE). J. Rosenberg; A. Keranen; B. B. Lowekamp; A. B. Roach. IETF. March 2012. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc6544~~https://www.rfc-editor.org/rfc/rfc6544
[RFC7064]: URI Scheme for the Session Traversal Utilities for NAT (STUN) Protocol. S. Nandakumar; G. Salgueiro; P. Jones; M. Petit-Huguenin. IETF. November 2013. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc7064~~https://www.rfc-editor.org/rfc/rfc7064
[RFC7065]: Traversal Using Relays around NAT (TURN) Uniform Resource Identifiers. M. Petit-Huguenin; S. Nandakumar; G. Salgueiro; P. Jones. IETF. November 2013. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc7065~~https://www.rfc-editor.org/rfc/rfc7065
[RFC7656]: A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources. J. Lennox; K. Gross; S. Nandakumar; G. Salgueiro; B. Burman, Ed.. IETF. November 2015. Informational. URL: ~~https://tools.ietf.org/html/rfc7656~~https://www.rfc-editor.org/rfc/rfc7656
[RFC7675]: Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness. M. Perumal; D. Wing; R. Ravindranath; T. Reddy; M. Thomson. IETF. October 2015. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc7675~~https://www.rfc-editor.org/rfc/rfc7675
[RFC7874]: WebRTC Audio Codec and Processing Requirements. JM. Valin; C. Bran. IETF. May 2016. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc7874~~https://www.rfc-editor.org/rfc/rfc7874
[RFC8174]: Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words. B. Leiba. IETF. May 2017. Best Current Practice. URL: ~~https://tools.ietf.org/html/rfc8174~~https://www.rfc-editor.org/rfc/rfc8174
[RFC8261]: Datagram Transport Layer Security (DTLS) Encapsulation of SCTP Packets. M. Tuexen; R. Stewart; R. Jesup; S. Loreto. IETF. November 2017. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8261~~https://www.rfc-editor.org/rfc/rfc8261
[RFC8445]: Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal. A. Keranen; C. Holmberg; J. Rosenberg. IETF. July 2018. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8445~~https://www.rfc-editor.org/rfc/rfc8445
[RFC8656]: Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN). T. Reddy, Ed.; A. Johnston, Ed.; P. Matthews; J. Rosenberg. IETF. February 2020. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc8656
[RFC8826]: Security Considerations for WebRTC~~for WebRTC~~. E. Rescorla. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8826~~https://www.rfc-editor.org/rfc/rfc8826
~~[RFC8829]~~: ~~JavaScript Session Establishment Protocol (JSEP). J. Uberti; C. Jennings; E. Rescorla, Ed.. IETF. January 2021. Proposed Standard. URL: https://tools.ietf.org/html/rfc8829~~
[RFC8831]: WebRTC Data Channels. R. Jesup; S. Loreto; M. Tüxen. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8831~~https://www.rfc-editor.org/rfc/rfc8831
[RFC8832]: WebRTC Data Channel Establishment Protocol. R. Jesup; S. Loreto; M. Tüxen. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8832~~https://www.rfc-editor.org/rfc/rfc8832
[RFC8834]: Media Transport and Use of RTP in WebRTC. C. Perkins; M. Westerlund; J. Ott. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8834~~https://www.rfc-editor.org/rfc/rfc8834
[RFC8835]: Transports for WebRTC. H. Alvestrand. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8835~~https://www.rfc-editor.org/rfc/rfc8835
[RFC8838]: Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol. E. Ivov; J. Uberti; P. Saint-Andre. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8838~~https://www.rfc-editor.org/rfc/rfc8838
[RFC8841]: Session Description Protocol (SDP) Offer/Answer Procedures for Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer Security (DTLS) Transport. C. Holmberg; R. Shpount; S. Loreto; G. Camarillo. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8841~~https://www.rfc-editor.org/rfc/rfc8841
[RFC8843]: Negotiating Media Multiplexing Using the Session Description Protocol (SDP). C. Holmberg; H. Alvestrand; C. Jennings. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8843~~https://www.rfc-editor.org/rfc/rfc8843
[RFC8851]: RTP Payload Format Restrictions. A.B. Roach, Ed.. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8851~~https://www.rfc-editor.org/rfc/rfc8851
[RFC8853]: Using Simulcast in Session Description Protocol (SDP) and RTP Sessions. B. Burman; M. Westerlund; S. Nandakumar; M. Zanaty. IETF. January 2021. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc8853~~https://www.rfc-editor.org/rfc/rfc8853
[RFC8863]: Interactive Connectivity Establishment Patiently Awaiting Connectivity (ICE PAC). C. Holmberg; J. Uberti. IETF. January 2021. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc8863
[RFC9429]: JavaScript Session Establishment Protocol (JSEP). J. Uberti; C. Jennings; E. Rescorla, Ed. IETF. April 2024. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc9429
[SDP]: An Offer/Answer Model with Session Description Protocol (SDP). J. Rosenberg; H. Schulzrinne. IETF. June 2002. Proposed Standard. URL: ~~https://tools.ietf.org/html/rfc3264~~https://www.rfc-editor.org/rfc/rfc3264
[STUN-PARAMETERS]: STUN Error Codes. IETF. IANA. April 2011. IANA Parameter Assignment. URL: https://www.iana.org/assignments/stun-parameters/stun-parameters.xhtml#stun-parameters-6
[WebCryptoAPI]: Web Cryptography API. Mark Watson. W3C. 26 January 2017. W3C Recommendation. URL: https://www.w3.org/TR/WebCryptoAPI/
[WEBIDL]: ~~Web IDL. Boris Zbarsky. W3C. 15 December 2016. W3C Editor's Draft. URL: https://heycam.github.io/webidl/~~; Web IDL Standard. Edgar Chen; Timothy Gu. WHATWG. Living Standard. URL: https://webidl.spec.whatwg.org/
[WEBRTC-STATS]: Identifiers for WebRTC's Statistics API. Harald Alvestrand; Varun Singh; Henrik Boström. W3C. ~~20 January 2021~~6 March 2025. ~~W3C Candidate Recommendation~~CRD. URL: https://www.w3.org/TR/webrtc-stats/
[X509V3]: ITU-T Recommendation X.509 version 3 (1997). "Information Technology - Open Systems Interconnection - The Directory Authentication Framework" ISO/IEC 9594-8:1997.. ITU.
[X690]: Recommendation X.690 — Information Technology — ASN.1 Encoding Rules — Specification of Basic Encoding Rules (BER), Canonical Encoding Rules (CER), and Distinguished Encoding Rules (DER). ITU. URL: https://www.itu.int/ITU-T/studygroups/com17/languages/X.690-0207.pdf

[API-DESIGN-PRINCIPLES]: Web Platform Design Principles. Martin Thomson; Jeffrey Yasskin. W3C. 6 March 2025. W3C Working Group Note. URL: https://www.w3.org/TR/design-principles/
[INDEXEDDB]: Indexed Database API. Nikunj Mehta; Jonas Sicking; Eliot Graff; Andrei Popescu; Jeremy Orlow; Joshua Bell. W3C. 8 January 2015. W3C Recommendation. URL: https://www.w3.org/TR/IndexedDB/
[RFC4103]: RTP Payload for Text Conversation. G. Hellstrom; P. Jones. IETF. June 2005. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc4103
[RFC6236]: Negotiation of Generic Image Attributes in the Session Description Protocol (SDP). I. Johansson; K. Jung. IETF. May 2011. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc6236
[RFC7728]: RTP Stream Pause and Resume. B. Burman; A. Akram; R. Even; M. Westerlund. IETF. February 2016. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc7728
[RFC8825]: Overview: Real-Time Protocols for Browser-Based Applications. H. Alvestrand. IETF. January 2021. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc8825
[RFC8827]: WebRTC Security Architecture. E. Rescorla. IETF. January 2021. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc8827
[RFC8828]: WebRTC IP Address Handling Requirements. J. Uberti; G. Shieh. IETF. January 2021. Proposed Standard. URL: https://www.rfc-editor.org/rfc/rfc8828

Abstract

Status of This Document

5.6.2 `RTCIceCandidatePair` Dictionary

Dictionary `RTCIceCandidatePair` Members

6.2.4 Closing procedure

B.1 Normative references

Abstract

Status of This Document

5.6.2 RTCIceCandidatePair Dictionary

Dictionary RTCIceCandidatePair Members

6.2.4 Closing procedure

B.1 Normative references

5.6.2 `RTCIceCandidatePair` Dictionary

Dictionary `RTCIceCandidatePair` Members