Abstract

This document defines a set of ECMAScript APIs in WebIDL to allow media 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 developed by the Media Capture Task Force.

Status of This Document

This section describes the status of this document at the time of its publication. Other documents may supersede this document. 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/.

The API is based on preliminary work done in the WHATWG.

While the specification is feature complete and is expected to be stable, there are also a number of known substantive issues on the specification that will be addressed during the Candidate Recommendation period based on implementation experience feedback.

It might also evolve based on feedback gathered as its associated test suite evolves. This test suite will be used to build an implementation report of the API.

To go into Proposed Recommendation status, the group expects to demonstrate implementation of each feature in at least two deployed browsers, and at least one implementation of each optional feature. Mandatory feature with only one implementation may be marked as optional in a revised Candidate Recommendation where applicable.

The following features are marked as risk:

This document was published by the Web Real-Time Communications Working Group as a Candidate Recommendation. This document is intended to become a W3C Recommendation. Comments regarding this document are welcome. Please send them to public-webrtc@w3.org (subscribe, archives). W3C publishes a Candidate Recommendation to indicate that the document is believed to be stable and to encourage implementation by the developer community. This Candidate Recommendation is expected to advance to Proposed Recommendation no earlier than 15 April 2018.

Publication as a Candidate Recommendation does not imply endorsement by the W3C Membership. This is a draft document and may be updated, replaced or obsoleted by other documents at any time. It is inappropriate to cite this document as other than work in progress.

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 1 March 2017 W3C Process Document.

1. Introduction

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:

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 Media Capture Task Force. An overview of the system can be found in [ RTCWEB-OVERVIEW] and [RTCWEB-SECURITY].

2. Conformance

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

The key words MAY, MUST, MUST NOT, SHALL, and SHOULD are to be interpreted as described in [RFC2119].

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

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

Implementations that use ECMAScript to implement the APIs defined in this specification MUST implement them in a manner consistent with the ECMAScript Bindings defined in the Web IDL specification [WEBIDL-1], as this specification uses that specification and terminology.

3. Terminology

The EventHandler interface, representing a callback used for event handlers, and the ErrorEvent interface are defined in [HTML51].

The concepts queue a task, fire a simple event and networking task source are defined in [HTML51].

The terms event, event handlers and event handler event types are defined in [HTML51].

The terms MediaStream, MediaStreamTrack, and MediaStreamConstraints are defined in [GETUSERMEDIA].

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 [TRICKLE-ICE] Section 2.

The term RTCStatsType is defined in [WEBRTC-STATS].

When referring to exceptions, the terms throw and create are defined in [WEBIDL-1].

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

4. Peer-to-peer connections

4.1 Introduction

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 XMLHttpRequest [XMLHttpRequest] or Web Sockets [ WEBSOCKETS-API].

4.2 Configuration

4.2.1 RTCConfiguration Dictionary

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";
    DOMString                peerIdentity;
    sequence<RTCCertificate> certificates;
    [EnforceRange]
    octet                    iceCandidatePoolSize = 0;
};
Dictionary RTCConfiguration Members
iceServers of type sequence<RTCIceServer>

An array of objects describing servers available to be used by ICE, such as STUN and TURN servers.

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.

peerIdentity of type DOMString

Sets the target peer identity for the RTCPeerConnection. The RTCPeerConnection will not establish a connection to a remote peer unless it can be successfully authenticated with the provided name.

certificates of type sequence<RTCCertificate>

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.

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 [JSEP] (section 3.5.4. and section 4.1.1.).

4.2.2 RTCIceCredentialType Enum

enum RTCIceCredentialType {
    "password",
    "oauth"
};
Enumeration description
password The credential is a long-term authentication username and password, as described in [RFC5389], Section 10.2.
oauth

An OAuth 2.0 based authentication method, as described in [ RFC7635]. It uses the OAuth 2.0 Implicit Grant type, with PoP (Proof-of-Possession) Token type, as described in [ RFC6749] and [OAUTH-POP-KEY-DISTRIBUTION].

The OAuth Client and the Auhorization Server roles are defined in [RFC6749] Section 1.1.

If [RFC7635] is used in the WebRTC context then the OAuth Client is responsible for refreshing the credential information, and updating the ICE Agent with fresh new credentials before the accessToken expires. The OAuth Client can use the RTCPeerConnection setConfiguration method to periodically refresh the TURN credentials.

For OAuth Authentication, the ICE Agent requires three pieces of credential information. The credential is composed of a kid, which the RTCIceServer username member is used for, and macKey and accessToken, which are placed in the RTCOAuthCredential dictionary. All of this information can be extracted from the OAuth response parameters, which are received from the Authorization Server. The relevant OAuth response parameters are the "kid", the "key", and the "access_token". These can be used to extract all the necessary credential infromation (the kid, macKey, and accessToken) that are required by the ICE Agent for the Authentication.

The [OAUTH-POP-KEY-DISTRIBUTION] defines alg parameter in Section 4.1 and 6. and describes that if the Authorization Server doesn't have prior knowledge of the capabilities of the client, then the OAuth Client needs to provide information about the ICE Agent HMAC alg capabilities. This information helps the Authorization Server to generate the approriate HMAC key. The HMAC alg defines the input key length, and HMAC algorithm Familly (e.g. SHA), and HMAC algorithm type (e.g. symmetric/asymmetric).

The OAuth Client sends an OAuth Request to the Authorization Server with OAuth param alg and further OAuth related parameters, to get an OAuth Response with the access_token, key, kid, and further OAuth related parameters.

However, this specification uses a simplified alg approach. The length of the HMAC key (RTCOAuthCredential.macKey) MAY be any integer number of bytes greater than 20 (160 bits). This negates the need to query the HMAC Algorithm capabilities of the ICE Agent, and still allows for hash agility as described by [STUN-BIS], Section 15.3.

Note
According to [RFC7635] Section 4.1, the HMAC key MUST be a symmetric key.

Currently the STUN/TURN protocols use only SHA-1 and SHA-2 family hash algorithms for Message Integrity Protection, as defined in [RFC5389] Section 15.4, and [STUN-BIS] Section 14.6.

When [RFC7635] is used in WebRTC context, this specification adds the following additional consideration to it.

The OAuth Client SHOULD obtain the mac_key by requesting an alg value of HS256. This will result in a 256-bit HMAC key.

HS256 is defined in [RFC7518] Section 3.1. It is recommended here because:

  • The OAuth respose key parameter is received in JWK format according to [OAUTH-POP-KEY-DISTRIBUTION] Section 4.2. JWK's algorithms are normatively registered in the IANA "JSON Web Signature and Encryption Algorithms" registry.
  • STUN/TURN currently use SHA family HMAC algorithms only.
  • The key MUST be symmetric, according to [RFC7635].
  • A 256-bit key is large enough to support all currently defined STUN message integrity attributes.

More details about OAuth PoP Client can be found in [ OAUTH-POP-KEY-DISTRIBUTION] Section 4.

More details about Access-Token can be found in [ RFC7635], Section 6.2.

4.2.3 RTCOAuthCredential Dictionary

The RTCOAuthCredential dictionary is used to describe the OAuth auth credential information which is used by the STUN/TURN client (inside the ICE Agent) to authenticate against a STUN/TURN server, as described in [RFC7635]. Note that the kid parameter is not located in this dictionary, but in RTCIceServer's username member.

dictionary RTCOAuthCredential {
    required DOMString macKey;
    required DOMString accessToken;
};
Dictionary RTCOAuthCredential Members
macKey of type DOMString, required

The "mac_key", as described in [RFC7635], Section 6.2, in a base64-url encoded format. It is used in STUN message integrity hash calculation (as the password is used in password based authentication). Note that the OAuth response "key" parameter is a JSON Web Key (JWK) or a JWK encrypted with a JWE format. Also note that this is the only OAuth parameter whose value is not used directly, but must be extracted from the "k" parameter value from the JWK, which contains the needed base64-encoded "mac_key".

accessToken of type DOMString, required

The "access_token", as described in [RFC7635], Section 6.2, in a base64-encoded format. This is an encrypted self-contained token that is opaque to the application. Authenticated encryption is used for message encryption and integrity protection. The access token contains a non-encrypted nonce value, which is used by the Authorization Server for unique mac_key generation. The second part of the token is protected by Authenticated Encryption. It contains the mac_key, a timestamp and a lifetime. The timestamp combined with lifetime provides expiry information; this information describes the time window during which the token credential is valid and accepted by the TURN server.

An example of an RTCOAuthCredential dictionary is:

Example 1
{
    macKey: "WmtzanB3ZW9peFhtdm42NzUzNG0=",
    accessToken: "AAwg3kPHWPfvk9bDFL936wYvkoctMADzQ5VhNDgeMR3+ZlZ35byg972fW8QjpEl7bx91YLBPFsIhsxloWcXPhA=="
}

4.2.4 RTCIceServer Dictionary

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 or RTCOAuthCredential)  credential;
             RTCIceCredentialType               credentialType = "password";
};
Dictionary RTCIceServer Members
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, and credentialType is "password", then this attribute specifies the username to use with that TURN server.

If this RTCIceServer object represents a TURN server, and credentialType is "oauth", then this attribute specifies the Key ID (kid) of the shared symmetric key, which is shared between the TURN server and the Authorization Server, as described in [RFC7635]. It is an ephemeral and unique key identifier. The kid allows the TURN server to select the appropriate keying material for decryption of the Access-Token, so the key identified by this kid is used in the Authenticated Encryption of the "access_token". The kid value is equal with the OAuth response "kid" parameter, as defined in [RFC7515] Section 4.1.4.

credential of type (DOMString or RTCOAuthCredential)

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

If credentialType is "password", credential is a DOMString, and represents a long-term authentication password, as described in [ RFC5389], Section 10.2.

If credentialType is "oauth", credential is an RTCOAuthCredential, which contains the OAuth access token and MAC key.

credentialType of type RTCIceCredentialType, defaulting to "password"

If this RTCIceServer object represents a TURN server, then this attribute specifies how credential should be used when that TURN server requests authorization.

An example array of RTCIceServer objects is:

Example 2
[
     { urls: "stun:stun1.example.net" },
     { urls: ["turns:turn.example.org", "turn:turn.example.net"],
       username: "user",
       credential: "myPassword",
       credentialType: "password" },
     { urls: "turns:turn2.example.net",
       username: "22BIjxU93h/IgwEb",
       credential: {
                       macKey: "WmtzanB3ZW9peFhtdm42NzUzNG0=",
                       accessToken: "AAwg3kPHWPfvk9bDFL936wYvkoctMADzQ5VhNDgeMR3+ZlZ35byg972fW8QjpEl7bx91YLBPFsIhsxloWcXPhA=="
                     },
       credentialType: "oauth" }
     }
]

4.2.5 RTCIceTransportPolicy Enum

As described in [JSEP] (section 4.1.1.), if the iceTransportPolicy member of the RTCConfiguration is specified, it defines the ICE candidate policy [JSEP] (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"
};
Enumeration description (non-normative)
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.ip .

4.2.6 RTCBundlePolicy Enum

As described in [JSEP] (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"
};
Enumeration description (non-normative)
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.

4.2.7 RTCRtcpMuxPolicy Enum

As described in [JSEP] (section 4.1.1.), the RtcpMuxPolicy affects what ICE candidates are gathered to support non-multiplexed RTCP.

enum RTCRtcpMuxPolicy {
    // At risk due to lack of implementers' interest.
    "negotiate",
    "require"
};
Enumeration description (non-normative)
negotiate Gather ICE candidates for both RTP and RTCP candidates. If the remote-endpoint is capable of multiplexing RTCP, multiplex RTCP on the RTP candidates. If it is not, use both the RTP and RTCP candidates separately. Note that, as stated in [JSEP] (section 4.1.1.), the user agent MAY not implement non-multiplexed RTCP, in which case it will reject attempts to construct an RTCPeerConnection with the negotiate policy.
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.
Feature at Risk 1

Aspects of this specification supporting non-multiplexed RTP/RTCP are marked as features at risk, since there is no clear commitment from implementers. This includes:

  1. The value negotiate, since there is no clear commitment from implementers for the behavior associated with this.
  2. Support for the rtcpTransport attribute within the RTCRtpSender and RTCRtpReceiver .

4.2.8 Offer/Answer Options

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

dictionary RTCOfferAnswerOptions {
    boolean voiceActivityDetection = true;
};
Dictionary RTCOfferAnswerOptions Members
voiceActivityDetection of type boolean, defaulting to true

Many codecs and systems are capable of detecting "silence" and changing their behavior in this case by doing things such as not transmitting any media. In many cases, such as when dealing with emergency calling or sounds other than spoken voice, it is desirable to be able to turn off this behavior. This option allows the application to provide information about whether it wishes this type of processing enabled or disabled.

dictionary RTCOfferOptions : RTCOfferAnswerOptions {
    boolean iceRestart = false;
};
Dictionary RTCOfferOptions Members
iceRestart of type boolean, defaulting to false

When the value of this dictionary member is true, the generated description will have ICE credentials that are different from the current credentials (as visible in the localDescription attribute's SDP). Applying the generated description will restart ICE, as described in section 9.1.1.1 of [ICE].

When the value of this dictionary member is false, and the localDescription attribute has valid ICE credentials, the generated description will have the same ICE credentials as the current value from the localDescription attribute.

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

dictionary RTCAnswerOptions : RTCOfferAnswerOptions {
};

4.3 State Definitions

4.3.1 RTCSignalingState Enum

enum RTCSignalingState {
    "stable",
    "have-local-offer",
    "have-remote-offer",
    "have-local-pranswer",
    "have-remote-pranswer",
    "closed"
};
Enumeration description
stable There is no offer­answer 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.
signalling state transition diagram
Figure 1 Non-normative signalling state transitions diagram

An example set of transitions might be:

Caller transition:
  • new RTCPeerConnection(): stable
  • setLocal(offer): have-local-offer
  • setRemote(pranswer): have-remote-pranswer
  • setRemote(answer): stable
Callee transition:
  • new RTCPeerConnection(): stable
  • setRemote(offer): have-remote-offer
  • setLocal(pranswer): have-local-pranswer
  • setLocal(answer): stable

4.3.2 RTCIceGatheringState Enum

enum RTCIceGatheringState {
    "new",
    "gathering",
    "complete"
};
Enumeration 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 "completed" gathering state.

4.3.3 RTCPeerConnectionState Enum

enum RTCPeerConnectionState {
    "new",
    "connecting",
    "connected",
    "disconnected",
    "failed",
    "closed"
};
Enumeration description
new Any of the RTCIceTransport s or RTCDtlsTransport s are in the "new" state and none of the transports are in the "connecting", "checking", "failed" or "disconnected" state, or all transports are in the "closed" state, or there are no transports.
connecting Any of the RTCIceTransport s or RTCDtlsTransport s are in the "connecting" or "checking" state and none of them is in the "failed" state.
connected All RTCIceTransport s and RTCDtlsTransport s are in the "connected", "completed" or "closed" state and at least one of them is in the "connected" or "completed" state.
disconnected Any of the RTCIceTransport s or RTCDtlsTransport s are in the "disconnected" state and none of them are in the "failed" or "connecting" or "checking" state.
failed Any of the RTCIceTransport s or RTCDtlsTransport s are in a "failed" state.
closed The RTCPeerConnection object's [[IsClosed]] slot is true.

4.3.4 RTCIceConnectionState Enum

enum RTCIceConnectionState {
    "new",
    "checking",
    "connected",
    "completed",
    "failed",
    "disconnected",
    "closed"
};
Enumeration description
new Any of the RTCIceTransport s are in the "new" state and none of them are in the "checking", "failed" or "disconnected" state, or all RTCIceTransport s are in the "closed" state, or there are no transports.
checking Any of the RTCIceTransport s are in the "checking" state and none of them are in the "failed" or "disconnected" state.
connected All RTCIceTransport s are in the "connected", "completed" or "closed" state and at least one of them is in the "connected" state.
completed All RTCIceTransport s are in the "completed" or "closed" state and at least one of them is in the "completed" state.
failed Any of the RTCIceTransport s are in the "failed" state.
disconnected Any of the RTCIceTransport s are in the "disconnected" state and none of them are in the "failed" state.
closed The RTCPeerConnection object's [[IsClosed]] slot is true.

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.

4.4 RTCPeerConnection Interface

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

4.4.1 Operation

Calling new RTCPeerConnection(configuration) creates an RTCPeerConnection object.

configuration.servers 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 signaling state, a connection state, an ICE gathering state, and an ICE connection state. These are initialized when the object is created.

The ICE protocol implementation of an RTCPeerConnection is represented by an ICE agent [ICE]. 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 [JSEP]. 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.

4.4.1.1 Constructor

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

  1. Let connection be a newly created RTCPeerConnection object.

  2. If the certificates value in configuration is non-empty, check that the expires on each value is in the future. If a certificate has expired, throw an InvalidAccessError; otherwise, store the certificates. If no certificates value was specified, one or more new RTCCertificate instances are generated for use with this RTCPeerConnection instance. This MAY happen asynchronously and the value of certificates remains undefined for the subsequent steps.

  3. If configuration.rtcpMuxPolicy is "negotiate", and the user agent does not implement non-muxed RTCP, throw a NotSupportedError.

  4. Initialize connection's ICE Agent.

  5. Let connection have a [[Configuration]] internal slot. Set the configuration specified by configuration.

  6. Let connection have an [[IsClosed]] internal slot, initialized to false.

  7. Let connection have a [[NegotiationNeeded]] internal slot, initialized to false.

  8. Let connection have an [[SctpTransport]] internal slot, initialized to null.

  9. Let connection have an [[Operations]] internal slot, representing an operations queue, initialized to an empty list.

  10. Let connection have an [[LastOffer]] internal slot, initialized to "".

  11. Let connection have an [[LastAnswer]] internal slot, initialized to "".

  12. Set connection's signaling state to "stable".

  13. Set connection's ICE connection state to "new".

  14. Set connection's ICE gathering state to "new".

  15. Set connection's connection state to "new".

  16. Set connection's pendingLocalDescription , currentLocalDescription , pendingRemoteDescription and currentRemoteDescription to null.

  17. Return connection.

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

4.4.1.2 Enqueue an operation

To enqueue an operation to an RTCPeerConnection object's operation queue, run the following steps:

  1. Let connection be the RTCPeerConnection object.

  2. If connection's [[IsClosed]] slot is true, return a promise rejected with a newly created InvalidStateError.

  3. Let operation be the operation to be enqueued.

  4. Let p be a new promise.

  5. Append operation to [[Operations]].

  6. If the length of [[Operations]] is exactly 1, execute operation.

  7. Upon fulfillment or rejection of the promise returned by the operation, run the following steps:

    1. If connection's [[IsClosed]] slot 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's [[IsClosed]] slot 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]].

  8. Return p.

4.4.1.3 Update the connection state

An RTCPeerConnection object has an aggregated connection state. Whenever the state of an RTCDtlsTransport or RTCIceTransport changes or when the [[IsClosed]] slot turns true, the user agent MUST update the connection state by queueing a task that runs the following steps:

  1. Let connection be this RTCPeerConnection object.

  2. Let newState be the value of deriving a new state value as described by the RTCPeerConnectionState enum.

  3. If connection's connection state is equal to newState, abort these steps.

  4. Let connection's connection state be newState.

  5. Fire a simple event named connectionstatechange at connection.

4.4.1.4 Update the ICE gathering state

To update the ICE gathering state of an RTCPeerConnection instance connection, the user agent MUST queue a task that runs the following steps:

  1. If connection's [[IsClosed]] slot is true, abort these steps.

  2. Let newState be the value of deriving a new state value as described by the RTCIceGatheringState enum.

  3. If connection's ICE gathering state is equal to newState, abort these steps.

  4. Set connection's ice gathering state to newState.

  5. Fire a simple event named icegatheringstatechange at connection.

  6. If newState is "completed", fire an ice candidate event named icecandidate with 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 .
4.4.1.5 Update the ICE connection state

To update the ICE connection state of an RTCPeerConnection instance connection, the user agent MUST queue a task that runs the following steps:

  1. If connection's [[IsClosed]] slot is true, abort these steps.

  2. Let newState be the value of deriving a new state value as described by the RTCIceConnectionState enum.

  3. If connection's ICE connection state is equal to newState, abort these steps.

  4. Set connection's ice connection state to newState.

  5. Fire a simple event named iceconnectionstatechange at connection.

4.4.1.6 Set the RTCSessionDescription

To set an RTCSessionDescription description on an RTCPeerConnection object connection, enqueue the following steps to connection's operation queue:

  1. Let p be a new promise.

  2. In parallel, start the process to apply description as described in [JSEP] (section 5.5. and section 5.6.).

    1. If the process to apply description fails for any reason, then user agent MUST queue a task that runs the following steps:

      1. If connection's [[IsClosed]] slot is true, then abort these steps.

      2. If the description's type is invalid for the current signaling state of connection as described in [JSEP] (section 5.5. and section 5.6.), then reject p with a newly created InvalidStateError and abort these steps.

      3. If description is set as a local description, if description.type is offer and description.sdp is not equal to connection's [[LastOffer]] slot, then reject p with a newly created InvalidModificationError and abort these steps.
      4. If description is set as a local description, if description.type is "rollback" and signaling state is "stable" then reject p with a newly created InvalidStateError and abort these steps.

      5. If description is set as a local description, if description.type is "answer" or "pranswer" and description.sdp is not equal to connection's [[LastAnswer]] slot, then reject p with a newly created InvalidModificationError and abort these steps.
      6. 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.

      7. If the content of description is invalid, then reject p with a newly created InvalidAccessError and abort these steps.

      8. For all other errors, reject p with a newly created OperationError.

    2. If description is applied successfully, the user agent MUST queue a task that runs the following steps:

      1. If connection's [[IsClosed]] slot is true, then abort these steps.

      2. If description is set as a local description, then run one of the following steps:

      3. Otherwise, if description is set as a remote description, then run one of the following steps:

      4. If connection's signaling state changed above, fire a simple event named signalingstatechange at connection.

      5. If description is of type "answer", and it initiates the closure of an existing SCTP association, as defined in [SCTP-SDP], Sections 10.3 and 10.4, set the value of connection's [[SctpTransport]] internal slot to null.

      6. 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 [ SCTP-SDP], Sections 10.3 and 10.4, set the value of connection's [[SctpTransport]] internal slot to a newly created RTCSctpTransport .

        2. If description negotiates the DTLS role of the SCTP transport, and there is an RTCDataChannel with a null id , then generate an ID according to [ RTCWEB-DATA-PROTOCOL]. If no available ID could be generated, then run the following steps:

          1. Let channel be the RTCDataChannel object for which an ID could not be generated.

          2. Set channel's [[ReadyState]] slot to "closed".

          3. Fire an event named error with an OperationError exception at channel.

          4. Fire a simple event named close at channel.

      7. If description is set as a local description, then run the following steps for each media description in description that is not yet associated with an RTCRtpTransceiver object:

        1. Let transceiver be the RTCRtpTransceiver used to create the media description.

        2. Set transceiver's mid value to the mid of the corresponding media description.

        3. If transceiver's [[Stopped]] slot is true, abort these sub steps.

        4. If description is of type "answer" or "pranswer", then set transceiver's [[CurrentDirection]] slot to an RTCRtpTransceiverDirection value representing the direction of the corresponding media description.

        5. If the media description is indicated as using an existing media transport according to [ BUNDLE], let transport and rtcpTransport be the RTCDtlsTransport objects representing the RTP and RTCP components of that transport, respectively.

        6. Otherwise, let transport and rtcpTransport be newly created RTCDtlsTransport objects, each with a new underlying RTCIceTransport . Though if RTCP multiplexing is negotiated according to [RFC5761], or if connection's RTCRtcpMuxPolicy is require , do not create any RTCP-specific transport objects, and instead let rtcpTransport equal transport.

        7. Set transceiver.[[Sender]].[[SenderTransport]] to transport.

        8. Set transceiver.[[Sender]].[[SenderRtcpTransport]] to rtcpTransport.

        9. Set transceiver.[[Receiver]].[[ReceiverTransport]] to transport.

        10. Set transceiver.[[Receiver]].[[ReceiverRtcpTransport]] to rtcpTransport.

      8. If description is set as a remote description, then run the following steps:

        1. Let trackEvents be an empty list.

        2. Run the following steps for each media description in description:

          1. Let direction be an RTCRtpTransceiverDirection value representing the direction from the media description, but with the send and receive directions reversed to represent this peer's point of view.

          2. As described by [JSEP] (section 5.9.), attempt to find an existing RTCRtpTransceiver object, transceiver, to represent the media description.

          3. If no suitable transceiver is found (transceiver is unset), run the following steps:

            1. Create an RTCRtpSender, sender, from the media description.

            2. Create an RTCRtpReceiver, receiver, from the media description.

            3. Create an RTCRtpTransceiver with sender, receiver and direction set to "recvonly", and let transceiver be the result.

          4. Set transceiver's mid value to the mid of the corresponding media description. If the media description has no MID, and transceiver's mid is unset, generate a random value as described in [JSEP] (section 5.9.).

          5. If direction is "sendrecv" or "recvonly", and transceiver's [[CurrentDirection]] slot is neither "sendrecv" nor "recvonly", process the addition of a remote track for the media description, given transceiver and trackEvents.

          6. If direction is "sendonly" or "inactive", and transceiver's [[CurrentDirection]] slot is neither "sendonly" nor "inactive", process the removal of a remote track for the media description, given transceiver.

          7. If description is of type "answer" or "pranswer", then run the following steps:

            1. Set transceiver's [[CurrentDirection]] slot to direction.

            2. Let transport and rtcpTransport be the RTCDtlsTransport objects representing the RTP and RTCP components of the media transport used by transceiver's associated media description, according to [BUNDLE].

            3. Set transceiver.[[Sender]].[[SenderTransport]] to transport.

            4. Set transceiver.[[Sender]].[[SenderRtcpTransport]] to rtcpTransport.

            5. Set transceiver.[[Receiver]].[[ReceiverTransport]] to transport.

            6. Set transceiver.[[Receiver]].[[ReceiverRtcpTransport]] to rtcpTransport.

          8. If the media description is rejected, and transceiver is not already stopped, stop the RTCRtpTransceiver transceiver.

        3. For each RTCTrackEvent trackEvent in trackEvents, fire a track event named track with trackEvent at the connection object.

      9. If description is of type "rollback", then run the following steps:

        1. If the mid value of an RTCRtpTransceiver was set to a non-null value by the RTCSessionDescription that is being rolled back, set the mid value of that transceiver to null, as described by [JSEP] (section 4.1.8.2.).

        2. If an RTCRtpTransceiver was created by applying the RTCSessionDescription that is being rolled back, and a track has not been attached to it via addTrack, remove that transceiver from connection's set of transceivers, as described by [JSEP] (section 4.1.8.2.).

        3. Restore the value of connection's [[SctpTransport]] internal slot to its value at the last stable signaling state.

      10. If connection's signaling state is now "stable", update the negotiation-needed flag. If connection's [[NegotiationNeeded]] slot was true both before and after this update, queue a task that runs the following steps:

        1. If connection's [[IsClosed]] slot is true, abort these steps.

        2. If connection's [[NegotiationNeeded]] slot is false, abort these steps.

        3. Fire a simple event named negotiationneeded at connection.

      11. Resolve p with undefined.

  3. Return p.

4.4.1.7 Set the configuration

To set a configuration, run the following steps:

  1. Let configuration be the RTCConfiguration dictionary to be processed.
  2. Let connection be the target RTCPeerConnection object.
  3. If configuration.peerIdentity is set and its value differs from the target peer identity, throw an InvalidModificationError.
  4. If configuration.certificates is set and the set of certificates differs from the ones used when connection was constructed, throw an InvalidModificationError.
  5. If the value of configuration.bundlePolicy differs from the connection's bundle policy, throw an InvalidModificationError.
  6. If the value of configuration.rtcpMuxPolicy differs from the connection's rtcpMux policy, throw an InvalidModificationError.
  7. If the value of configuration.iceCandidatePoolSize differs from the connection's previously set iceCandidatePoolSize, and setLocalDescription has already been called, throw an InvalidModificationError.
  8. Set the ICE Agent's ICE transports setting to the value of configuration.iceTransportPolicy . As defined in [JSEP] (section 4.1.16.), 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.

  9. Set the ICE Agent's prefetched ICE candidate pool size as defined in [JSEP] (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 [JSEP] (section 4.1.16.).

  10. Let validatedServers be an empty list.

  11. If configuration.iceServers is defined, then run the following steps for each element:

    1. Let server be the current list element.

    2. If server.urls is a string, let server.urls be a list consisting of just that string.

    3. For each url in server.urls run the following steps:

      1. 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 [ RFC7064] fails, throw a SyntaxError. If scheme name is stun or stuns, and parsing the url using the syntax defined in [ RFC7065] fails, throw a SyntaxError.

      2. If scheme name is turn or turns, and either of server.username or server.credential are omitted, then throw an InvalidAccessError.

      3. If scheme name is turn or turns, and server.credentialType is "password", and server.credential is not a DOMString, then throw an InvalidAccessError and abort these steps.

      4. If scheme name is turn or turns, and server.credentialType is "oauth", and server.credential is not an RTCOAuthCredential, then throw an InvalidAccessError and abort these steps.

    4. Append server to validatedServers.

    Let validatedServers be the ICE Agent's ICE servers list.

    As defined in [JSEP] (section 4.1.16.), 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.

  12. Store the configuration in the [[Configuration]] internal slot.

4.4.2 Interface Definition

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.

[Constructor(optional RTCConfiguration configuration),
 Exposed=Window]
interface RTCPeerConnection : EventTarget {
    Promise<RTCSessionDescriptionInit> createOffer(optional RTCOfferOptions options);
    Promise<RTCSessionDescriptionInit> createAnswer(optional RTCAnswerOptions options);
    Promise<void>                      setLocalDescription(RTCSessionDescriptionInit description);
    readonly attribute RTCSessionDescription? localDescription;
    readonly attribute RTCSessionDescription? currentLocalDescription;
    readonly attribute RTCSessionDescription? pendingLocalDescription;
    Promise<void>                      setRemoteDescription(RTCSessionDescriptionInit description);
    readonly attribute RTCSessionDescription? remoteDescription;
    readonly attribute RTCSessionDescription? currentRemoteDescription;
    readonly attribute RTCSessionDescription? pendingRemoteDescription;
    Promise<void>                      addIceCandidate((RTCIceCandidateInit or RTCIceCandidate) candidate);
    readonly attribute RTCSignalingState      signalingState;
    readonly attribute RTCIceGatheringState   iceGatheringState;
    readonly attribute RTCIceConnectionState  iceConnectionState;
    readonly attribute RTCPeerConnectionState connectionState;
    readonly attribute boolean?               canTrickleIceCandidates;
    static sequence<RTCIceServer>      getDefaultIceServers();
    RTCConfiguration                   getConfiguration();
    void                               setConfiguration(RTCConfiguration configuration);
    void                               close();
             attribute EventHandler           onnegotiationneeded;
             attribute EventHandler           onicecandidate;
             attribute EventHandler           onicecandidateerror;
             attribute EventHandler           onsignalingstatechange;
             attribute EventHandler           oniceconnectionstatechange;
             attribute EventHandler           onicegatheringstatechange;
             attribute EventHandler           onconnectionstatechange;
};
Constructors
RTCPeerConnection
See the RTCPeerConnection constructor algorithm.
Attributes
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 description.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 represents the local RTCSessionDescription 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.

The currentLocalDescription attribute MUST return the last value that algorithms in this specification set it to, complete with any local candidates that have been generated by the ICE Agent since the offer or answer was created. Prior to being set, it returns null.

pendingLocalDescription of type RTCSessionDescription, readonly, nullable

The pendingLocalDescription attribute represents a local RTCSessionDescription 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. This attribute is updated by setLocalDescription .

The pendingLocalDescription attribute MUST return the last value that algorithms in this specification set it to, complete with any local candidates that have been generated by the ICE Agent since the offer or answer was created. Prior to being set, it returns 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 description.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 represents the last remote RTCSessionDescription 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.

The currentRemoteDescription attribute MUST return the value that algorithms in this specification set it to, complete with any remote candidates that have been supplied via addIceCandidate() since the offer or answer was created. Prior to being set, it returns null.

pendingRemoteDescription of type RTCSessionDescription, readonly, nullable

The pendingRemoteDescription attribute represents a remote RTCSessionDescription 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. This attribute is updated by setLocalDescription .

The pendingRemoteDescription attribute MUST return the value that algorithms in this specification set it to, completed with any remote candidates that have been supplied via addIceCandidate() since the offer or answer was created. Prior to being set, it returns null.

signalingState of type RTCSignalingState, readonly

The signalingState attribute MUST return the RTCPeerConnection object's signaling state.

iceGatheringState of type RTCIceGatheringState, readonly

The iceGatheringState attribute MUST return the ICE gathering state of the RTCPeerConnection instance.

iceConnectionState of type RTCIceConnectionState, readonly

The iceConnectionState attribute MUST return the ICE connection state of the RTCPeerConnection instance.

connectionState of type RTCPeerConnectionState, readonly

The connectionState attribute MUST return the connection state of the RTCPeerConnection instance.

canTrickleIceCandidates of type boolean, readonly, nullable

The canTrickleIceCandidates attribute indicates whether the remote peer is able to accept trickled ICE candidates [TRICKLE-ICE]. The value is determined based on whether a remote description indicates support for trickle ICE, as defined in [JSEP] (section 4.1.15.). 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 .
Methods
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 [JSEP]. As an offer, the generated SDP will contain the full set of codec/RTP/RTCP capabilities supported 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 [ICE], 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 and as input to requests for identity assertions.

If the RTCPeerConnection is configured to generate Identity assertions by calling setIdentityProvider , then the session description SHALL contain an appropriate assertion.

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.(This is a fingerprinting vector.)

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

  1. Let connection be the RTCPeerConnection object on which the method was invoked.

  2. If connection's [[IsClosed]] slot is true, return a promise rejected with a newly created InvalidStateError.

  3. If connection is configured with an identity provider, then begin the identity assertion request process if it has not already begun.

  4. Return the result of enqueuing the following steps to connection's operation queue:

    1. Let p be a new promise.

    2. In parallel, begin the steps to create an offer, given p.

    3. Return p.

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

  1. If connection was not constructed with a set of certificates, and one has not yet been generated, wait for it to be generated.

  2. Let provider be connection's currently configured identity provider if one has been configured, or null otherwise.

  3. If provider is non-null, wait for the identity assertion request process to complete.

  4. If provider was unable to produce an identity assertion, reject p with a newly created NotReadableError and abort these steps.

  5. Inspect the system state to determine the currently available resources as necessary for generating the offer, as described in [JSEP] (section 4.1.6.).

  6. If this inspection failed for any reason, reject p with a newly created OperationError and abort these steps.

  7. Queue a task that runs the final steps to create an offer, given p.

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

  1. If connection's [[IsClosed]] slot is true, then abort these steps.

  2. If connection was modified in such a way that additional inspection of the system state is necessary, or if its configured indentity provider is no longer provider, then in parallel begin the steps to create an offer again, given 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 steps to create an offer in parallel, a video RTCRtpTransceiver was added, requiring additional inspection of video system resources.
  3. Given the information that was obtained from previous inspection, the current state of connection and its RTCRtpTransceiver s, and the identity assertion from provider (if non-null), generate an SDP offer, sdpString, as described in [JSEP] (section 5.2.).

  4. Let offer be a newly created RTCSessionDescriptionInit dictionary with its type member initialized to the string "offer" and its sdp member initialized to sdpString.

  5. Set the [[LastOffer]] internal slot to sdpString.

  6. 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 [JSEP].

The generated SDP will also contain the ICE agent's usernameFragment, password and ICE options (as defined in [ICE], 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 and as input to requests for identity assertions.

An answer can be marked as provisional, as described in [JSEP] (section 4.1.8.1.), by setting the type to "pranswer".

If the RTCPeerConnection is configured to generate Identity assertions by calling setIdentityProvider , then the session description SHALL contain an appropriate assertion.

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

  1. Let connection be the RTCPeerConnection object on which the method was invoked.

  2. If connection's [[IsClosed]] slot is true, return a promise rejected with a newly created InvalidStateError.

  3. If connection is configured with an identity provider, then begin the identity assertion request process if it has not already begun.

  4. Return the result of enqueuing the following steps to connection's operation queue:

    1. If connection's signaling state is neither "have-remote-offer" nor "have-local-pranswer", return a promise rejected with a newly created InvalidStateError.

    2. Let p be a new promise.

    3. In parallel, begin the steps to create an answer, given p.

    4. Return p.

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

  1. If connection was not constructed with a set of certificates, and one has not yet been generated, wait for it to be generated.

  2. Let provider be connection's currently configured identity provider if one has been configured, or null otherwise.

  3. If provider is non-null, wait for the identity assertion request process to complete.

  4. If provider was unable to produce an identity assertion, reject p with a newly created NotReadableError and abort these steps.

  5. Inspect the system state to determine the currently available resources as necessary for generating the answer, as described in [JSEP] (section 4.1.7.).

  6. If this inspection failed for any reason, reject p with a newly created OperationError and abort these steps.

  7. 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:

  1. If connection's [[IsClosed]] slot is true, then abort these steps.

  2. If connection was modified in such a way that additional inspection of the system state is necessary, or if its configured indentity provider is no longer provider, then in parallel begin the steps to create an answer again, given 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 steps to create an answer in parallel, the direction was changed to "sendrecv", requiring additional inspection of video encoding resources.
  3. Given the information that was obtained from previous inspection and the current state of connection and its RTCRtpTransceiver s, and the identity assertion from provider (if non-null), generate an SDP answer, sdpString, as described in [JSEP] (section 5.3.).

  4. Let answer be a newly created RTCSessionDescriptionInit dictionary with its type member initialized to the string "answer" and its sdp member initialized to sdpString.

  5. Set the [[LastAnswer]] internal slot to sdpString.

  6. Resolve p with answer.

setLocalDescription

The setLocalDescription method instructs the RTCPeerConnection to apply the supplied RTCSessionDescriptionInit 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.

As noted in [JSEP] (section 5.4.) the SDP returned from createOffer or createAnswer MUST NOT be changed before passing it to setLocalDescription. As a result, when the method is invoked, the user agent MUST run the following steps:

  1. Let description be the first argument to setLocalDescription.
  2. Let lastOffer be the result returned by the last call to createOffer.
  3. Let lastAnswer be the result returned by the last call to createAnswer.
  4. If description.sdp is the empty string and description.type is "answer" or "pranswer", set description.sdp to lastAnswer.
  5. If description.sdp is the empty string and description.type is "offer", set description.sdp to lastOffer.
  6. Return the result of setting the RTCSessionDescription indicated by description.
Note

As noted in [JSEP] (section 5.8.), 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 return the result of setting the RTCSessionDescription indicated by the method's first argument.

In addition, a remote description is processed to determine and verify the identity of the peer.

If an a=identity attribute is present in the session description, the browser validates the identity assertion..

If the "peerIdentity" configuration is applied to the RTCPeerConnection , this establishes a target peer identity of the provided value. Alternatively, if the RTCPeerConnection has previously authenticated the identity of the peer (that is, there is a current value for peerIdentity ), then this also establishes a target peer identity.

The target peer identity cannot be changed once set. Once set, if a different value is provided, the user agent MUST reject the returned promise with a newly created InvalidModificationError and abort this operation. The RTCPeerConnection MUST be closed if the validated peer identity does not match the target peer identity.

If there is no target peer identity, then setRemoteDescription does not await the completion of identity validation.

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:

  1. Let candidate be the method's argument.

  2. Let connection be the RTCPeerConnection object on which the method was invoked.

  3. If both sdpMid and sdpMLineIndex are null, return a promise rejected with a newly created TypeError.

  4. Return the result of enqueuing the following steps to connection's operation queue:

    1. If remoteDescription is null return a promise rejected with a newly created InvalidStateError.

    2. Let p be a new promise.

    3. 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 , reject p with a newly created OperationError and abort these steps.

    4. 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 , reject p with a newly created OperationError and abort these steps.

    5. If candidate.usernameFragment is neither undefined nor null, and is not equal to any username fragment present in the corresponding media description of an applied remote description, reject p with a newly created OperationError and abort these steps.

    6. In parallel, add the ICE candidate candidate as described in [JSEP] (section 4.1.17.). 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.

      1. If candidate could not be successfully added the user agent MUST queue a task that runs the following steps:

        1. If connection's [[IsClosed]] slot is true, then abort these steps.

        2. Reject p with a DOMException object whose name attribute has the value OperationError and abort these steps.

      2. If candidate is applied successfully, the user agent MUST queue a task that runs the following steps:

        1. If connection's [[IsClosed]] slot is true, then abort these steps.

        2. If connection.pendingRemoteDescription is non-null, and represents the ICE generation for which candidate was processed, add candidate to connection.pendingRemoteDescription .

        3. If connection.currentRemoteDescription is non-null, and represents the ICE generation for which candidate was processed, add candidate to connection.currentRemoteDescription .

        4. Resolve p with undefined.

    7. Return p.

getDefaultIceServers

Returns a list of ICE servers that are configured into the browser. A browser might be configured to use local or private STUN or TURN servers. This method allows an application to learn about these servers and optionally use them.

This list is likely to be persistent and is the same across origins. It thus increases the fingerprinting surface of the browser. In privacy-sensitive contexts, browsers can consider mitigations such as only providing this data to whitelisted origins (or not providing it at all.)(This is a fingerprinting vector.)

Note

Since the use of this information is left to the discretion of application developers, configuring a user agent with these defaults does not per se increase a user's ability to limit the exposure of their IP addresses.

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 [JSEP] (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:

  1. Let connection be the RTCPeerConnection on which the method was invoked.

  2. If connection's [[IsClosed]] slot is true, throw an InvalidStateError.

  3. Set the configuration specified by configuration.

close

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

  1. Let connection be the RTCPeerConnection object on which the method was invoked.

  2. If connection's [[IsClosed]] slot is true, abort these steps.

  3. Set connection's [[IsClosed]] slot to true.

  4. Set connection's signaling state to "closed".

  5. Let transceivers be the result of executing the CollectTransceivers algorithm. For every RTCRtpTransceiver transceiver in transceivers, run the following steps:

    1. If transceiver's [[Stopped]] slot is true, abort these steps.

    2. Let sender be transceiver's [[Sender]].

    3. Let receiver be transceiver's [[Receiver]].

    4. Stop sending media with sender.

    5. Send an RTCP BYE for each RTP stream that was being sent by sender, as specified in [RFC3550].

    6. Stop receiving media with receiver.

    7. Set the readyState of receiver's [[ReceiverTrack]] to "ended".

    8. Set transceiver's [[Stopped]] slot to true.

  6. Set the [[ReadyState]] slot of each of connection's RTCDataChannel s to "closed".

  7. Set the [[SctpTransportState]] slot of each of connection's RTCSctpTransport s to "closed".

  8. Set the [[DtlsTransportState]] slot of each of connection's RTCDtlsTransport s to "closed".

  9. Destroy connection's ICE Agent, abruptly ending any active ICE processing and releasing any relevant resources (e.g. TURN permissions).

  10. Set the [[IceTransportState]] slot of each of connection's RTCIceTransport s to "closed".

4.4.3 Legacy Interface Extensions

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));
};
4.4.3.1 Method extensions
partial interface RTCPeerConnection {
    Promise<void> createOffer(RTCSessionDescriptionCallback successCallback,
                              RTCPeerConnectionErrorCallback failureCallback,
                              optional RTCOfferOptions options);
    Promise<void> setLocalDescription(RTCSessionDescriptionInit description,
                                      VoidFunction successCallback,
                                      RTCPeerConnectionErrorCallback failureCallback);
    Promise<void> createAnswer(RTCSessionDescriptionCallback successCallback,
                               RTCPeerConnectionErrorCallback failureCallback);
    Promise<void> setRemoteDescription(RTCSessionDescriptionInit description,
                                       VoidFunction successCallback,
                                       RTCPeerConnectionErrorCallback failureCallback);
    Promise<void> addIceCandidate((RTCIceCandidateInit or RTCIceCandidate) candidate,
                                  VoidFunction successCallback,
                                  RTCPeerConnectionErrorCallback failureCallback);
};
Methods
createOffer

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

  1. Let successCallback be the method's first argument.

  2. Let failureCallback be the callback indicated by the method's second argument.

  3. Let options be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection 's createOffer() method with options as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p with value offer, invoke successCallback with offer as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. Return a promise resolved with undefined.

setLocalDescription

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

  1. Let description be the method's first argument.

  2. Let successCallback be the callback indicated by the method's second argument.

  3. Let failureCallback be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection 's setLocalDescription method with description as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p, invoke successCallback with undefined as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. 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:

  1. Let successCallback be the method's first argument.

  2. Let failureCallback be the callback indicated by the method's second argument.

  3. Run the steps specified by RTCPeerConnection 's createAnswer() method with no arguments, and let p be the resulting promise.

  4. Upon fulfillment of p with value answer, invoke successCallback with answer as the argument.

  5. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  6. Return a promise resolved with undefined.

setRemoteDescription

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

  1. Let description be the method's first argument.

  2. Let successCallback be the callback indicated by the method's second argument.

  3. Let failureCallback be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection 's setRemoteDescription method with description as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p, invoke successCallback with undefined as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. Return a promise resolved with undefined.

addIceCandidate

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

  1. Let candidate be the method's first argument.

  2. Let successCallback be the callback indicated by the method's second argument.

  3. Let failureCallback be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection 's addIceCandidate() method with candidate as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p, invoke successCallback with undefined as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. Return a promise resolved with undefined.

Callback Definitions

These callbacks are only used on the legacy APIs.

RTCPeerConnectionErrorCallback
callback RTCPeerConnectionErrorCallback = void (DOMException error);
Callback RTCPeerConnectionErrorCallback Parameters
error of type DOMException
An error object encapsulating information about what went wrong.
RTCSessionDescriptionCallback
Callback RTCSessionDescriptionCallback Parameters
description of type RTCSessionDescriptionInit
The object containing the SDP [SDP].
4.4.3.2 Legacy configuration extensions
Attributes
offerToReceiveAudio of type boolean

Whenever this is given a non-false value, and the RTCPeerConnection has no non-stopped "sendrecv" or "recvonly" audio transceivers, createOffer() MUST as its first step invoke the equivalent of addTransceiver("audio") on the RTCPeerConnection object, except that this MUST NOT Update the negotiation-needed flag, and, provided this does not fail, proceed with createOffer()'s regular steps.

In all other situations, it will be disregarded.

offerToReceiveVideo of type boolean

Whenever this is given a non-false value, and the RTCPeerConnection has no non-stopped "sendrecv" or "recvonly" video transceivers, createOffer() MUST as its first step invoke the equivalent of addTransceiver("video") on the RTCPeerConnection object, except that this MUST NOT Update the negotiation-needed flag, and, provided this does not fail, proceed with createOffer()'s regular steps.

In all other situations, it will be disregarded.

4.4.4 Garbage collection

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.

4.5 Error Handling

4.5.1 General Principles

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.

Legacy-methods may only throw exceptions to indicate invalid state and other programming errors. For example, when a legacy-method is called when the RTCPeerConnection is in an invalid state or a state in which that particular method is not allowed to be executed, it will throw an exception. In all other cases, legacy methods MUST provide an error object to the error callback.

4.6 Session Description Model

4.6.1 RTCSdpType

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

enum RTCSdpType {
    "offer",
    "pranswer",
    "answer",
    "rollback"
};
Enumeration 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 and answer 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.

4.6.2 RTCSessionDescription Class

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

[Constructor(RTCSessionDescriptionInit descriptionInitDict),
 Exposed=Window]
interface RTCSessionDescription {
    readonly attribute RTCSdpType type;
    readonly attribute DOMString  sdp;
    [Default] object toJSON();
};
Constructors
RTCSessionDescription
The RTCSessionDescription() constructor takes a dictionary argument, descriptionInitDict, whose content is used to initialize the new RTCSessionDescription object. This constructor is deprecated; it exists for legacy compatibility reasons only.
Attributes
type of type RTCSdpType, readonly
The type of this RTCSessionDescription.
sdp of type DOMString, readonly
The string representation of the SDP [SDP].
Methods
toJSON()
When called, run [WEBIDL]'s default toJSON operation.
dictionary RTCSessionDescriptionInit {
    required RTCSdpType type;
             DOMString  sdp = "";
};
Dictionary RTCSessionDescriptionInit Members
type of type RTCSdpType, required
DOMString sdp
sdp of type DOMString
The string representation of the SDP [SDP]; if type is "rollback", this member is unused.

4.7 Session Negotiation Model

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.

4.7.1 Setting Negotiation-Needed

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.

4.7.2 Clearing Negotiation-Needed

This section is non-normative.

The negotiation-needed flag is cleared when an RTCSessionDescription of type "answer" is applied, and the supplied description matches the state of the RTCRtpTransceiver s and RTCDataChannel s 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.

4.7.3 Updating the Negotiation-Needed flag

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 queue a task to update the negotiation-needed flag.

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

  1. If connection's [[IsClosed]] slot is true, abort these steps.

  2. If connection's signaling state 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 an RTCSessionDescription.

  3. If the result of checking if negotiation is needed is false, clear the negotiation-needed flag by setting connection's [[NegotiationNeeded]] slot to false, and abort these steps.

  4. If connection's [[NegotiationNeeded]] slot is already true, abort these steps.

  5. Set connection's [[NegotiationNeeded]] slot to true.

  6. Queue a task that runs the following steps:

    1. If connection's [[IsClosed]] slot is true, abort these steps.

    2. If connection's [[NegotiationNeeded]] slot is false, abort these steps.

    3. Fire a simple event named negotiationneeded at connection.

    Note

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

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

  1. If any implementation-specific negotiation is required, as described at the start of this section, return true.

  2. Let description be connection's currentLocalDescription.

  3. If connection has created any RTCDataChannel s, and no m= section in description has been negotiated yet for data, return true.

  4. For each transceiver in connection's set of transceivers, perform the following checks:

    1. If transceiver isn't stopped and isn't yet associated with an m= section in description, return true.

    2. If transceiver isn't stopped and is associated with an m= section in description then perform the following checks:

      1. If transceiver's [[Direction]] slot is "sendrecv" or "sendonly", and the associated m= section in description doesn't contain an "a=msid" line, return true.

      2. If description is of type "offer", and the direction of the associated m= section in neither connection's currentLocalDescription nor currentRemoteDescription matches transceiver's [[Direction]] slot, return true.

      3. If description is of type "answer", and the direction of the associated m= section in the description does not match transceiver's [[Direction]] slot intersected with the offered direction (as described in [JSEP] (section 5.3.1.)), return true.

    3. If transceiver is stopped and is associated with an m= section, but the associated m= section is not yet rejected in connection's currentLocalDescription or currentRemoteDescription , return true.

  5. If all the preceding checks were performed and true was not returned, nothing remains to be negotiated; return false.

4.8 Interfaces for Connectivity Establishment

4.8.1 RTCIceCandidate Interface

This interface describes an ICE candidate, described in [ ICE] 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.

[Constructor(optional RTCIceCandidateInit candidateInitDict),
 Exposed=Window]
interface RTCIceCandidate {
    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?              ip;
    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;
    RTCIceCandidateInit toJSON();
};
Constructor
RTCIceCandidate

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

When invoked, run the following steps:

  1. If both the sdpMid and sdpMLineIndex dictionary members in candidateInitDict are null, throw a TypeError.
  2. Let iceCandidate be a newly created RTCIceCandidate object.
  3. Initialize the following attributes of iceCandidate to null: foundation, component, priority, ip, protocol, port, type, tcpType, relatedAddress, and relatedPort.
  4. Set the candidate, sdpMid, sdpMLineIndex, usernameFragment attributes of iceCandidate with the corresponding dictionary member values of candidateInitDict.
  5. 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 attributes in iceCandidate to the field values of the parsed result.
  6. 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.

Attributes

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

candidate of type DOMString, readonly
This carries the candidate-attribute as defined in section 15.1 of [ICE]. If this RTCIceCandidate represents an end-of-candidates indication, 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 RTCIceTransport s.
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.
ip of type DOMString, readonly, nullable

The IP address of the candidate. This corresponds to the connection-address field in candidate-attribute.

Note

The IP 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 IP 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 IP addresses can also be used as temporary or persistent cross-origin states, and thus contribute to the fingerprinting surface of the device.(This is a fingerprinting vector.)

Applications can avoid exposing IP 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 IP addresses exposed to the application itself, browsers can offer their users different policies regarding sharing local IP addresses, as defined in [ RTCWEB-IP-HANDLING].

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 [ICE].
Methods
toJSON()
To invoke the toJSON() operation of the RTCIceCandidate interface, run the following steps:
  1. Let json be a new RTCIceCandidateInit dictionary.
  2. For each attribute identifier attr in «"candidate", "sdpMid", "sdpMLineIndex", "description"»:
    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.
  3. Return json.
dictionary RTCIceCandidateInit {
    DOMString       candidate = "";
    DOMString?      sdpMid = null;
    unsigned short? sdpMLineIndex = null;
    DOMString       usernameFragment;
};
Dictionary RTCIceCandidateInit Members
candidate of type DOMString, defaulting to ""
This carries the candidate-attribute as defined in section 15.1 of [ICE]. 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
This carries the ufrag as defined in section 15.4 of [ICE].
4.8.1.1 candidate-attribute Grammar

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 [ICE]. 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.

4.8.1.2 RTCIceProtocol Enum

The RTCIceProtocol represents the protocol of the ICE candidate.

Enumeration description
udp A UDP candidate, as described in [ICE].
tcp A TCP candidate, as described in [RFC6544].
4.8.1.3 RTCIceTcpCandidateType Enum

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

Enumeration 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.

4.8.1.4 RTCIceCandidateType Enum

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

Enumeration description
host A host candidate, as defined in Section 4.1.1.1 of [ ICE].
srflx A server reflexive candidate, as defined in Section 4.1.1.2 of [ICE].
prflx A peer reflexive candidate, as defined in Section 4.1.1.2 of [ICE].
relay A relay candidate, as defined in Section 7.1.3.2.1 of [ ICE].

4.8.2 RTCPeerConnectionIceEvent

The icecandidate event of the RTCPeerConnection uses the RTCPeerConnectionIceEvent interface.

Firing an ice candidate event named e with an RTCIceCandidate candidate means that an event with the name e, which does not bubble (except where otherwise stated) and is not cancelable (except where otherwise stated), and which uses the RTCPeerConnectionIceEvent interface with the candidate attribute set to the new ICE candidate, MUST be created and dispatched at the given target.

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

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

4.8.3 RTCPeerConnectionIceErrorEvent

The icecandidateerror event of the RTCPeerConnection uses the RTCPeerConnectionIceErrorEvent interface.

[Constructor(DOMString type, RTCPeerConnectionIceErrorEventInit eventInitDict),
 Exposed=Window]
interface RTCPeerConnectionIceErrorEvent : Event {
    readonly attribute DOMString      hostCandidate;
    readonly attribute DOMString      url;
    readonly attribute unsigned short errorCode;
    readonly attribute USVString      errorText;
};
Constructors
RTCPeerConnectionIceErrorEvent
Attributes
hostCandidate of type DOMString, readonly

The hostCandidate attribute is the local IP address and port 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 use of multiple interfaces has been prohibited for privacy reasons, this attribute will be set to 0.0.0.0:0 or [::]:0, as appropriate.

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      hostCandidate;
             DOMString      url;
    required unsigned short errorCode;
             USVString      statusText;
};
Dictionary RTCPeerConnectionIceErrorEventInit Members
hostCandidate of type DOMString

The local IP address and port used to communicate with the STUN or TURN server.

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.

statusText of type USVString

The STUN reason text returned by the STUN or TURN server.

4.9 Priority and QoS Model

Many applications have multiple media flows of the same data type and often some of the flows are more important than others. WebRTC uses the priority and Quality of Service (QoS) framework described in [ RTCWEB-TRANSPORT] and [TSVWG-RTCWEB-QOS] to provide priority and DSCP marking for packets that will help provide QoS in some networking environments. The priority setting can be used to indicate the relative priority of various flows. The priority API allows the JavaScript applications to tell the browser whether a particular media flow is high, medium, low or of very low importance to the application by setting the priority property of RTCRtpEncodingParameters objects to one of the following values.

4.9.1 RTCPriorityType Enum

enum RTCPriorityType {
    "very-low",
    "low",
    "medium",
    "high"
};
Enumeration description
very-low See [RTCWEB-TRANSPORT], Section 4. Corresponds to "below normal" as defined in [RTCWEB-DATA].
low See [RTCWEB-TRANSPORT], Section 4. Corresponds to "normal" as defined in [RTCWEB-DATA].
medium See [RTCWEB-TRANSPORT], Section 4. Corresponds to "high" as defined in [RTCWEB-DATA].
high See [RTCWEB-TRANSPORT], Section 4. Corresponds to "extra high" as defined in [RTCWEB-DATA].

Applications that use this API should be aware that often better overall user experience is obtained by lowering the priority of things that are not as important rather than raising the priority of the things that are.

4.10 Certificate Management

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);
};

Methods

generateCertificate, static

The generateCertificate function causes the user agent to create and store 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 keygenAlgorithm value MUST be a valid argument to window.crypto.subtle.generateKey; that is, the value MUST produce a non-error result when normalized according to the WebCrypto algorithm normalization process [WebCryptoAPI] with an operation name of generateKey and a [[supportedAlgorithms]] value specific to production of certificates for RTCPeerConnection. If the algorithm normalization process produces an error, the call to generateCertificate MUST be rejected with that error.

Signatures produced by the generated key are used to authenticate the DTLS connection. The identified algorithm (as identified by the name of the normalized AlgorithmIdentifier) MUST be an asymmetric algorithm that can be used to produce a signature.

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.

A user agent MUST reject a call to generateCertificate() with a DOMException of type NotSupportedError if the keygenAlgorithm parameter identifies an algorithm that the user agent cannot or will not use to generate a certificate for RTCPeerConnection.

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.

4.10.1 RTCCertificateExpiration Dictionary

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

dictionary RTCCertificateExpiration {
    [EnforceRange]
    DOMTimeStamp expires;
};
expires

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 that the RTCCertificate is valid for relative to the current time.

When generateCertificate is called with an object argument, the user agent attempts to convert the object into an RTCCertificateExpiration . If this is unsuccessful, immediately return a promise that is rejected with a newly created TypeError and abort processing.

A user agent generates a certificate that has an expiration date set to the current time plus the value of the expires attribute. The expires attribute of the returned RTCCertificate is set to the expiration time of the certificate. A user agent MAY choose to limit the value of the expires attribute.

4.10.2 RTCCertificate Interface

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 ([[KeyingMaterial]]) and a certificate ([[Certificate]]]]) that RTCPeerConnection uses to authenticate with a peer.

[Exposed=Window]
interface RTCCertificate {
    readonly attribute DOMTimeStamp expires;
    static sequence<AlgorithmIdentifier> getSupportedAlgorithms();
    sequence<RTCDtlsFingerprint>  getFingerprints();
};
Attributes
expires of type DOMTimeStamp, 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.

Methods
getSupportedAlgorithms

Returns a sequence providing a representative set of supported certificate algorithms. At least one algorithm MUST be returned.

Note

For example, the "RSASSA-PKCS1-v1_5" algorithm dictionary, RsaHashedKeyGenParams, contains fields for the modulus length, public exponent, and hash algorithm. Implementations are likely to support a wide range of modulus lengths and exponents, but a finite number of hash algorithms. So in this case, it would be reasonable for the implementation to return one AlgorithmIdentifier for each supported hash algorithm that can be used with RSA, using default/recommended values for modulusLength and publicExponent (such as 1024 and 65537, respectively).

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.

Note that an RTCCertificate might not directly hold private keying material, this might be stored in a secure module.

The RTCCertificate object can be stored and retrieved from persistent storage by an application. When a user agent is required to obtain a structured clone [HTML51] of an RTCCertificate object, it performs the following steps:

  1. Let input and memory be the corresponding inputs defined by the internal structured cloning algorithm, where input represents an RTCCertificate object to be cloned.
  2. Let output be a newly constructed RTCCertificate object.
  3. Copy the value of the expires attribute from input to output.
  4. Let the [[Certificate]] internal slot of output be set to the result of invoking the internal structured clone algorithm recursively on the corresponding internal slots of input, with the slot contents as the new " input" argument and memory as the new " memory" argument.
  5. Let the [[KeyingMaterial]] internal slot of output refer to the same private keying material represented by the [[KeyingMaterial]] internal slot of input.

5. RTP Media API

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.

When sending media, the sender may need to rescale or resample the media to meet various requirements including the envelope negotiated by SDP.

Following the rules in [JSEP] (section 3.6.), the video MAY be downscaled in order to fit the SDP constraints. 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.

When video is rescaled, for example for certain combinations of width or height and scaleResolutionDownBy values, situations when the resulting width or height is not an integer may occur. In such situations the user agent MUST use the integer part of the result ( https://tc39.github.io/ecma262/#eqn-floor). What to transmit if the integer part of the scaled width or height is zero is implementation-specific.

The actual encoding and transmission of MediaStreamTracks is managed through objects called RTCRtpSender s. Similarly, the reception and decoding of MediaStreamTracks is managed through objects called RTCRtpReceiver s. 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; volume, 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. RTCRtpSender s and RTCRtpReceiver s are always created at the same time as an RTCRtpTransceiver , which they will remain attached to for their lifetime. RTCRtpTransceiver s 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.

Note

There are several ways to initiate the sending of a MediaStreamTrack over a peer-to-peer connection. One way is to use the addTrack method on the RTCPeerConnection . Another way is to use the replaceTrack method on an existing RTCRtpSender . Yet another way is to create a new RTCRtpSender via the addTransceiver method (with or without a MediaStreamTrack argument). While addTrack checks if the MediaStreamTrack given as an argument is already being sent to avoid sending the same MediaStreamTrack twice, the other ways do not, allowing the same MediaStreamTrack (possibly using different RTCRtpParameters with different RTCRtpSender s) to be sent several times simultaneously. Doing this implies that at the receiving end of the peer-to-peer connection there are several MediaStreamTracks with an identical id.

5.1 RTCPeerConnection Interface Extensions

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);
    void                        removeTrack(RTCRtpSender sender);
    RTCRtpTransceiver           addTransceiver((MediaStreamTrack or DOMString) trackOrKind,
                                               optional RTCRtpTransceiverInit init);
    attribute EventHandler ontrack;
};

Attributes

ontrack of type EventHandler

The event type of this event handler is track .

Methods

getSenders

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

The getSenders method MUST return the result of executing the CollectSenders algorithm.

We define the CollectSenders algorithm as follows:

  1. Let transceivers be the result of executing the CollectTransceivers algorithm.
  2. Let senderset be a new empty set.
  3. For each transceiver in transceivers,
    1. Let sender be transceiver's [[Sender]].
    2. Add sender to senderset.
  4. Let senders be a new sequence consisting of all the RTCRtpSender objects in senderset. The conversion from the senders set to the sequence is user agent defined and the order does not have to be stable between calls.
  5. Return senders.
getReceivers

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

The getReceivers method MUST return the result of executing the CollectReceivers algorithm.

We define the CollectReceivers algorithm as follows:

  1. Let transceivers be the result of executing the CollectTransceivers algorithm.
  2. Let receiverset be a new empty set.
  3. For each transceiver in transceivers,
    1. Let receiver be transceiver's [[Receiver]].
    2. Add receiver to receiverset.
  4. Let receivers be a new sequence consisting of all the RTCRtpReceiver objects in receiverset. The conversion from the receivers set to the sequence is user agent defined and the order does not have to be stable between calls.
  5. 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:

  1. Let transceivers be a new sequence that represents a snapshot of all the RTCRtpTransceiver objects in this RTCPeerConnection object's set of transceivers. The conversion from the transceiver set to the sequence is user agent defined and the order does not have to be stable between calls.
  2. 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:

  1. Let connection be the RTCPeerConnection object on which this method was invoked.

  2. Let track be the MediaStreamTrack object indicated by the method's first argument.

  3. 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.

  4. If connection's [[IsClosed]] slot is true, throw an InvalidStateError.

  5. Let senders be the result of executing the CollectSenders algorithm. If an RTCRtpSender for track already exists in senders, throw an InvalidAccessError.

  6. 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 track added, as defined in [JSEP] (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:

  7. If sender is not null, run the following steps to use that sender:

    1. Set sender's [[SenderTrack]] to track.

    2. Set sender's [[AssociatedMediaStreams]] to streams.

    3. Let transceiver be the RTCRtpTransceiver associated with sender.

    4. If transceiver's [[Direction]] slot is recvonly, set transceiver's [[Direction]] slot to sendrecv.

    5. If transceiver's [[Direction]] slot is inactive, set transceiver's [[Direction]] slot to sendonly.

  8. If sender is null, run the following steps:

    1. Create an RTCRtpSender with track and streams and let sender be the result.

    2. Create an RTCRtpReceiver with track.kind as 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

  9. A track could have contents that are inaccessible to the application. This can be due to being marked with a peerIdentity option or 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, unless they are also marked with peerIdentity and they meet the requirements for sending (see isolated streams and RTCPeerConnection).

    All other tracks that are not accessible to the application MUST NOT be sent to the peer, with silence (audio), black frames (video) or equivalently absent content being sent in place of track content.

    Note that this property can change over time.

  10. Update the negotiation-needed flag for connection.

  11. 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 [JSEP] (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 muted event is fired at the track.

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

  1. Let sender be the argument to removeTrack.

  2. Let connection be the RTCPeerConnection object on which the method was invoked.

  3. If connection's [[IsClosed]] slot is true, throw an InvalidStateError.

  4. If sender was not created by connection, throw an InvalidAccessError.

  5. Let senders be the result of executing the CollectSenders algorithm.

  6. If sender is not in senders (which indicates that it was removed due to setting an RTCSessionDescription of type "rollback"), then abort these steps.

  7. If sender's [[SenderTrack]] is null, abort these steps.

  8. Set sender's [[SenderTrack]] to null.

  9. Let transceiver be the RTCRtpTransceiver object corresponding to sender.

  10. If transceiver's [[Direction]] slot is sendrecv, set transceiver's [[Direction]] slot to recvonly.

  11. If transceiver's [[Direction]] slot is sendonly, set transceiver's [[Direction]] slot to inactive.

  12. 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 [JSEP] (section 5.2.2.).

The initial value of mid is null. Setting a new RTCSessionDescription may change it to a non-null value, as defined in [JSEP] (section 5.5. and section 5.6.).

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:

  1. Let init be the second argument.

  2. Let streams be init's streams member.

  3. Let sendEncodings be init's sendEncodings member.

  4. Let direction be init's direction member.

  5. If the first argument is a string, let it be kind and run the following steps:

    1. If kind is not a legal MediaStreamTrack kind, throw a TypeError.

    2. Let track be null.

  6. If the first argument is a MediaStreamTrack , let it be track and let kind be track.kind.

  7. Verify that each rid value in sendEncodings is composed only of alphanumeric characters (a-z, A-Z, 0-9) up to a maximum of 16 characters. If one of the RIDs does not meet these requirements, throw a TypeError.

  8. If any RTCRtpEncodingParameters dictionary in sendEncodings contains a read-only parameter other than rid , throw an InvalidAccessError.

  9. Verify that each scaleResolutionDownBy value in sendEncodings is greater than or equal to 1.0. If one of the scaleResolutionDownBy values does not meet this requirement, throw a RangeError.

  10. Create an RTCRtpSender with track, 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 [JSEP] (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 [JSEP] (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.

  11. Create an RTCRtpReceiver with kind and let receiver be the result. This specification does not define how to configure createOffer to receive multiple RTP encodings. However when setRemoteDescription is called with a corresponding remote description that is able to send multiple RTP encodings as defined in [JSEP], the RTCRtpReceiver may receive multiple RTP encodings and the parameters retrieved via the transceiver's receiver.getParameters() will reflect the encodings negotiated.

  12. Create an RTCRtpTransceiver with sender, receiver and direction, and let transceiver be the result.

  13. Add transceiver to connection's set of transceivers

  14. Update the negotiation-needed flag for connection.

  15. Return transceiver.

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

Dictionary RTCRtpTransceiverInit Members

direction of type RTCRtpTransceiverDirection, defaulting to "sendrecv"
The direction of the RTCRtpTransceiver.
streams of type sequence<MediaStream>

When the remote PeerConnection's ontrack 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"
};
RTCRtpTransceiverDirection Enumeration 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.

5.1.1 Processing Remote MediaStreamTracks

An application can reject incoming media descriptions by calling RTCRtpTransceiver.stop() to stop both directions, or set the transceiver's direction to "sendonly" to reject only the incoming side.

To process the addition of a remote track for an incoming media description [JSEP] (section 5.9.) given RTCRtpTransceiver transceiver and trackEvents, the user agent MUST run the following steps:

  1. Let receiver be transceiver's [[Receiver]].

  2. Let track be receiver's [[ReceiverTrack]].

  3. Set the associated remote streams given receiver and a list of the MSIDs that the media description indicates track is to be associated with.

  4. Let streams be receiver's [[AssociatedRemoteMediaStreams]] slot.

  5. Add a new RTCTrackEvent with transceiver, track, and streams to trackEvents.

To process the removal of a remote track for an incoming media description [JSEP] (section 5.9.) given RTCRtpTransceiver transceiver, the user agent MUST run the following steps:

  1. Let receiver be transceiver's [[Receiver]].

  2. Let track be receiver's [[ReceiverTrack]].

  3. Set the associated remote streams for the media description, given receiver and an empty list.

  4. If track.muted is false, update the muted state of track with the value true.

To set the associated remote streams given RTCRtpReceiver receiver and a list msids, the user agent MUST run the following steps:

  1. Let connection be the RTCPeerConnection object associated with receiver.

  2. 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.

  3. Let streams be a list of the MediaStream objects created for this connection with the ids corresponding to msids.

  4. For each stream in receiver's [[AssociatedRemoteMediaStreams]] that is not present in streams, remove track from stream.

    Note

    This will result in a removetrack event being fired at each MediaStream as described in [GETUSERMEDIA].

  5. For each stream in streams that is not present in receiver's [[AssociatedRemoteMediaStreams]], add track to stream.

    Note

    This will result in an addtrack event being fired at each MediaStream as described in [GETUSERMEDIA].

  6. Set receiver's [[AssociatedRemoteMediaStreams]] slot to streams.

5.2 RTCRtpSender Interface

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 list of MediaStream objects, streams, and optionally a list of RTCRtpEncodingParameters objects, sendEncodings, run the following steps:

  1. Let sender be a new RTCRtpSender object.

  2. Let sender have a [[SenderTrack]] internal slot initialized to track.

  3. Let sender have a [[SenderTransport]] internal slot initialized to null.

  4. Let sender have a [[SenderRtcpTransport]] internal slot initialized to null.

  5. Let sender have an [[AssociatedMediaStreams]] internal slot, representing a list of MediaStream objects that the MediaStreamTrack object of this sender is associated with. The [[AssociatedMediaStreams]] slot is used when sender is represented in SDP as described in [JSEP] (section 5.2.1.).

  6. Set sender's [[AssociatedMediaStreams]] slot to streams.

  7. Let sender have a [[SendEncodings]] internal slot, representing a list of RTCRtpEncodingParameters dictionaries.

  8. 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 RTCRtpEncodingParameters with active set to true.

    Note
    Providing a single, default RTCRtpEncodingParameters allows the application to set encoding parameters using setParameters , even when simulcast isn't used.
  9. Let sender have a [[LastReturnedParameters]] internal slot, which will be used to match getParameters and setParameters transactions.

  10. Return sender.

[Exposed=Window]
interface RTCRtpSender {
    readonly attribute MediaStreamTrack? track;
    readonly attribute RTCDtlsTransport? transport;
    readonly attribute RTCDtlsTransport? rtcpTransport;
    // Feature at risk
    static RTCRtpCapabilities getCapabilities(DOMString kind);
    Promise<void>           setParameters(optional RTCRtpParameters parameters);
    RTCRtpParameters        getParameters();
    Promise<void>           replaceTrack(MediaStreamTrack? withTrack);
    Promise<RTCStatsReport> getStats();
};

Attributes

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 track.muted is set to true, the RTCRtpSender sends silence (audio) or a black frame (video). 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.

rtcpTransport of type RTCDtlsTransport, readonly, nullable

The rtcpTransport attribute is the transport over which RTCP is sent and received. Prior to construction of the RTCDtlsTransport object, the rtcpTransport attribute will be null. When RTCP mux is used (or bundling, which mandates RTCP mux), rtcpTransport will be null, and both RTP and RTCP traffic will flow over the transport described by transport.

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

Methods

getCapabilities, static

The getCapabilities() method returns the most optimistic view of the capabilities of the system for sending media of the given kind. It does not reserve any resources, ports, or other state but is meant to provide a way to discover the types of capabilities of the browser including which codecs may be supported. User agents MUST support kind values of "audio" and "video". If the system has no capabilities corresponding to the value of the kind argument, getCapabilities returns null.

These capabilities provide generally persistent cross-origin information on the device and thus increases the fingerprinting surface of the application. In privacy-sensitive contexts, browsers can consider mitigations such as reporting only a common subset of the capabilities.(This is a fingerprinting vector.)

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:

  1. Let parameters be the method's first argument.
  2. Let sender be the RTCRtpSender object on which setParameters is invoked.
  3. Let transceiver be the RTCRtpTransceiver object associated with sender (i.e. sender is transceiver's [[Sender]]).
  4. Let N be the number of RTCRtpEncodingParameters stored in sender's internal [[SendEncodings]] slot.
  5. If transceiver's [[Stopped]] slot is true, return a promise rejected with a newly created InvalidStateError.
  6. If sender's [[LastReturnedParameters]] internal slot is empty, meaning getParameters has never been called, return a promise rejected with a newly created InvalidStateError.
  7. If parameters.encodings.length is different from N, or if any parameter in parameters, is marked as a Read-only parameter, has a value that is different from the corresponding parameter value in sender's [[LastReturnedParameters]] internal slot, return a promise rejected with a newly created InvalidModificationError. Note that this also applies to transactionId.
  8. For each value of i from 0 to the number of encodings, check whether parameters.encodings[i].codecPayloadType (if set) corresponds to a value of parameters.codecs[j].payloadType where j goes from 0 to the number of codecs. If there is no correspondence, or if the MIME subtype portion of parameters.codecs[j].mimeType is equal to "red", "cn", "telephone-event", "rtx" or a forward error correction codec ("ulpfec" [RFC5109] or "flexfec" [FLEXFEC]), reject p with a newly created InvalidAccessError.

  9. If the scaleResolutionDownBy parameter in the parameters argument has a value less than 1.0, return a promise rejected with a newly created RangeError.
  10. Let p be a new promise.
  11. In parallel, configure the media stack to use parameters to transmit sender's [[SenderTrack]].
    1. If the media stack is successfully configured with parameters, queue a task to run the following steps:
      1. Set sender's internal [[SendEncodings]] slot to parameters.encodings.
      2. 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.
  12. Return p.

If the application selects a codec via codecPayloadType , and this codec is removed from a subsequent offer/answer negotiation, codecPayloadType will be unset in the next call to getParameters , and the implementation will fall back to its default codec selection policy until a new codec is selected.

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 RTCRtpParameters 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 RTCRtpParameters dictionary is constructed as follows:

  • transactionId is set to a new unique identifier, used to match this getParameters call to a setParameters call that may occur later.
  • 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.
  • The codecs sequence is populated based on the codecs that have been negotiated for sending, and which the user agent is currently capable of sending.
  • 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.
  • degradationPreference is set to the last value passed into setParameters, or the default value of "balanced" if setParameters hasn't been called.

The returned RTCRtpParameters dictionary MUST be stored in the RTCRtpSender object's [[LastReturnedParameters]] internal slot.

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

Example 3
var params = sender.getParameters();
// ... make changes to RTCRtpParameters
params.encodings[0].active = false;
sender.setParameters(params)

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

replaceTrack

Attempts to replace the track being sent with another track provided (or with a null track), without renegotiation.

To avoid track identifiers changing on the remote receiving end when a track is replaced, the sender MUST retain the original track identifier and stream associations and use these in subsequent negotiations.

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

  1. Let sender be the RTCRtpSender object on which replaceTrack is invoked.
  2. Let transceiver be the RTCRtpTransceiver object associated with sender.

  3. Let connection be the RTCPeerConnection object that created sender.

  4. If connection's [[IsClosed]] slot is true, return a promise rejected with a newly created InvalidStateError and abort these steps.

  5. If transceiver's [[Stopped]] slot is true, return a promise rejected with a newly created InvalidStateError.

  6. Let withTrack be the argument to this method.

  7. If withTrack is non-null and withTrack.kind differs from the transceiver kind of transceiver, return a promise rejected with a newly created TypeError.

  8. If transceiver is not yet associated with a media description, then set sender's track attribute to withTrack, and return a promise resolved with undefined.

  9. Let p be a new promise.

  10. Run the following steps in parallel:

    1. Determine if negotiation is needed to transmit withTrack in place of the sender's existing track. Negotiation is not needed if withTrack is null or if the sender's existing track is ended (which appears as though the track was muted). Ignore which MediaStream the track resides in and the id attribute of the track in this determination. If negotiation is needed, then reject p with a newly created InvalidModificationError and abort these steps.

    2. If withTrack is null, have the sender stop sending, without negotiating. Otherwise, have the sender switch seamlessly to transmitting withTrack instead of the sender's existing track, without negotiating.

    3. Queue a task that runs the following steps:

      1. If connection's [[IsClosed]] slot is true, abort these steps.

      2. Set sender's track attribute to withTrack.

      3. Resolve p with undefined.

  11. Return p.

Note

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

  1. Changing a resolution to a value outside of the negotiated imageattr bounds, as described in [ RFC6236].
  2. Changing a frame rate to a value that causes the block rate for the codec to be exceeded.
  3. A video track differing in raw vs. pre-encoded format.
  4. An audio track having a different number of channels.
  5. 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.
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:

  1. Let selector be the RTCRtpSender object on which the method was invoked.

  2. 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. Resolve p with the resulting RTCStatsReport object, containing the gathered stats.

  3. Return p.

dictionary RTCRtpParameters {
    DOMString                                 transactionId;
    sequence<RTCRtpEncodingParameters>        encodings;
    sequence<RTCRtpHeaderExtensionParameters> headerExtensions;
    RTCRtcpParameters                         rtcp;
    sequence<RTCRtpCodecParameters>           codecs;
    RTCDegradationPreference                  degradationPreference;
};

Dictionary RTCRtpParameters Members

transactionId of type DOMString

An 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>

A sequence containing parameters for RTP encodings of media.

headerExtensions of type sequence<RTCRtpHeaderExtensionParameters>

A sequence containing parameters for RTP header extensions. Read-only parameter.

rtcp of type RTCRtcpParameters

Parameters used for RTCP. Read-only parameter.

codecs of type sequence<RTCRtpCodecParameters>

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.

degradationPreference of type RTCDegradationPreference

When bandwidth is constrained and the RtpSender needs to choose between degrading resolution or degrading framerate, degradationPreference indicates which is preferred. If unset, the RtpSender defaults to the balanced policy.

For an RtpReceiver, degradationPreference is inapplicable and will always be undefined.

dictionary RTCRtpEncodingParameters {
    octet           codecPayloadType;
    RTCDtxStatus    dtx;
    boolean         active = true;
    RTCPriorityType priority = "low";
    unsigned long   ptime;
    unsigned long   maxBitrate;
    double          maxFramerate;
    DOMString       rid;
    double          scaleResolutionDownBy;
};

Dictionary RTCRtpEncodingParameters Members

codecPayloadType of type octet

For an RTCRtpSender , used to select a codec to be sent. Must reference a payload type from the codecs member of RTCRtpParameters . If left unset, the implementation will select a codec according to its default policy. This field is not used for RTCRtpReceiver s.

dtx of type RTCDtxStatus

For an RTCRtpSender , indicates whether discontinuous transmission will be used. Setting it to disabled causes discontinuous transmission to be turned off. Setting it to enabled causes discontinuous transmission to be turned on if it was negotiated (either via a codec-specific parameter or via negotiation of the CN codec); if it was not negotiated (such as when setting voiceActivityDetection to false), then discontinuous operation will be turned off regardless of the value of dtx, and media will be sent even when silence is detected. This attribute is ignored by a receiver or video sender.

active of type boolean, defaulting to true

For an RTCRtpSender , 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. For an RTCRtpReceiver , a value of true indicates that this encoding is being decoded. A value of false indicates this encoding is no longer being decoded.

priority of type RTCPriorityType, defaulting to "low"

Indicates the priority of this encoding. It is specified in [ RTCWEB-TRANSPORT], Section 4.

ptime of type unsigned long

For an RTCRtpSender , indicates the preferred duration of media represented by a packet in milliseconds for this encoding. Typically, this is only relevant for audio encoding. The user agent MUST use this duration if possible, and otherwise use the closest available duration. This value MUST take precedence over any "ptime" attribute in the remote description, whose processing is described in [JSEP] (section 5.9.). Note that the user agent MUST still respect the limit imposed by any "maxptime" attribute, as defined in [RFC4566], Section 6.

maxBitrate of type unsigned long

Indicates the maximum bitrate that can be used to send this encoding. The encoding may also be further constrained by other limits (such as maxFramerate or 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.

maxFramerate of type double

Indicates the maximum framerate that can be used to send this encoding, in frames per second.

rid of type DOMString

If set, this RTP encoding will be sent with the RID header extension as defined by [JSEP] (section 5.2.1.). The RID is not modifiable via setParameters. It can only be set or modified in addTransceiver.

scaleResolutionDownBy of type double

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, the sender will not apply any scaling, (i.e., scaleResolutionDownBy will be 1.0).

Usage of the attributes is defined in the table below:

Attribute Type Receiver/Sender Read/Write
codecPayloadType octet Sender Read/Write
dtx RTCDtxStatus Sender Read/Write
active boolean Sender Read/Write
priority RTCPriorityType Sender Read/Write
ptime unsigned long Sender Read/Write
maxBitrate unsigned long Sender Read/Write
maxFramerate double Sender Read/Write
scaleResolutionDownBy double Sender Read/Write
rid DOMString Receiver/Sender Read-only
enum RTCDtxStatus {
    "disabled",
    "enabled"
};
RTCDtxStatus Enumeration description
disabled

Discontinuous transmission is disabled.

enabled

Discontinuous transmission is enabled if negotiated.

enum RTCDegradationPreference {
    "maintain-framerate",
    "maintain-resolution",
    "balanced"
};
RTCDegradationPreference Enumeration description
maintain-framerate

Degrade resolution in order to maintain framerate.

maintain-resolution

Degrade framerate in order to maintain resolution.

balanced

Degrade a balance of framerate and resolution.

dictionary RTCRtcpParameters {
    DOMString cname;
    boolean   reducedSize;
};

Dictionary RTCRtcpParameters Members

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 RTCRtpHeaderExtensionParameters {
    DOMString      uri;
    unsigned short id;
    boolean        encrypted;
};

Dictionary RTCRtpHeaderExtensionParameters Members

uri of type DOMString

The URI of the RTP header extension, as defined in [ RFC5285]. Read-only parameter.

id of type unsigned short

The value put in the RTP packet to identify the header extension. Read-only parameter.

encrypted of type boolean

Whether the header extension is encryted or not. Read-only parameter.

dictionary RTCRtpCodecParameters {
    octet          payloadType;
    DOMString      mimeType;
    unsigned long  clockRate;
    unsigned short channels;
    DOMString      sdpFmtpLine;
};

Dictionary RTCRtpCodecParameters Members

payloadType of type octet

The RTP payload type used to identify this codec. Read-only parameter.

mimeType of type DOMString

The codec MIME media type/subtype. Valid media types and subtypes are listed in [IANA-RTP-2]. Read-only parameter.

clockRate of type unsigned long

The codec clock rate expressed in Hertz. Read-only parameter.

channels of type unsigned short

The number of channels (mono=1, stereo=2). Read-only parameter.

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 [JSEP] (section 5.7.). For an RTCRtpSender, these parameters come from the remote description, and for an RTCRtpReceiver, they come from the local description. Read-only parameter.

dictionary RTCRtpCapabilities {
    sequence<RTCRtpCodecCapability>           codecs;
    sequence<RTCRtpHeaderExtensionCapability> headerExtensions;
};

Dictionary RTCRtpCapabilities Members

codecs of type sequence<RTCRtpCodecCapability>

Supported media codecs as well as entries for RTX, RED and FEC mechanisms. There will only be a single entry in codecs[] for retransmission via RTX, with sdpFmtpLine not present.

headerExtensions of type sequence<RTCRtpHeaderExtensionCapability>

Supported RTP header extensions.

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

Dictionary RTCRtpCodecCapability Members

The RTCRtpCodecCapability dictionary provides information about codec capabilities. Only capability combinations that would utilize distinct payload types in a generated SDP offer are provided. For example:

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

The codec MIME media type/subtype. Valid media types and subtypes are listed in [IANA-RTP-2].

clockRate of type unsigned long

The codec clock rate expressed in Hertz.

channels of type unsigned short

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.

dictionary RTCRtpHeaderExtensionCapability {
    DOMString uri;
};

Dictionary RTCRtpHeaderExtensionCapability Members

uri of type DOMString

The URI of the RTP header extension, as defined in [ RFC5285].

5.3 RTCRtpReceiver Interface

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

To create an RTCRtpReceiver with kind, kind, and optionally an id string, id, run the following steps:

  1. Let receiver be a new RTCRtpReceiver object.

  2. Let track be a new MediaStreamTrack object [GETUSERMEDIA]. The source of track is a remote source provided by receiver.

  3. Initialize track.kind to kind.

  4. If an id string, id, was given as input to this algorithm, initialize track.id to id. (Otherwise the value generated when track was created will be used.)

  5. Initialize track.label to the result of concatenating the string "remote " with kind.

  6. Initialize track.readyState to live.

  7. Initialize track.muted to true. See the MediaStreamTrack section about how the muted attribute reflects if a MediaStreamTrack is receiving media data or not.

  8. Let receiver have a [[ReceiverTrack]] internal slot initialized to track.

  9. Let receiver have a [[ReceiverTransport]] internal slot initialized to null.

  10. Let receiver have a [[ReceiverRtcpTransport]] internal slot initialized to null.

  11. 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.

  12. Return receiver.

[Exposed=Window]
interface RTCRtpReceiver {
    readonly attribute MediaStreamTrack  track;
    readonly attribute RTCDtlsTransport? transport;
    readonly attribute RTCDtlsTransport? rtcpTransport;
    // Feature at risk
    static RTCRtpCapabilities             getCapabilities(DOMString kind);
    RTCRtpParameters                      getParameters();
    sequence<RTCRtpContributingSource>    getContributingSources();
    sequence<RTCRtpSynchronizationSource> getSynchronizationSources();
    Promise<RTCStatsReport>               getStats();
};

Attributes

track of type MediaStreamTrack, readonly

The track attribute is the track that is associated with this RTCRtpReceiver object receiver. When one of the SSRCs for RTP source media streams received by receiver is removed (either due to reception of a BYE or via timeout), the mute event is fired at track. If and when packets are received again, the unmute event is fired at track.

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.

rtcpTransport of type RTCDtlsTransport, readonly, nullable

The rtcpTransport attribute is the transport over which RTCP is sent and received. Prior to construction of the RTCDtlsTransport object, the rtcpTransport attribute will be null. When RTCP mux is used (or bundling, which mandates RTCP mux), rtcpTransport will be null, and both RTP and RTCP traffic will flow over transport.

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

Methods

getCapabilities, static

The getCapabilities() method returns the most optimistic view of the capabilities of the system for receiving media of the given kind. It does not reserve any resources, ports, or other state but is meant to provide a way to discover the types of capabilities of the browser including which codecs may be supported. User agents MUST support kind values of "audio" and "video". If the system has no capabilities corresponding to the value of the kind argument, getCapabilities returns null.

These capabilities provide generally persistent cross-origin information on the device and thus increases the fingerprinting surface of the application. In privacy-sensitive contexts, browsers can consider mitigations such as reporting only a common subset of the capabilities.(This is a fingerprinting vector.)

getParameters

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

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

  • encodings is populated based on RIDs present in the current remote description. Every member of the RTCRtpEncodingParameters dictionaries other than the RID fields is left undefined.

  • The headerExtensions sequence is populated based on the header extensions that the receiver is currently prepared to receive.
  • The codecs sequence is populated based on the codecs that the receiver is currently prepared to receive.

    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 undefined.
  • transactionId and degradationPreference are left undefined.
getContributingSources

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

getSynchronizationSources

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

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:

  1. Let selector be the RTCRtpReceiver object on which the method was invoked.

  2. 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. Resolve p with the resulting RTCStatsReport object, containing the gathered stats.

  3. Return p.

The RTCRtpContributingSource and RTCRtpSynchronizationSource objects contain information about a given contributing source (CSRC) or synchronization source (SSRC), including the most recent time a packet that the source contributed to was played out. The browser MUST keep information from RTP packets received in the previous 10 seconds. When the first audio frame contained in an RTP packet is delivered to the RTCRtpReceiver 's MediaStreamTrack for playout, the user agent MUST queue a task to update the relevant RTCRtpContributingSource and RTCRtpSynchronizationSource objects based on the contents of the packet. The RTCRtpSynchronizationSource object corresponding to the SSRC identifier is updated each time, and if the RTP packet contains CSRC identifiers, then the RTCRtpContributingSource objects corresponding to those CSRC identifiers are also updated.

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 implementaion does not need to literally queue a task for every packet, as long as the end result is that within a single event loop task execution, all RTCRtpSynchronizationSource and RTCRtpContributingSource objects for a particular RTCRtpReceiver return information from a single point in the RTP stream.
[Exposed=Window]
interface RTCRtpContributingSource {
    readonly attribute DOMHighResTimeStamp timestamp;
    readonly attribute unsigned long       source;
    readonly attribute byte?               audioLevel;
};

Attributes

timestamp of type DOMHighResTimeStamp, readonly

The timestamp of type DOMHighResTimeStamp [HIGHRES-TIME], indicating the most recent time of playout of an RTP packet containing the source. The timestamp is defined in [ HIGHRES-TIME] and corresponds to a local clock.

source of type unsigned long, readonly

The CSRC identifier of the contributing source.

audioLevel of type byte, readonly , nullable

The audio level contained in the last RTP packet played from this source. audioLevel will be the level value defined in [RFC6465] if the RFC 6465 header extension is present, and otherwise null. RFC 6465 defines the level as a 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.

[Exposed=Window]
interface RTCRtpSynchronizationSource {
    readonly attribute DOMHighResTimeStamp timestamp;
    readonly attribute unsigned long       source;
    readonly attribute byte                audioLevel;
    readonly attribute boolean?            voiceActivityFlag;
};

Attributes

timestamp of type DOMHighResTimeStamp, readonly

The timestamp of type DOMHighResTimeStamp [HIGHRES-TIME], indicating the most recent time of playout of an RTP packet from the source. The timestamp is defined in [ HIGHRES-TIME] and corresponds to a local clock.

source of type unsigned long, readonly

The SSRC identifier of the synchronization source.

audioLevel of type byte, readonly, nullable

The audio level contained in the last RTP packet played from this source. audioLevel will be the level value defined in [RFC6464], if the RFC 6464 header extension is present. If the RFC 6464 extension header is not present, the browser will compute a value for audioLevel as if it had come from RFC 6464.

voiceActivityFlag of type boolean, readonly, nullable

Whether the last RTP packet played from this source contains voice activity (true) or not (false). If the RFC 6464 extension header was not present, or if the peer has signaled that it is not using the V bit by setting the "vad" extension attribute to "off", as described in [RFC6464], Section 4, voiceActivityFlag will be null.

5.4 RTCRtpTransceiver Interface

The RTCRtpTransceiver interface represents a combination of an RTCRtpSender and an RTCRtpReceiver that share a common mid. As defined in [JSEP] (section 3.4.1.), an RTCRtpTransceiver is said to be associated with a media description if its mid property is non-null; otherwise it is said to be disassociated. Conceptually, an associated transceiver is one that's represented in the last applied session description.

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:

  1. Let transceiver be a new RTCRtpTransceiver object.

  2. Let transceiver have a [[Sender]] internal slot, initialized to sender.

  3. Let transceiver have a [[Receiver]] internal slot, initialized to receiver.

  4. Let transceiver have a [[Stopped]] internal slot, initialized to false.

  5. Let transceiver have a [[Direction]] internal slot, initialized to direction.

  6. Let transceiver have a [[CurrentDirection]] internal slot, initialized to null.

  7. 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 an RTCSessionDescription.
[Exposed=Window]
interface RTCRtpTransceiver {
    readonly attribute DOMString?                  mid;
    [SameObject]
    readonly attribute RTCRtpSender                sender;
    [SameObject]
    readonly attribute RTCRtpReceiver              receiver;
    readonly attribute boolean                     stopped;
    readonly attribute RTCRtpTransceiverDirection  direction;
    readonly attribute RTCRtpTransceiverDirection? currentDirection;
    void setDirection(RTCRtpTransceiverDirection direction);
    void stop();
    void setCodecPreferences(sequence<RTCRtpCodecCapability> codecs);
};

Attributes

mid of type DOMString, readonly, nullable

The mid attribute is the mid negotatiated and present in the local and remote descriptions as defined in [JSEP] (section 5.2.1. and section 5.3.1.). Before negotiation is complete, the mid value may be null. After rollbacks, the value may change from a non-null value to null.

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.

stopped of type boolean, readonly

The stopped attribute indicates that the sender of this transceiver will no longer send, and that the receiver will no longer receive. It is true if either stop has been called or if setting the local or remote description has caused the RTCRtpTransceiver to be stopped. On getting, this attribute MUST return the value of the [[Stopped]] slot.

direction of type RTCRtpTransceiverDirection, readonly

As defined in [JSEP] (section 4.2.4.), the direction attribute indicates the preferred direction of this transceiver, which will be used in calls to createOffer and createAnswer . On getting, this attribute MUST return the value of the [[Direction]] slot.

currentDirection of type RTCRtpTransceiverDirection, readonly, nullable

As defined in [JSEP] (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, or if the transceiver is stopped , the value is null. On getting, this attribute MUST return the value of the [[CurrentDirection]] slot.

Methods

setDirection

The setDirection method sets the direction of the RTCRtpTransceiver. Calls to setDirection() do 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 [JSEP] (section 5.2.2. and section 5.3.2.). Calling setDirection() updates the negotiation-needed flag for the RTCRtpTransceiver's associated RTCPeerConnection .

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

  1. Let transceiver be the RTCRtpTransceiver object on which the method is invoked.

  2. Let connection be the RTCPeerConnection object associated with transceiver.

  3. If connection's [[IsClosed]] slot is true, throw an InvalidStateError.

  4. If transceiver's [[Stopped]] slot is true, throw an InvalidStateError.

  5. Let newDirection be the argument to setDirection.

  6. If newDirection is equal to transceiver's [[Direction]] slot, abort these steps.

  7. Set transceiver's [[Direction]] slot to newDirection.

  8. Update the negotiation-needed flag for connection.

stop

The stop method irreversibly stops the RTCRtpTransceiver. The sender of this transceiver will no longer send, the receiver will no longer receive. Calling stop() updates the negotiation-needed flag for the RTCRtpTransceiver's associated RTCPeerConnection .

Stopping a transceiver will cause future calls to createOffer to generate a zero port in the media description for the corresponding transceiver, as defined in [JSEP] (section 4.2.1).

When this method is invoked, to stop the RTCRtpTransceiver transceiver, the user agent MUST run the following steps:

  1. If transceiver's [[Stopped]] slot is true, abort these steps.

  2. Let connection be the RTCPeerConnection object on which the transceiver is to be stopped.

  3. If connection's [[IsClosed]] slot is true, throw an InvalidStateError.

  4. Let sender be transceiver's [[Sender]].

  5. Let receiver be transceiver's [[Receiver]].

  6. Stop sending media with sender.

  7. Send an RTCP BYE for each RTP stream that was being sent by sender, as specified in [RFC3550].

  8. Stop receiving media with receiver.

  9. receiver's [[ReceiverTrack]] is now said to be ended.

  10. Set transceiver's [[Stopped]] slot to true.

  11. Set transceiver's [[CurrentDirection]] slot to null.

  12. Update the negotiation-needed flag for connection.

When a remote description is applied with a zero port in the media description for the corresponding transceiver, as defined in [JSEP] (section 4.2.2), the user agent MUST run the above steps as if stop had been called. In addition, since the receiver's [[ReceiverTrack]] has ended, the steps described in track ended MUST be followed.

setCodecPreferences

The setCodecPreferences method overrides the default codec preferences used by the user agent. When generating a session description using either createOffer or createAnswer, the user agent MUST use the indicated codecs, in the order specified in the codecs argument, for the media section corresponding to this RTCRtpTransceiver. Note that calls to createAnswer will use only the common subset of these codecs and the codecs that appear in the offer.

This method allows applications to disable the negotiation of specific codecs. It also allows an application to cause a remote peer to prefer the codec that appears first in the list for sending.

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 default value.

The codecs sequence passed into setCodecPreferences can only contain codecs that are returned by RTCRtpSender.getCapabilities(kind) or RTCRtpReceiver.getCapabilities(kind), where kind is the kind of the RTCRtpTransceiver on which the method is called. Additionally, the RTCRtpCodecParameters dictionary members cannot be modified. If codecs does not fulfill these requirements, the user agent MUST throw an InvalidAccessError.

5.4.1 "Hold" functionality

Together, the setDirection and replaceTrack methods enable developers to implement "hold" scenarios.

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

Example 4
// Assume we have an audio transceiver and a music track named musicTrack
audio.sender.replaceTrack(musicTrack);
// Mute received audio
audio.receiver.track.enabled = false;
// Set the direction to send-only (requires negotiation)
audio.setDirection("sendonly");

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

Example 5
// Stop sending audio
audio.sender.replaceTrack(null);
// Set the direction recvonly (requires negotiation)
audio.setDirection("recvonly");
// Apply the sendonly offer, and then call createAnswer
// and send a recvonly answer
pc.setRemoteDescription(sendonlyOffer).then(doAnswer).catch(onSignalingError);

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

Example 6
//assume we have an audio transceiver and a microphone track named micTrack
audio.sender.replaceTrack(micTrack);
// Unmute received audio
audio.receiver.track.enabled = true;
// // Set the direction to sendrecv (requires negotiation)
audio.setDirection("sendrecv");

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

Example 7
// Start sending audio
audio.sender.replaceTrack(micTrack);
// Set the direction sendrecv (requires negotiation)
audio.setDirection("sendrecv");
// Apply the sendrecv offer, and then call createAnswer
// and send a sendrecv answer
pc.setRemoteDescription(sendrecvOffer).then(doAnswer).catch(onSignalingError);

5.5 RTCDtlsTransport Interface

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 RTCRtpTransceiver s if such a group has been negotiated via [BUNDLE].

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 .

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

  1. Let transport be the RTCDtlsTransport object to receive the state update.

  2. Let newState be the new state.

  3. Set transport's [[DtlsTransportState]] slot to newState.

  4. Fire a simple event named statechange at transport.

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

Attributes

transport of type RTCIceTransport, readonly

The transport 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 .

Methods

getRemoteCertificates

Returns the certificate chain in use by the remote side, with each certificate encoded in binary Distinguished Encoding Rules (DER) [X690]. getRemoteCertificates() will return an empty list prior to selection of the remote certificate, which will be completed by the time RTCDtlsTransportState transitions to "connected".

RTCDtlsTransportState Enum

enum RTCDtlsTransportState {
    "new",
    "connecting",
    "connected",
    "closed",
    "failed"
};
Enumeration description
new DTLS has not started negotiating yet.
connecting DTLS is in the process of negotiating a secure connection.
connected DTLS has completed negotiation of a secure connection.
closed The transport has been closed.
failed The transport has failed as the result of an error (such as a failure to validate the remote fingerprint).

5.5.1 RTCDtlsFingerprint Dictionary

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

dictionary RTCDtlsFingerprint {
    DOMString algorithm;
    DOMString value;
};
Dictionary RTCDtlsFingerprint Members
algorithm of type DOMString

One of the the hash function algorithms defined in the 'Hash function Textual Names' registry, initially specified in [ RFC4572] Section 8.

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.

5.6 RTCIceTransport Interface

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 RTCRtpTransceiver s if such a group has been negotiated via [BUNDLE].

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.

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

  1. Let connection be the RTCPeerConnection object associated with this ICE Agent.

  2. If connection's [[IsClosed]] slot is true, abort these steps.

  3. Let transport be the RTCIceTransport for which candidate gathering began.

  4. Set transport's [[IceGathererState]] slot to gathering .

  5. Fire a simple event named gatheringstatechange at transport.

  6. Update the ICE gathering state of connection.

When the ICE Agent indicates that it finished gathering a generation of candidates for an RTCIceTransport , the user agent MUST queue a task that runs the following steps:

  1. Let connection be the RTCPeerConnection object associated with this ICE Agent.

  2. If connection's [[IsClosed]] slot is true, abort these steps.

  3. Let transport be the RTCIceTransport for which candidate gathering finished.

  4. Create an RTCIceCandidate instance newCandidate, with sdpMid and sdpMLineIndex set to the values associated with this RTCIceTransport , with usernameFragment set to the username fragment of the generation of candidates for which gathering finished, with candidate set to an empty string, and with all other nullable members set to null.

  5. Fire an ice candidate event named icecandidate with newCandidate at connection.

  6. 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.
  7. Set transport's [[IceGathererState]] slot to complete .

  8. Fire a simple event named gatheringstatechange at transport.

  9. Update the ICE gathering state of connection.

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:

  1. Let connection be the RTCPeerConnection object associated with this ICE Agent.

  2. If connection's [[IsClosed]] slot is true, abort these steps.

  3. Let transport be the RTCIceTransport for which this candidate is being made available.

  4. If connection.pendingLocalDescription is non-null, and represents the ICE generation for which candidate was gathered, add candidate to connection.pendingLocalDescription .

  5. If connection.currentLocalDescription is non-null, and represents the ICE generation for which candidate was gathered, add candidate to connection.currentLocalDescription .

  6. Create an RTCIceCandidate instance to represent the candidate. Let newCandidate be that object.

  7. Add newCandidate to transport's set of local candidates.

  8. Fire an ice candidate event named icecandidate with newCandidate at connection.

When the ICE Agent indicates that the RTCIceTransportState for an RTCIceTransport has changed, the user agent MUST queue a task that runs the following steps:

  1. Let connection be the RTCPeerConnection object associated with this ICE Agent.

  2. If connection's [[IsClosed]] slot is true, abort these steps.

  3. Let transport be the RTCIceTransport whose state is changing.

  4. Let newState be the new indicated RTCIceTransportState .

  5. Set transport's [[IceTransportState]] slot to newState.

  6. Fire a simple event named statechange at transport.

  7. Update the ICE connection state of connection.

  8. Update the connection state of connection.

When the ICE Agent indicates that the selected candidate pair for an RTCIceTransport has changed, the user agent MUST queue a task that runs the following steps:

  1. Let connection be the RTCPeerConnection object associated with this ICE Agent.

  2. If connection's [[IsClosed]] slot is true, abort these steps.

  3. Let transport be the RTCIceTransport whose selected candidate pair is changing.

  4. Let newCandidatePair be a newly created RTCIceCandidatePair representing the indicated pair if one is selected, and null otherwise.

  5. Set transport's [[SelectedCandidatePair]] slot to newCandidatePair.

  6. Fire a simple event named selectedcandidatepairchange at transport.

An RTCIceTransport object has the following internal slots:

[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;
};

Attributes

role of type RTCIceRole, readonly

The role attribute MUST return the ICE role of the transport.

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 gathering state 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 RTCIceTransportgathering state 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.

Methods

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()

getSelectedCandidatePair

Returns the selected candidate pair on which packets are sent. This method MUST return the value of the [[SelectedCandidatePair]] slot.

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;
};

Dictionary RTCIceParameters Members

usernameFragment of type DOMString

The ICE username fragment as defined in [ICE], Section 7.1.2.3.

password of type DOMString

The ICE password as defined in [ICE], Section 7.1.2.3.

dictionary RTCIceCandidatePair {
    RTCIceCandidate local;
    RTCIceCandidate remote;
};

Dictionary RTCIceCandidatePair Members

local of type RTCIceCandidate

The local ICE candidate.

remote of type RTCIceCandidate

The remote ICE candidate.

RTCIceGathererState Enum

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

RTCIceTransportState Enum

enum RTCIceTransportState {
    "new",
    "checking",
    "connected",
    "completed",
    "failed",
    "disconnected",
    "closed"
};
Enumeration description
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 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.
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).
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".
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 have lost consent.
disconnected The ICE Agent has determined that connectivity is currently lost for this RTCIceTransport . This is more aggressive than failed, and 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 failed to find a connection (or consent checks [RFC7675] once successful, have now failed), but it is still gathering and/or waiting for additional remote candidates.
closed The RTCIceTransport has shut down and is no longer responding to STUN requests.

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 transitions might be:

ICE transport state transition diagram
Figure 2 Non-normative ICE transport state transition diagram

RTCIceRole Enum

enum RTCIceRole {
    "controlling",
    "controlled"
};
Enumeration description
controlling A controlling agent as defined by [ICE], Section 3.
controlled A controlled agent as defined by [ICE], Section 3.

RTCIceComponent Enum

enum RTCIceComponent {
    "rtp",
    "rtcp"
};
Enumeration description
rtp The ICE Transport is used for RTP (or RTCP multiplexing), as defined in [ICE], 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 [ICE], Section 4.1.1.1. This represents the component-id value 2 when encoded in candidate-attribute.

5.7 RTCTrackEvent

The track event uses the RTCTrackEvent interface.

Firing a track event named e with an RTCRtpReceiver receiver, a MediaStreamTrack track and a MediaStream[] streams, means that an event with the name e, which does not bubble (except where otherwise stated) and is not cancelable (except where otherwise stated), and which uses the RTCTrackEvent interface with the receiver attribute set to receiver, track attribute set to track, streams attribute set to streams, MUST be created and dispatched at the given target.

[Constructor(DOMString type, RTCTrackEventInit eventInitDict),
 Exposed=Window]
interface RTCTrackEvent : Event {
    readonly attribute RTCRtpReceiver           receiver;
    readonly attribute MediaStreamTrack         track;
    [SameObject]
    readonly attribute FrozenArray<MediaStream> streams;
    readonly attribute RTCRtpTransceiver        transceiver;
};

Constructors

RTCTrackEvent

Attributes

receiver of type RTCRtpReceiver, readonly

The receiver attribute represents the RTCRtpReceiver object associated with the event.

track of type MediaStreamTrack, readonly

The track attribute represents the MediaStreamTrack object that is associated with the RTCRtpReceiver identified by receiver.

streams of type FrozenArray<MediaStream>, readonly

The streams attribute returns an array of MediaStream objects representing the MediaStreams that this event's track is a part of.

transceiver of type RTCRtpTransceiver, readonly

The transceiver attribute represents the RTCRtpTransceiver object associated with the event.

dictionary RTCTrackEventInit : EventInit {
    required RTCRtpReceiver        receiver;
    required MediaStreamTrack      track;
             sequence<MediaStream> streams = [];
    required RTCRtpTransceiver     transceiver;
};

Dictionary RTCTrackEventInit Members

receiver of type RTCRtpReceiver, required

The receiver attribute represents the RTCRtpReceiver object associated with the event.

track of type MediaStreamTrack, required

The track attribute represents the MediaStreamTrack object that is associated with the RTCRtpReceiver identified by receiver.

streams of type sequence<MediaStream>, defaulting to []

The streams attribute returns an array of MediaStream objects representing the MediaStreams that this event's track is a part of.

transceiver of type RTCRtpTransceiver, required

The transceiver attribute represents the RTCRtpTransceiver object associated with the event.

6. Peer-to-peer Data API

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 WebSockets [WEBSOCKETS-API].

6.1 RTCPeerConnection Interface Extensions

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;
};

Attributes

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 .

Methods

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.

  1. Let connection be the RTCPeerConnection object on which the method is invoked.

  2. If connection's [[IsClosed]] slot is true, throw an InvalidStateError.

  3. Let channel be a newly created RTCDataChannel object.

  4. Let channel have a [[DataChannelLabel]] internal slot initialized to the value of the first argument.

  5. If [[DataChannelLabel]] is longer than 65535 bytes, throw a TypeError.
  6. Let options be the second argument.

  7. Let channel have a [[MaxPacketLifeTime]] internal slot initialized to option's maxPacketLifeTime member, if present, otherwise null.

  8. Let channel have a [[ReadyState]] internal slot initialized to "connecting".

  9. Let channel have a [[MaxRetransmits]] internal slot initialized to option's maxRetransmits member, if present, otherwise null.

  10. Let channel have an [[Ordered]] internal slot initialized to option's ordered member.

  11. Let channel have a [[DataChannelProtocol]] internal slot initialized to option's protocol member.

  12. If [[DataChannelProtocol]] is longer than 65535 bytes long, throw a TypeError.
  13. Let channel have a [[Negotiated]] internal slot initialized to option's negotiated member.

  14. Let channel have an [[DataChannelId]] internal slot initialized to option's id member, 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 [ RTCWEB-DATA-PROTOCOL].
  15. If [[Negotiated]] is true and [[DataChannelId]] is null, throw a TypeError.

  16. Let channel have an [[DataChannelPriority]] internal slot initialized to option's priority member.

  17. If both [[MaxPacketLifeTime]] and [[MaxRetransmits]] attributes are set (not null), throw a TypeError.

  18. 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.

  19. 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.

  20. 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 [RTCWEB-DATA-PROTOCOL], 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 once the DTLS role is determined during the process of setting an RTCSessionDescription.
  21. If channel is the first RTCDataChannel created on connection, update the negotiation-needed flag for connection.

  22. Return channel and continue the following steps in parallel.

  23. Create channel's associated underlying data transport and configure it according to the relevant properties of channel.

6.1.1 RTCSctpTransport Interface

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

[Exposed=Window]
interface RTCSctpTransport {
    readonly attribute RTCDtlsTransport      transport;
    readonly attribute RTCSctpTransportState state;
    readonly attribute unsigned long         maxMessageSize;
             attribute EventHandler          onstatechange;
};
Attributes
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, it returns the value of the [[SctpTransportState]] internal slot.

maxMessageSize of type unsigned long, readonly

The maximum size of data that can be passed to RTCDataChannel 's send() method.

onstatechange of type EventHandler

The event type of this event handler is statechange .

6.1.2 RTCSctpTransportState Enum

RTCSctpTransportState indicates the state of the SCTP transport.

enum RTCSctpTransportState {
    "new",
    "connecting",
    "connected",
    "closed"
};
Enumeration description
new

The RTCSctpTransport has not started negotiating yet.

connecting

The RTCSctpTransport is in the process of negotiating an association. This is the initial state when an RTCSctpTransport is created by applying an Answer.

connected

The RTCSctpTransport has completed negotiation of an association.

closed

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) or via receipt of a SHUTDOWN or ABORT chunk.

6.2 RTCDataChannel

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 [RTCWEB-DATA] and [ RTCWEB-DATA-PROTOCOL].

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 id s.

Each RTCDataChannel has an associated underlying data transport that is used to transport actual data to the other peer. 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 [RTCWEB-DATA].

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.

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

  1. If the associated RTCPeerConnection object's [[IsClosed]] slot is true, abort these steps.

  2. Let channel be the RTCDataChannel object to be announced.

  3. Set channel's [[ReadyState]] slot to open.

  4. Fire a simple 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:

  1. If the associated RTCPeerConnection object's [[IsClosed]] slot is true, abort these steps.

  2. Let channel be a newly created RTCDataChannel object.

  3. 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 [RTCWEB-DATA-PROTOCOL].

  4. Initialize channel's [[DataChannelLabel]], [[Ordered]], [[MaxPacketLifeTime]], [[MaxRetransmits]], [[DataChannelProtocol]], and [[DataChannelId]] internal slots to the corresponding values in configuration.

  5. Initialize channel's [[Negotiated]] internal slot to false.

  6. Initialize channel's [[DataChannelPriority]] internal slot based on the integer priority value in configuration, according to the following mapping:

    configuration priority value RTCPriorityType value
    0 to 128 very-low
    129 to 256 low
    257 to 512 medium
    513 and greater high
  7. Set channel's [[ReadyState]] slot to connecting.

  8. Fire a datachannel event named datachannel with channel at the RTCPeerConnection object.

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, unless the procedure was initiated by the close method, queue a task that sets the object's [[ReadyState]] slot to closing. This will eventually render the data transport closed.

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

  1. Let channel be the RTCDataChannel object whose transport was closed.

  2. Set channel's [[ReadyState]] slot to closed.

  3. If the transport was closed with an error, fire an RTCError event at channel with errorDetail set to "sctp-failure".

  4. Fire a simple event named close at channel.

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 [ RTCWEB-DATA] 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:

  1. Let channel be the RTCDataChannel object for which the user agent could not create an underlying data transport.

  2. Set channel's [[ReadyState]] slot to closed.

  3. Fire an RTCError event at channel with errorDetail set to "data-channel-failure".

  4. Fire a simple event named close at channel.

[Exposed=Window]
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 RTCPriorityType     priority;
    readonly attribute RTCDataChannelState readyState;
    readonly attribute unsigned long       bufferedAmount;
             attribute unsigned long       bufferedAmountLowThreshold;
             attribute EventHandler        onopen;
             attribute EventHandler        onbufferedamountlow;
             attribute EventHandler        onerror;
             attribute EventHandler        onclose;
    void close();
             attribute EventHandler        onmessage;
             attribute DOMString           binaryType;
    void send(USVString data);
    void send(Blob data);
    void send(ArrayBuffer data);
    void send(ArrayBufferView data);
};

Attributes

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 other 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 initally 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 [ RTCWEB-DATA-PROTOCOL]. 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.

priority of type RTCPriorityType, readonly

The priority attribute returns the priority for this RTCDataChannel . The priority is assigned by the user agent at channel creation time. On getting, the attribute MUST return the value of the [[DataChannelPriority]] 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 return the number of bytes of application data (UTF-8 text and binary data) that have been queued using send() but that, as of the last time the event loop started executing a task, had not yet been transmitted to the network. (This thus includes any text sent during the execution of the current task, regardless of whether the user agent is able to transmit text asynchronously with script execution.) This does not include framing overhead incurred by the protocol, or buffering done by the operating system or network hardware. If the channel is closed, this attribute's value will only increase with each call to the send() method (the attribute does not reset to zero once the channel closes).

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.

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 DOMString

The binaryType attribute MUST, on getting, return the value to which it was last set. On setting, if the new value is either the string "blob" or the string "arraybuffer", then set the IDL attribute to this new value. Otherwise, throw a SyntaxError. When an RTCDataChannel object is created, the binaryType attribute MUST be initialized to the string "blob".

This attribute controls how binary data is exposed to scripts. See the [WEBSOCKETS-API] for more information.

Methods

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:

  1. Let channel be the RTCDataChannel object which is about to be closed.

  2. If channel's [[ReadyState]] slot is closing or closed, then abort these steps.

  3. Set channel's [[ReadyState]] slot to closing.

  4. 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.

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

Dictionary RTCDataChannelInit Members

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.

id of type unsigned short

Overrides the default selection of ID for this channel.

priority of type RTCPriorityType, defaulting to low

Priority of this channel.

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:

  1. Let channel be the RTCDataChannel object on which data is to be sent.

  2. If channel's [[ReadyState]] slot is not open, throw an InvalidStateError.

  3. Execute the sub step that corresponds to the type of the methods argument:

    • string object:

      Let data be the object and increase the bufferedAmount attribute by the number of bytes needed to express data as UTF-8.

    • Blob object:

      Let data be the raw data represented by the Blob object and increase the bufferedAmount attribute by the size of data, in bytes.

    • ArrayBuffer object:

      Let data be the data stored in the buffer described by the ArrayBuffer object and increase the bufferedAmount attribute by the length of the ArrayBuffer in bytes.

    • ArrayBufferView object:

      Let data be the data stored in the section of the buffer described by the ArrayBuffer object that the ArrayBufferView object references and increase the bufferedAmount attribute by the length of the ArrayBufferView in bytes.

  4. If the size of data exceeds the value of maxMessageSize on channel's associated RTCSctpTransport, throw a TypeError.

  5. 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 .
enum RTCDataChannelState {
    "connecting",
    "open",
    "closing",
    "closed"
};
RTCDataChannelState Enumeration description
connecting

The user agent is attempting to establish the underlying data transport. This is the initial state of an RTCDataChannel object created with createDataChannel .

open

The underlying data transport is established and communication is possible. This is the initial state of an RTCDataChannel object dispatched as a part of an RTCDataChannelEvent .

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.

6.3 RTCDataChannelEvent

The datachannel event uses the RTCDataChannelEvent interface.

Firing a datachannel event named e with an RTCDataChannel channel means that an event with the name e, which does not bubble (except where otherwise stated) and is not cancelable (except where otherwise stated), and which uses the RTCDataChannelEvent interface with the channel attribute set to channel, MUST be created and dispatched at the given target.

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

Constructors

RTCDataChannelEvent

Attributes

channel of type RTCDataChannel, readonly

The channel attribute represents the RTCDataChannel object associated with the event.

dictionary RTCDataChannelEventInit : EventInit {
    required RTCDataChannel channel;
};

Dictionary RTCDataChannelEventInit Members

channel of type RTCDataChannel, required

The RTCDataChannel object to be announced by the event.

6.4 Garbage Collection

An RTCDataChannel object MUST not be garbage collected if its

7. Peer-to-peer DTMF

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 [RTCWEB-AUDIO].

7.1 RTCRtpSender Interface Extensions

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

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

Attributes

dtmf of type RTCDTMFSender, readonly, nullable

The dtmf attribute returns an RTCDTMFSender which can be used to send DTMF, or null if unset. The attribute is set when the kind of an RTCRtpSender 's [[SenderTrack]] is "audio".

7.2 RTCDTMFSender

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

  1. Let dtmf be a newly created RTCDTMFSender object.

  2. Let dtmf have a [[CanInsertDtmf]] internal slot, initialized to false.

  3. Let dtmf have a [[Duration]] internal slot.

  4. Let dtmf have a [[InterToneGap]] internal slot.

  5. Let dtmf have a [[ToneBuffer]] internal slot.

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

Attributes

ontonechange of type EventHandler

The event type of this event handler is tonechange .

canInsertDTMF of type boolean, readonly

Whether the RTCDTMFSender is capable of sending DTMF.

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 .

Methods

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:

  1. Let sender be the RTCRtpSender used to send DTMF.
  2. Let transceiver be the RTCRtpTransceiver object associated with sender.

  3. If transceiver's [[Stopped]] slot is true, throw an InvalidStateError.
  4. If transceiver's [[CurrentDirection]] slot is recvonly or inactive, throw an InvalidStateError.
  5. Let dtmf be the RTCDTMFSender associated with sender.
  6. If dtmf's [[CanInsertDTMF]] internal slot is false, throw an InvalidStateError.
  7. Let tones be the method's first argument.
  8. If tones contains any unrecognized characters, throw an InvalidCharacterError.
  9. Set the object's [[ToneBuffer]] slot to tones.
  10. Set dtmf's [[Duration]] slot to the value of the duration parameter.
  11. Set dtmf's [[InterToneGap]] slot to the value of the interToneGap parameter.
  12. If the value of the duration parameter is less than 40 ms, set dtmf's [[Duration]] slot to 40 ms.
  13. If the value of the duration parameter is greater than 6000 ms, set dtmf's [[Duration]] slot to 6000 ms.
  14. If the value of the interToneGap parameter is less than 30 ms, set dtmf's [[InterToneGap]] slot to 30 ms.
  15. If the value of the interToneGap parameter is greater than 6000 ms, set dtmf's [[InterToneGap]] slot to 6000 ms.
  16. If [[ToneBuffer]] slot is an empty string, abort these steps.
  17. If a Playout task is scheduled to be run, abort these steps; otherwise queue a task that runs the following steps (Playout task):
    1. If transceiver's [[Stopped]] slot is true, abort these steps.
    2. If transceiver's [[CurrentDirection]] slot is recvonly or inactive, abort these steps.
    3. If the [[ToneBuffer]] slot contains the empty string, fire an event named tonechange with 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 steps labelled Playout task.
    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 steps labelled Playout task.
    7. Fire a tonechange event named tonechange with a string consisting of 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.

7.3 RTCDTMFToneChangeEvent

The tonechange event uses the RTCDTMFToneChangeEvent interface.

Firing a tonechange event named e with a DOMString tone means that an event with the name e, which does not bubble (except where otherwise stated) and is not cancelable (except where otherwise stated), and which uses the RTCDTMFToneChangeEvent interface with the tone attribute set to tone, MUST be created and dispatched at the given target.

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

Constructors

RTCDTMFToneChangeEvent

Attributes

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 {
    required DOMString tone;
};

Dictionary RTCDTMFToneChangeEventInit Members

tone of type DOMString

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.

8. Statistics Model

8.1 Introduction

The basic statistics model is that the browser maintains a set of statistics 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 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.

8.2 RTCPeerConnection Interface Extensions

The Statistics API extends the RTCPeerConnection interface as described below.

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

Methods

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:

  1. Let selectorArg be the method's first argument.

  2. Let connection be the RTCPeerConnection object on which the method was invoked.

  3. If selectorArg is null, let selector be null.

  4. If selectorArg is a MediaStreamTrack let selector be an RTCRtpSender or RTCRtpReceiver on connection which track member 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.

  5. Let p be a new promise.

  6. Run the following steps in parallel:

    1. Gather the stats indicated by selector according to the stats selection algorithm.

    2. Resolve p with the resulting RTCStatsReport object, containing the gathered stats.

  7. Return p.

8.3 RTCStatsReport Object

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>;
};

This interface has "entries", "forEach", "get", "has", "keys", "values", @@iterator methods and a "size" getter brought by readonly maplike.

Use these to retrieve the various dictionaries descended from RTCStats that this stats report is composed of. The set of supported property names [WEBIDL-1] is defined as the ids of all the RTCStats -derived dictionaries that have been generated for this stats report.

8.4 RTCStats Dictionary

An RTCStats dictionary represents the stats gathered by inspecting a specific object relevant to a selector. 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;
};

Dictionary RTCStats Members

timestamp of type DOMHighResTimeStamp

The timestamp, of type DOMHighResTimeStamp [HIGHRES-TIME], associated with this object. The time is relative to the UNIX epoch (Jan 1, 1970, UTC). 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. User agents are free to pick any format for the id as long as it meets the requirements above.

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

8.5 The stats selection algorithm

The stats selection algorithm is as follows:

  1. Let result be an empty RTCStatsReport.
  2. If selector is null, gather stats for the whole connection, add them to result, return result, and abort these steps.
  3. 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.
  4. 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.
  5. Return result.

8.6 Mandatory To Implement Stats

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 PeerConnection, with the attributes that are listed when they are valid for that object:

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

8.7 GetStats Example

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
var baselineReport, currentReport;
var selector = pc.getSenders()[0].track;

pc.getStats(selector).then(function (report) {
    baselineReport = report;
})
.then(function() {
    return new Promise(function(resolve) {
        setTimeout(resolve, aBit); // ... wait a bit
    });
})
.then(function() {
    return pc.getStats(selector);
})
.then(function (report) {
    currentReport = report;
    processStats();
})
.catch(function (error) {
  log(error.toString());
});

function processStats() {
    // compare the elements from the current report with the baseline
    currentReport.forEach (now => {
        if (now.type != "outbound-rtp")
            return;

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

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

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

            // if fractionLost is > 0.3, we have probably found the culprit
            var fractionLost = (packetsSent - packetsReceived) / packetsSent;
        }
    }
}

9. Identity

9.1 Identity Provider Interaction

WebRTC offers and answers (and hence the channels established by RTCPeerConnection objects) can be authenticated by using a web-based Identity Provider (IdP). The idea is that the entity sending an offer or answer acts as the Authenticating Party (AP) and obtains an identity assertion from the IdP which it attaches to the session description. The consumer of the session description (i.e., the RTCPeerConnection on which setRemoteDescription is called) acts as the Relying Party (RP) and verifies the assertion.

The interaction with the IdP is designed to decouple the browser from any particular identity provider; the browser need only know how to load the IdP's JavaScript, the location of which is determined by the IdP's identity, and the generic interface to generating and validating assertions. The IdP provides whatever logic is necessary to bridge the generic protocol to the IdP's specific requirements. Thus, a single browser can support any number of identity protocols, including being forward compatible with IdPs which did not exist at the time the browser was written.

9.1.1 Identity Provider Selection

An IdP is used to generate an identity assertion as follows:

  1. If the setIdentityProvider() method has been called, the IdP provided shall be used.
  2. If the setIdentityProvider() method has not been called, then the user agent MAY use an IdP configured into the browser.

In order to verify assertions, the IdP domain name and protocol are taken from the domain and protocol fields of the identity assertion.

9.1.2 Instantiating an IdP Proxy

In order to communicate with the IdP, the user agent loads the IdP JavaScript from the IdP. The URI for the IdP script is a well-known URI formed from the domain and protocol fields, as specified in [RTCWEB-SECURITY-ARCH].

The IdP MAY generate an HTTP redirect to another "https" origin, the browser MUST treat a redirect to any other scheme as a fatal error.

The user agent instantiates an isolated interpreted context, a JavaScript realm that operates in the origin of the loaded JavaScript. Note that a redirect will change the origin of the loaded script.

The realm is populated with a global that implements both the RTCIdentityProviderGlobalScope and WorkerGlobalScope [WEBWORKERS] interfaces.

The user agent provides an instance of RTCIdentityProviderRegistrar named rtcIdentityProvider in the global scope of the realm. This object is used by the IdP to interact with the user agent.

[Global,
 Exposed=RTCIdentityProviderGlobalScope]
interface RTCIdentityProviderGlobalScope : WorkerGlobalScope {
    readonly attribute RTCIdentityProviderRegistrar rtcIdentityP