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.

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

The specification is feature complete and is expected to be stable with no further substantive change. Since the previous Candidate Recommendation, the following substantive changes have been brought to the specification:

Its associated 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 at risk:

Introduction

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 WebRTC Working Group. An overview of the system can be found in [[RTCWEB-OVERVIEW]] and [[RTCWEB-SECURITY]].

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

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

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

Terminology

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

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

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

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

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

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

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

The term {{Blob}} is defined in [[!FILEAPI]].

The term media description is defined in [[!RFC4566]].

The term media transport is defined in [[!RFC7656]].

The term generation is defined in [[!TRICKLE-ICE]] Section 2.

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

When referring to exceptions, the terms [= exception/throw =] and [= exception/create =] are defined in [[!WEBIDL]].

The callback {{VoidFunction}} is defined in [[!WEBIDL]].

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

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

The terms bundle, bundle-only and bundle-policy are defined in [[!JSEP]].

The AlgorithmIdentifier is defined in [[!WebCryptoAPI]].

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

Peer-to-peer connections

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 [[?xhr]] or Web Sockets.

Configuration

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;
  RTCBundlePolicy bundlePolicy;
  RTCRtcpMuxPolicy rtcpMuxPolicy;
  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}}.

Indicates which candidates the [= ICE Agent =] is allowed to use.

bundlePolicy of type {{RTCBundlePolicy}}.

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

rtcpMuxPolicy of type {{RTCRtcpMuxPolicy}}.

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

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 {{RTCPeerConnection/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]].

RTCIceCredentialType Enum

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

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 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 {{RTCIceCredentialType/"password"}}, then this attribute specifies the username to use with that TURN server.

credential of type {{DOMString}}

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

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

To support additional values of {{credentialType}}, {{credential}} may evolve in future as a union.

credentialType of type {{RTCIceCredentialType}}, defaulting to {{RTCIceCredentialType/"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:

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

RTCIceTransportPolicy Enum

As described in [[!JSEP]], if the {{RTCConfiguration/iceTransportPolicy}} member of the {{RTCConfiguration}} is specified, it defines the ICE candidate policy [[!JSEP]] 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.

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.

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}}.{{RTCIceCandidate/address}}.

RTCBundlePolicy Enum

As described in [[!JSEP]], 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.

RTCRtcpMuxPolicy Enum

As described in [[!JSEP]], the {{RTCRtcpMuxPolicy}} affects what ICE candidates are gathered to support non-multiplexed RTCP. The only value defined in this spec is {{RTCRtcpMuxPolicy/"require"}}.

enum RTCRtcpMuxPolicy {
  "require"
};
Enumeration description (non-normative)
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.

Offer/Answer Options

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

dictionary RTCOfferAnswerOptions {};

Dictionary RTCOfferAnswerOptions Members

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, or the relevant {{RTCPeerConnection}} object's [[\LocalIceCredentialsToReplace]] slot is not empty, then the generated description will have ICE credentials that are different from the current credentials (as visible in the {{RTCPeerConnection/currentLocalDescription}} 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 relevant {{RTCPeerConnection}} object's [[\LocalIceCredentialsToReplace]] slot is empty, and the {{RTCPeerConnection/currentLocalDescription}} attribute has valid ICE credentials, then the generated description will have the same ICE credentials as the current value from the {{RTCPeerConnection/currentLocalDescription}} attribute.

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

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

dictionary RTCAnswerOptions : RTCOfferAnswerOptions {};

State Definitions

RTCSignalingState Enum

enum RTCSignalingState {
  "stable",
  "have-local-offer",
  "have-remote-offer",
  "have-local-pranswer",
  "have-remote-pranswer",
  "closed"
};
Enumeration description
stable There is no offer/answer exchange in progress. This is also the initial state, in which case the local and remote descriptions are empty.
have-local-offer A local description, of type {{RTCSdpType/"offer"}}, has been successfully applied.
have-remote-offer A remote description, of type {{RTCSdpType/"offer"}}, has been successfully applied.
have-local-pranswer A remote description of type {{RTCSdpType/"offer"}} has been successfully applied and a local description of type {{RTCSdpType/"pranswer"}} has been successfully applied.
have-remote-pranswer A local description of type {{RTCSdpType/"offer"}} has been successfully applied and a remote description of type {{RTCSdpType/"pranswer"}} has been successfully applied.
closed The {{RTCPeerConnection}} has been closed; its [[\IsClosed]] slot is true.
signalling state transition diagram
Non-normative signalling state transitions diagram. Method calls abbreviated.

An example set of transitions might be:

Caller transition:
  • new RTCPeerConnection(): {{RTCSignalingState/"stable"}}
  • setLocalDescription(offer): {{RTCSignalingState/"have-local-offer"}}
  • setRemoteDescription(pranswer): {{RTCSignalingState/"have-remote-pranswer"}}
  • setRemoteDescription(answer): {{RTCSignalingState/"stable"}}
Callee transition:
  • new RTCPeerConnection(): {{RTCSignalingState/"stable"}}
  • setRemoteDescription(offer): {{RTCSignalingState/"have-remote-offer"}}
  • setLocalDescription(pranswer): {{RTCSignalingState/"have-local-pranswer"}}
  • setLocalDescription(answer): {{RTCSignalingState/"stable"}}

RTCIceGatheringState Enum

enum RTCIceGatheringState {
  "new",
  "gathering",
  "complete"
};
Enumeration description
new Any of the {{RTCIceTransport}}s are in the {{RTCIceGathererState/"new"}} gathering state and none of the transports are in the {{RTCIceGathererState/"gathering"}} state, or there are no transports.
gathering Any of the {{RTCIceTransport}}s are in the {{RTCIceGathererState/"gathering"}} state.
complete At least one {{RTCIceTransport}} exists, and all {{RTCIceTransport}}s are in the {{RTCIceGathererState/"complete"}} gathering state.

RTCPeerConnectionState Enum

enum RTCPeerConnectionState {
  "closed",
  "failed",
  "disconnected",
  "new",
  "connecting",
  "connected"
};
Enumeration description
closed The {{RTCPeerConnection}} object's [[\IsClosed]] slot is true.
failed The previous state doesn't apply and any {{RTCIceTransport}}s are in the {{RTCIceTransportState/"failed"}} state or any {{RTCDtlsTransport}}s are in the {{RTCDtlsTransportState/"failed"}} state.
disconnected None of the previous states apply and any {{RTCIceTransport}}s are in the {{RTCIceTransportState/"disconnected"}} state.
new None of the previous states apply and all {{RTCIceTransport}}s are in the {{RTCIceTransportState/"new"}} or {{RTCIceTransportState/"closed"}} state, and all {{RTCDtlsTransport}}s are in the {{RTCDtlsTransportState/"new"}} or {{RTCDtlsTransportState/"closed"}} state, or there are no transports.
connecting None of the previous states apply and all {{RTCIceTransport}}s are in the {{RTCIceTransportState/"new"}} or {{RTCIceTransportState/"checking"}} state, and all {{RTCDtlsTransport}}s are in the {{RTCDtlsTransportState/"new"}} or {{RTCDtlsTransportState/"connecting"}} state.
connected None of the previous states apply and all {{RTCIceTransport}}s are in the {{RTCIceTransportState/"connected"}}, {{RTCIceTransportState/"completed"}} or {{RTCIceTransportState/"closed"}} state, and all {{RTCDtlsTransport}}s are in the {{RTCDtlsTransportState/"connected"}} or {{RTCDtlsTransportState/"closed"}} state.

RTCIceConnectionState Enum

enum RTCIceConnectionState {
  "closed",
  "failed",
  "disconnected",
  "new",
  "checking",
  "completed",
  "connected"
};
Enumeration description
closed The {{RTCPeerConnection}} object's [[\IsClosed]] slot is true.
failed The previous state doesn't apply and any {{RTCIceTransport}}s are in the {{RTCIceTransportState/"failed"}} state.
disconnected None of the previous states apply and any {{RTCIceTransport}}s are in the {{RTCIceTransportState/"disconnected"}} state.
new None of the previous states apply and all {{RTCIceTransport}}s are in the {{RTCIceTransportState/"new"}} or {{RTCIceTransportState/"closed"}} state, or there are no transports.
checking None of the previous states apply and any {{RTCIceTransport}}s are in the {{RTCIceTransportState/"new"}} or {{RTCIceTransportState/"checking"}} state.
completed None of the previous states apply and all {{RTCIceTransport}}s are in the {{RTCIceTransportState/"completed"}} or {{RTCIceTransportState/"closed"}} state.
connected None of the previous states apply and all {{RTCIceTransport}}s are in the {{RTCIceTransportState/"connected"}}, {{RTCIceTransportState/"completed"}} or {{RTCIceTransportState/"closed"}} state.

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.

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.

Operation

Calling new {{RTCPeerConnection}}(configuration) creates an {{RTCPeerConnection}} object.

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

An {{RTCPeerConnection}} object has a 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 .

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

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

Constructor

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

  1. If any of the steps enumerated below fails for a reason not specified here, [= exception/throw =] an {{UnknownError}} with the {{DOMException/message}} attribute set to an appropriate description.

  2. Let connection be a newly created {{RTCPeerConnection}} object.

  3. Let connection have a [[\DocumentOrigin]] internal slot, initialized to the current settings object's origin.

  4. Let configuration be the method's first argument.
  5. If the {{RTCConfiguration/certificates}} value in configuration is non-empty, run the following steps for each certificate in certificates:

    1. If the value of certificate.{{RTCCertificate/expires}} is less than the current time, [= exception/throw =] an {{InvalidAccessError}}.

    2. If certificate.[[\Origin]] is not same origin with connection.[[\DocumentOrigin]], [= exception/throw =] an {{InvalidAccessError}}.

    3. Store certificate.

  6. Else, generate one or more new {{RTCCertificate}} instances with this {{RTCPeerConnection}} instance and store them. This MAY happen asynchronously and the value of {{RTCConfiguration/certificates}} remains undefined for the subsequent steps. As noted in Section 4.3.2.3 of [[RTCWEB-SECURITY]], WebRTC utilizes self-signed rather than Public Key Infrastructure (PKI) certificates, so that the expiration check is to ensure that keys are not used indefinitely and additional certificate checks are unnecessary.

  7. Initialize connection's [= ICE Agent =].

  8. If the value of configuration.{{RTCConfiguration/iceTransportPolicy}} is undefined, set it to {{RTCIceTransportPolicy/"all"}}.

  9. If the value of configuration.{{RTCConfiguration/bundlePolicy}} is undefined, set it to {{RTCBundlePolicy/"balanced"}}.

  10. If the value of configuration.{{RTCConfiguration/rtcpMuxPolicy}} is undefined, set it to {{RTCRtcpMuxPolicy/"require"}}.

  11. Let connection have a [[\Configuration]] internal slot. [= Set the configuration =] specified by configuration.

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

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

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

  15. Let connection have an [[\Operations]] internal slot, representing an [= operations chain =], initialized to an empty list.

  16. Let connection have an [[\LastCreatedOffer]] internal slot, initialized to "".

  17. Let connection have an [[\LastCreatedAnswer]] internal slot, initialized to "".

  18. Let connection have an [[\EarlyCandidates]] internal slot, initialized to an empty list.

  19. Set connection's [= signaling state =] to {{RTCSignalingState/"stable"}}.

  20. Set connection's [= ICE connection state =] to {{RTCIceConnectionState/"new"}}.

  21. Set connection's [= ICE gathering state =] to {{RTCIceGatheringState/"new"}}.

  22. Set connection's [= connection state =] to {{RTCPeerConnectionState/"new"}}.

  23. Let connection have a [[\PendingLocalDescription]] internal slot, initialized to null.

  24. Let connection have a [[\CurrentLocalDescription]] internal slot, initialized to null.

  25. Let connection have a [[\PendingRemoteDescription]] internal slot, initialized to null.

  26. Let connection have a [[\CurrentRemoteDescription]] internal slot, initialized to null.

  27. Let connection have a [[\LocalIceCredentialsToReplace]] internal slot, initialized to an empty set.

  28. Return connection.

Chain an asynchronous operation

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

To chain an operation to an {{RTCPeerConnection}} object's [= operations chain =], run the following steps:

  1. Let connection be the {{RTCPeerConnection}} object.

  2. If connection.[[\IsClosed]] is true, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.

  3. Let operation be the operation to be chained.

  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.[[\IsClosed]] is true, abort these steps.

    2. If the promise returned by operation was [= fulfilled =] with a value, [= fulfill =] p with that value.

    3. If the promise returned by operation was [= rejected =] with a value, [= reject =] p with that value.

    4. Upon [= fulfillment =] or [= rejection =] of p, execute the following steps:

      1. If connection.[[\IsClosed]] is true, abort these steps.

      2. Remove the first element of [[\Operations]].

      3. If [[\Operations]] is non-empty, execute the operation represented by the first element of [[\Operations]].

  8. Return p.

Update the connection state

An {{RTCPeerConnection}} object has an aggregated [= connection state =]. Whenever the state of an {{RTCDtlsTransport}} 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 an event =] named {{connectionstatechange}} at connection.

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.[[\IsClosed]] 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 an event =] named {{icegatheringstatechange}} at connection.

  6. If newState is {{RTCIceGatheringState/"complete"}}, [= fire an event =] named {{icecandidate}} using the {{RTCPeerConnectionIceEvent}} interface with the candidate attribute set to null at connection.

    The null candidate event is fired to ensure legacy compatibility. New code should monitor the gathering state of {{RTCIceTransport}} and/or {{RTCPeerConnection}}.

Set the RTCSessionDescription

To set a local RTCSessionDescription description on an {{RTCPeerConnection}} object connection, run the [= set an RTCSessionDescription =] algorithm with remote set to false.

To set a remote RTCSessionDescription description on an {{RTCPeerConnection}} object connection, run the [= set an RTCSessionDescription =] algorithm with remote set to true.

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

  1. Let p be a new promise.

  2. If description.{{RTCSessionDescription/type}} is {{RTCSdpType/"rollback"}} and connection's [= signaling state =] is either {{RTCSignalingState/"stable"}}, {{RTCSignalingState/"have-local-pranswer"}}, or {{RTCSignalingState/"have-remote-pranswer"}}, then [= reject =] p with a newly [= exception/create | created =] {{InvalidStateError}} and abort these steps.

  3. In parallel, start the process to apply description as described in [[!JSEP]], with the additional restriction that if applying description leads to modifying a transceiver transceiver, and transceiver.[[\Sender]].[[\SendEncodings]] is non-empty, and not equal to the encodings that would result from processing description, the process of applying description fails. This specification does not allow remotely initiated RID renegotiation.

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

      1. If connection.[[\IsClosed]] is true, then abort these steps.

      2. If description.{{RTCSessionDescription/type}} is invalid for the current [= signaling state =] of connection as described in [[!JSEP]], then [= reject =] p with a newly [= exception/create | created =] {{InvalidStateError}} and abort these steps.

      3. If the content of description is not valid SDP syntax, then [= reject =] p with an {{RTCError}} (with {{RTCError/errorDetail}} set to {{RTCErrorDetailType/"sdp-syntax-error"}} and the {{RTCError/sdpLineNumber}} attribute set to the line number in the SDP where the syntax error was detected) and abort these steps.

      4. If remote is true, the connection's {{RTCRtcpMuxPolicy}} is {{RTCRtcpMuxPolicy/require}} and the description does not use RTCP mux, then [= reject =] p with a newly [= exception/create | created =] {{InvalidAccessError}} and abort these steps.

      5. If the description attempted to renegotiate RIDs, as described above, then [= reject =] p with a newly [= exception/create | created =] {{InvalidAccessError}} and abort these steps.

      6. If the content of description is invalid, then [= reject =] p with a newly [= exception/create | created =] {{InvalidAccessError}} and abort these steps.

      7. For all other errors, [= reject =] p with a newly [= exception/create | created =] {{OperationError}}.

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

      1. If connection.[[\IsClosed]] is true, then abort these steps.

      2. If description is of type {{RTCSdpType/"offer"}} and the [= signaling state =] of connection is {{RTCSignalingState/"stable"}} then for each transceiver in connection's [= set of transceivers =], run the following steps:

        1. Set transceiver.[[\Sender]].[[\LastStableStateSenderTransport]] to transceiver.[[\Sender]].[[\SenderTransport]].

        2. Set transceiver.[[\Receiver]].[[\LastStableStateReceiverTransport]] to transceiver.[[\Receiver]].[[\ReceiverTransport]].

        3. Set transceiver.[[\Receiver]].[[\LastStableStateAssociatedRemoteMediaStreams]] to transceiver.[[\Receiver]].[[\AssociatedRemoteMediaStreams]].

        4. Set transceiver.[[\Receiver]].[[\LastStableStateReceiveCodecs]] to transceiver.[[\Receiver]].[[\ReceiveCodecs]].

      3. If remote is false, then run one of the following steps:

        1. If description is of type {{RTCSdpType/"offer"}}, set connection.[[\PendingLocalDescription]] to a new {{RTCSessionDescription}} object constructed from description, set connection's [= signaling state =] to {{RTCSignalingState/"have-local-offer"}}, and [= release early candidates =].

        2. If description is of type {{RTCSdpType/"answer"}}, then this completes an offer answer negotiation. Set connection.[[\CurrentLocalDescription]] to a new {{RTCSessionDescription}} object constructed from description, and set connection.[[\CurrentRemoteDescription]] to connection.[[\PendingRemoteDescription]]. Set both connection.[[\PendingRemoteDescription]] and connection.[[\PendingLocalDescription]] to null. Set both connection.[[\LastCreatedOffer]] and connection.[[\LastCreatedAnswer]] to "", set connection's [= signaling state =] to {{RTCSignalingState/"stable"}}, and [= release early candidates =]. Finally, if none of the ICE credentials in connection.[[\LocalIceCredentialsToReplace]] are present in description, then set connection.[[\LocalIceCredentialsToReplace]] to an empty set.

        3. If description is of type {{RTCSdpType/"pranswer"}}, then set connection.[[\PendingLocalDescription]] to a new {{RTCSessionDescription}} object constructed from description, set connection's [= signaling state =] to {{RTCSignalingState/"have-local-pranswer"}}, and [= release early candidates =].

      4. Otherwise, (if remote is true) run one of the following steps:

        1. If description is of type {{RTCSdpType/"offer"}}, set connection.[[\PendingRemoteDescription]] attribute to a new {{RTCSessionDescription}} object constructed from description, and set connection's [= signaling state =] to {{RTCSignalingState/"have-remote-offer"}}.

        2. If description is of type {{RTCSdpType/"answer"}}, then this completes an offer answer negotiation. Set connection.[[\CurrentRemoteDescription]] to a new {{RTCSessionDescription}} object constructed from description, and set connection.[[\CurrentLocalDescription]] to connection.[[\PendingLocalDescription]]. Set both connection.[[\PendingRemoteDescription]] and connection.[[\PendingLocalDescription]] to null. Set both connection.[[\LastCreatedOffer]] and connection.[[\LastCreatedAnswer]] to "", and set connection's [= signaling state =] to {{RTCSignalingState/"stable"}}. Finally, if none of the ICE credentials in connection.[[\LocalIceCredentialsToReplace]] are present in the newly set connection.[[\CurrentLocalDescription]], then set connection.[[\LocalIceCredentialsToReplace]] to an empty set.

        3. If description is of type {{RTCSdpType/"pranswer"}}, then set connection.[[\PendingRemoteDescription]] to a new {{RTCSessionDescription}} object constructed from description and set connection's [= signaling state =] to {{RTCSignalingState/"have-remote-pranswer"}}.

      5. If description is of type {{RTCSdpType/"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.[[\SctpTransport]] to null.

      6. Let trackEventInits, muteTracks, addList, removeList and errorList be empty lists.

      7. If description is of type {{RTCSdpType/"answer"}} or {{RTCSdpType/"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, [= create an RTCSctpTransport =] with an initial state of {{RTCSctpTransportState/"connecting"}} and assign the result to the [[\SctpTransport]] slot. Otherwise, if an SCTP association is established, but the max-message-size SDP attribute is updated, [= update the data max message size =] of connection.[[\SctpTransport]].

        2. If description negotiates the DTLS role of the SCTP transport, then for each {{RTCDataChannel}}, channel, with a null {{RTCDataChannel/id}}, run the following step:

          1. Give channel a new ID generated according to [[!RTCWEB-DATA-PROTOCOL]]. If no available ID could be generated, set channel.[[\ReadyState]] to {{RTCDataChannelState/"closed"}}, and add channnel to errorList.
      8. If description is not of type {{RTCSdpType/"rollback"}}, then run the following steps:

        1. If remote is false, then run the following steps for each [= media description =] in description:

          1. If the [= media description =] is not yet [= associated =] with an {{RTCRtpTransceiver}} object then run the following steps:

            1. Let transceiver be the {{RTCRtpTransceiver}} used to create the [= media description =].

            2. Set transceiver.{{RTCRtpTransceiver/mid}} value to the [= media stream "identification-tag" =] of the [= media description =].

            3. If transceiver.[[\Stopped]] is true, abort these sub steps.

            4. If the [= media description =] is indicated as using an existing [= media transport =] according to [[!BUNDLE]], let transport be the {{RTCDtlsTransport}} object representing the RTP/RTCP component of that transport.

            5. Otherwise, let transport be a newly created {{RTCDtlsTransport}} object with a new underlying {{RTCIceTransport}}.

            6. Set transceiver.[[\Sender]].[[\SenderTransport]] to transport.

            7. Set transceiver.[[\Receiver]].[[\ReceiverTransport]] to transport.

          2. Let transceiver be the {{RTCRtpTransceiver}} [= associated =] with the [= media description =].

          3. If transceiver.[[\Stopped]] is true, abort these sub steps.

          4. Let direction be an {{RTCRtpTransceiverDirection}} value representing the direction from the [= media description =].

          5. If direction is {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"recvonly"}}, set transceiver.[[\Receptive]] to true, otherwise set it to false.

          6. Set transceiver.[[\Receiver]].[[\ReceiveCodecs]] to the codecs that description negotiates for receiving and which the user agent is currently prepared to receive.

            If the direction is {{RTCRtpTransceiverDirection/"sendonly"}} or {{RTCRtpTransceiverDirection/"inactive"}}, the receiver is not prepared to receive anything, and the list will be empty.

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

            1. Set transceiver.[[\Sender]].[[\SendCodecs]] to the codecs that description negotiates for sending and which the user agent is currently capable of sending, and set transceiver.[[\Sender]].[[\LastReturnedParameters]] to null.

            2. If direction is {{RTCRtpTransceiverDirection/"sendonly"}} or {{RTCRtpTransceiverDirection/"inactive"}}, and transceiver.[[\FiredDirection]] is either {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"recvonly"}}, then run the following steps:

              1. [= Set the associated remote streams =] given transceiver.[[\Receiver]], an empty list, another empty list, and removeList.

              2. [= process the removal of a remote track =] for the [= media description =], given transceiver and muteTracks.

            3. Set transceiver.[[\CurrentDirection]] and transceiver.[[\FiredDirection]] to direction.

        2. Otherwise, (if remote is true) run the following steps for each [= media description =] in description:

          1. If the description is of type {{RTCSdpType/"offer"}} and contains a request to receive simulcast, use the order of the rid values specified in the simulcast attribute to create an {{RTCRtpEncodingParameters}} dictionary for each of the simulcast layers, populating the {{RTCRtpCodingParameters/rid}} member according to the corresponding rid value, and let sendEncodings be the list containing the created dictionaries. Otherwise, let sendEncodings be an empty list.

          2. Let supportedEncodings be the maximum number of encodings that the implementation can support. If the length of sendEncodings is greater than supportedEncodings, truncate sendEncodings so that its length is supportedEncodings.
          3. As described by [[!JSEP]], attempt to find an existing {{RTCRtpTransceiver}} object, transceiver, to represent the [= media description =].

          4. If a suitable transceiver was found (transceiver is set) and sendEncodings is non-empty, set transceiver.[[\Sender]].[[\SendEncodings]] to sendEncodings, and set transceiver.[[\Sender]].[[\LastReturnedParameters]] to null.

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

            1. [= Create an RTCRtpSender =], sender, from the [= media description =] using sendEncodings.

            2. [= Create an RTCRtpReceiver =], receiver, from the [= media description =].

            3. [= Create an RTCRtpTransceiver =] with sender, receiver and an {{RTCRtpTransceiverDirection}} value of {{RTCRtpTransceiverDirection/"recvonly"}}, and let transceiver be the result.

            4. Add transceiver to the connection's [= set of transceivers =].

          6. If description is of type {{RTCSdpType/"answer"}} or {{RTCSdpType/"pranswer"}}, and transceiver. [[\Sender]].[[\SendEncodings]] .length is greater than 1, then run the following steps:

            1. If description indicates that simulcast is not supported or desired, then remove all dictionaries in transceiver.[[\Sender]].[[\SendEncodings]] except the first one and abort these sub steps.

            2. If description rejects any of the offered layers, then remove the dictionaries that correspond to rejected layers from transceiver.[[\Sender]].[[\SendEncodings]].

            3. Update the paused status as indicated by [[MMUSIC-SIMULCAST]] of each simulcast layer by setting the {{RTCRtpEncodingParameters/active}} member on the corresponding dictionaries in transceiver.[[\Sender]].[[\SendEncodings]] to true for unpaused or to false for paused.

          7. Set transceiver.{{RTCRtpTransceiver/mid}} value to the [= media stream "identification-tag" =] of the corresponding [= media description =]. If the [=media description =] has no [= media stream "identification-tag" =], and transceiver.{{RTCRtpTransceiver/mid}} is unset then generate a random value as described in [[!JSEP]].

          8. 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. If the [= media description =] is rejected, set direction to {{RTCRtpTransceiverDirection/"inactive"}}.

          9. If direction is {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"recvonly"}}, let msids be a list of the MSIDs that the media description indicates transceiver.[[\Receiver]].[[\ReceiverTrack]] is to be associated with. Otherwise, let msids be an empty list.

            msids will be an empty list here if [= media description =] is rejected.
          10. [= Process remote tracks =] with transceiver, direction, msids, addList, removeList, and trackEventInits.

          11. Set transceiver.[[\Receiver]].[[\ReceiveCodecs]] to the codecs that description negotiates for receiving and which the user agent is currently prepared to receive.

          12. If description is of type {{RTCSdpType/"answer"}} or {{RTCSdpType/"pranswer"}}, then run the following steps:

            1. Set transceiver.[[\Sender]].[[\SendCodecs]] to the codecs that description negotiates for sending and which the user agent is currently capable of sending.

            2. Set transceiver.[[\CurrentDirection]] and transceiver.[[\Direction]]s to direction.

            3. Let transport be the {{RTCDtlsTransport}} object representing the RTP/RTCP component of the [= media transport =] used by transceiver's [= associated =] [= media description =], according to [[!BUNDLE]].

            4. Set transceiver.[[\Sender]].[[\SenderTransport]] to transport.

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

            6. Set the [[\IceRole]] of transport according to the rules of [[RFC8445]].

              The rules of [[RFC8445]] that apply here are:
              • If [[\IceRole]] is not {{RTCIceRole/unknown}}, do not modify [[\IceRole]].
              • If description is a local offer, set it to {{RTCIceRole/controlling}}.
              • If description is a remote offer, and contains a=ice-lite, set [[\IceRole]] to {{RTCIceRole/controlling}}.
              • If description is a remote offer, and does not contain a=ice-lite, set [[\IceRole]] to {{RTCIceRole/controlled}}.
              This ensures that [[\IceRole]] always has a value after the first offer is processed.
          13. If the [= media description =] is rejected, and transceiver.[[\Stopped]] is false, then [= stop the RTCRtpTransceiver =] transceiver.

      9. Otherwise, (if description is of type {{RTCSdpType/"rollback"}}) run the following steps:

        1. For each transceiver in the connection's [= set of transceivers =] run the following steps:

          1. If the transceiver was not [= associated =] with a [= media description =] prior to applying the {{RTCSessionDescription}} that is being rolled back, disassociate it and set transceiver.{{RTCRtpTransceiver/mid}} value to null.

          2. Set transceiver.[[\Sender]].[[\SenderTransport]] to transceiver.[[\Sender]].[[\LastStableStateSenderTransport]].

          3. Set transceiver.[[\Receiver]].[[\ReceiverTransport]] to transceiver.[[\Receiver]].[[\LastStableStateReceiverTransport]].

          4. Set transceiver.[[\Receiver]].[[\ReceiveCodecs]] to transceiver.[[\Receiver]].[[\LastStableStateReceiveCodecs]].

          5. If the signaling state of connection is {{RTCSignalingState/"have-remote-offer"}}, run the following sub steps:

            1. Let msids be a list of the ids of all {{MediaStream}} objects in transceiver.[[\Receiver]].[[\LastStableStateAssociatedRemoteMediaStreams]], or an empty list if there are none.

            2. Process remote tracks with transceiver, transceiver.[[\CurrentDirection]], msids, addList, removeList, and trackEventInits.

          6. If the transceiver was created by applying the {{RTCSessionDescription}} that is being rolled back, and a track has never been attached to it via {{RTCPeerConnection/addTrack()}}, then [= stop the RTCRtpTransceiver =] transceiver, and remove it from connection's [= set of transceivers =].

        2. Set connection.[[\PendingLocalDescription]] and connection.[[\PendingRemoteDescription]] to null, and set connection's [= signaling state =] to {{RTCSignalingState/"stable"}}.

      10. If description is of type {{RTCSdpType/"answer"}}, then run the following steps:

        1. For each transceiver in the connection's [= set of transceivers =] run the following steps:

          1. If transceiver is {{RTCRtpTransceiver/stopped}}, [= associated =] with an m= section and the associated m= section is rejected in connection.[[\CurrentLocalDescription]] or connection.[[\CurrentRemoteDescription]], remove the transceiver from the connection's [= set of transceivers =].

      11. If connection's [= signaling state =] is now {{RTCSignalingState/"stable"}}, [= update the negotiation-needed flag =]. If connection.[[\NegotiationNeeded]] was true both before and after this update,

        [= Chain =] a step to queue a task that runs the following steps, to connection's [= operations chain =]:

        1. If connection.[[\IsClosed]] is true, abort these steps.

        2. If connection.[[\NegotiationNeeded]] is false, abort these steps.

        3. [= Fire an event =] named {{negotiationneeded}} at connection.

      12. If connection's [= signaling state =] changed above, [= fire an event =] named {{signalingstatechange}} at connection.

      13. For each channel in errorList, [= fire an event =] named {{RTCDataChannel/error}} using the {{RTCErrorEvent}} interface with the {{RTCError/errorDetail}} attribute set to {{RTCErrorDetailType/"data-channel-failure"}} at channel.

      14. For each track in muteTracks, [= set the muted state =] of track to the value true.

      15. For each stream and track pair in removeList, [= remove the track =] track from stream.

      16. For each stream and track pair in addList, [= add the track =] track to stream.

      17. For each entry entry in trackEventInits, [= fire an event =] named {{track}} using the {{RTCTrackEvent}} interface with its {{RTCTrackEvent/receiver}} attribute initialized to entry.{{RTCTrackEventInit/receiver}}, its {{RTCTrackEvent/track}} attribute initialized to entry.{{RTCTrackEventInit/track}}, its {{RTCTrackEvent/streams}} attribute initialized to entry.{{RTCTrackEventInit/streams}} and its {{RTCTrackEvent/transceiver}} attribute initialized to entry.{{RTCTrackEventInit/transceiver}} at the connection object.

      18. [= Resolve =] p with undefined.

  4. Return p.

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.{{RTCConfiguration/certificates}} is set, run the following steps:

    1. If the length of configuration.{{RTCConfiguration/certificates}} is different from the length of connection.[[\Configuration]].{{RTCConfiguration/certificates}}, [= exception/throw =] an {{InvalidModificationError}}.

    2. Let index be initialized to 0.

    3. Let size be initialized to the length of configuration.{{RTCConfiguration/certificates}}.

    4. While index is less than size, run the following steps:

      1. If the ECMAScript object represented by the value of configuration.{{RTCConfiguration/certificates}} at index is not the same as the ECMAScript object represented by the value of connection.[[\Configuration]].{{RTCConfiguration/certificates}} at index, [= exception/throw =] an {{InvalidModificationError}}.

      2. Increment index by 1.

  4. If the value of configuration.{{RTCConfiguration/bundlePolicy}} is set and its value differs from the connection's bundle policy, [= exception/throw =] an {{InvalidModificationError}}.

  5. If the value of configuration.{{RTCConfiguration/rtcpMuxPolicy}} is set and its value differs from the connection's rtcpMux policy, [= exception/throw =] an {{InvalidModificationError}}.

  6. If the value of configuration.{{RTCConfiguration/iceCandidatePoolSize}} is set and its value differs from the connection's previously set {{RTCConfiguration/iceCandidatePoolSize}}, and {{RTCPeerConnection/setLocalDescription}} has already been called, [= exception/throw =] an {{InvalidModificationError}}.

  7. Set the [= ICE Agent =]'s ICE transports setting to the value of configuration.{{RTCConfiguration/iceTransportPolicy}}. As defined in [[!JSEP]], 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.

  8. Set the [= ICE Agent =]'s prefetched ICE candidate pool size as defined in [[!JSEP]] to the value of configuration.{{RTCConfiguration/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]].

  9. Let validatedServers be an empty list.

  10. If configuration.{{RTCConfiguration/iceServers}} is defined, then run the following steps for each element:

    1. Let server be the current list element.

    2. Let urls be server.{{RTCIceServer/urls}}.

    3. If urls is a string, set urls to a list consisting of just that string.

    4. If urls is empty, [= exception/throw =] a {{SyntaxError}}.

    5. For each url in 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, [= exception/throw =] a {{SyntaxError}}. If the scheme name is not implemented by the browser [= exception/throw =] a {{NotSupportedError}}. If scheme name is turn or turns, and parsing the url using the syntax defined in [[!RFC7065]] fails, [= exception/throw =] a {{SyntaxError}}. If scheme name is stun or stuns, and parsing the url using the syntax defined in [[!RFC7064]] fails, [= exception/throw =] a {{SyntaxError}}.

      2. If scheme name is turn or turns, and either of server.{{RTCIceServer/username}} or server.{{RTCIceServer/credential}} are omitted, then [= exception/throw =] an {{InvalidAccessError}}.

      3. If scheme name is turn or turns, and server.{{RTCIceServer/credentialType}} is {{RTCIceCredentialType/"password"}}, and server.{{RTCIceServer/credential}} is not a DOMString, then [= exception/throw =] an {{InvalidAccessError}}.

    6. Append server to validatedServers.

    Let validatedServers be the [= ICE Agent =]'s ICE servers list.

    As defined in [[!JSEP]], 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 size | ICE candidate pool =] has a nonzero size, any existing pooled candidates will be discarded, and new candidates will be gathered from the new servers.

  11. Store configuration in the [[\Configuration]] internal slot.

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.

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

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

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

currentLocalDescription of type {{RTCSessionDescription}}, readonly, nullable

The {{currentLocalDescription}} attribute MUST return [[\CurrentLocalDescription]].

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

pendingLocalDescription of type {{RTCSessionDescription}}, readonly, nullable

The {{pendingLocalDescription}} attribute MUST return [[\PendingLocalDescription]].

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

remoteDescription of type {{RTCSessionDescription}}, readonly, nullable

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

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

currentRemoteDescription of type {{RTCSessionDescription}}, readonly, nullable

The {{currentRemoteDescription}} attribute MUST return [[\CurrentRemoteDescription]].

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

pendingRemoteDescription of type {{RTCSessionDescription}}, readonly, nullable

The {{pendingRemoteDescription}} attribute MUST return [[\PendingRemoteDescription]].

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

signalingState of type {{RTCSignalingState}}, readonly

The {{signalingState}} attribute MUST return the {{RTCPeerConnection/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]]. Prior to the completion of {{RTCPeerConnection/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 {{MediaStreamTrack}}s 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]], except the user agent MUST treat a {{RTCRtpTransceiver/stopping}} transceiver as {{RTCRtpTransceiver/stopped}} for the purposes of JSEP in this case.

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

The generated SDP will also contain the [= ICE agent =]'s {{RTCIceParameters/usernameFragment}}, {{RTCIceParameters/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 {{RTCConfiguration/certificates}} value in configuration for the {{RTCPeerConnection}} provides the certificates configured by the application for the {{RTCPeerConnection}}. These certificates, along with any default certificates are used to produce a set of certificate fingerprints. These certificate fingerprints are used in the construction of SDP.

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

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

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

  2. If connection.[[\IsClosed]] is true, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.

  3. Return the result of [= chaining =] the result of [= creating an offer =] with connection to connection's [= operations chain =].

To create an offer given connection run the following steps:

  1. If connection's [= signaling state =] is neither {{RTCSignalingState/"stable"}} nor {{RTCSignalingState/"have-local-offer"}}, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.

  2. Let p be a new promise.

  3. In parallel, begin the [= in-parallel steps to create an offer =] given connection and p.

  4. Return p.

The in-parallel steps to create an offer given connection and 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. Inspect the offerer's system state to determine the currently available resources as necessary for generating the offer, as described in [[!JSEP]].

  3. If this inspection failed for any reason, [= reject =] p with a newly [= exception/create | created =] {{OperationError}} and abort these steps.

  4. Queue a task that runs the [= final steps to create an offer =] given connection and p.

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

  1. If connection.[[\IsClosed]] is true, then abort these steps.

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

    This may be necessary if, for example, {{createOffer}} was called when only an audio {{RTCRtpTransceiver}} was added to connection, but while performing the [= in-parallel steps to create an offer =], a video {{RTCRtpTransceiver}} was added, requiring additional inspection of video system resources.
  3. Given the information that was obtained from previous inspection, the current state of connection and its {{RTCRtpTransceiver}}s, generate an SDP offer, sdpString, as described in [[!JSEP]].

    1. As described in [[!BUNDLE]] (Section 7), if bundling is used (see {{RTCBundlePolicy}}) an offerer tagged m= section must be selected in order to negotiate a BUNDLE group. The user agent MUST choose the m= section that corresponds to the first non-stopped transceiver in the [= set of transceivers =] as the offerer tagged m= section. This allows the remote endpoint to predict which transceiver is the offerer tagged m= section without having to parse the SDP.

    2. The codec preferences of a [= media description =]'s [= associated =] transceiver is said to be the value of the {{RTCRtpTransceiver}}.[[\PreferredCodecs]] with the following filtering applied (or said not to be set if [[\PreferredCodecs]] is empty):

      1. If the {{RTCRtpTransceiver/direction}} is {{RTCRtpTransceiverDirection/"sendrecv"}}, exclude any codecs not included in the intersection of {{RTCRtpSender}}.{{RTCRtpSender/getCapabilities}}(kind).{{RTCRtpCapabilities/codecs}} and {{RTCRtpReceiver}}.{{RTCRtpReceiver/getCapabilities}}(kind).{{RTCRtpCapabilities/codecs}}.

      2. If the {{RTCRtpTransceiver/direction}} is {{RTCRtpTransceiverDirection/"sendonly"}}, exclude any codecs not included in {{RTCRtpSender}}.{{RTCRtpSender/getCapabilities}}(kind).{{RTCRtpCapabilities/codecs}}.

      3. If the {{RTCRtpTransceiver/direction}} is {{RTCRtpTransceiverDirection/"recvonly"}}, exclude any codecs not included in {{RTCRtpReceiver}}.{{RTCRtpReceiver/getCapabilities}}(kind).{{RTCRtpCapabilities/codecs}}.

      The filtering MUST NOT change the order of the codec preferences.

    3. If the length of the [[\SendEncodings]] slot of the {{RTCRtpSender}} is larger than 1, then for each encoding given in [[\SendEncodings]] of the {{RTCRtpSender}}, add an a=rid send line to the corresponding media section, and add an a=simulcast:send line giving the RIDs in the same order as given in the {{RTCRtpSendParameters/encodings}} field. No RID restrictions are set.

      [[SDP-SIMULCAST]] section 5.2 specifies that the order of RIDs in the a=simulcast line suggests a proposed order of preference. If the browser decides not to transmit all encodings, one should expect it to stop sending the last encoding in the list first.

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

  5. Set the [[\LastCreatedOffer]] 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 {{MediaStreamTrack}}s 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 {{RTCIceParameters/usernameFragment}}, {{RTCIceParameters/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 {{RTCConfiguration/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.

An answer can be marked as provisional, as described in [[!JSEP]], by setting the {{RTCSessionDescription/type}} to {{RTCSdpType/"pranswer"}}.

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.[[\IsClosed]] is true, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.

  3. Return the result of [= chaining =] the result of [= creating an answer =] with connection to connection's [= operations chain =].

To create an answer given connection run the following steps:

  1. If connection's [= signaling state =] is neither {{RTCSignalingState/"have-remote-offer"}} nor {{RTCSignalingState/"have-local-pranswer"}}, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.

  2. Let p be a new promise.

  3. In parallel, begin the [= in-parallel steps to create an answer =] given connection and p.

  4. Return p.

The in-parallel steps to create an answer given connection and 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. Inspect the answerer's system state to determine the currently available resources as necessary for generating the answer, as described in [[!JSEP]].

  3. If this inspection failed for any reason, [= reject =] p with a newly [= exception/create | created =] {{OperationError}} and abort these steps.

  4. 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.[[\IsClosed]] is true, then abort these steps.

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

    This may be necessary if, for example, {{createAnswer}} was called when an {{RTCRtpTransceiver}}'s direction was {{RTCRtpTransceiverDirection/"recvonly"}}, but while performing the [= in-parallel steps to create an answer =], the direction was changed to {{RTCRtpTransceiverDirection/"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, generate an SDP answer, sdpString, as described in [[!JSEP]].

    1. The codec preferences of an m= section's associated transceiver is said to be the value of the {{RTCRtpTransceiver}}.[[\PreferredCodecs]] with the following filtering applied (or said not to be set if [[\PreferredCodecs]] is empty):

      1. If the {{RTCRtpTransceiver/direction}} is {{RTCRtpTransceiverDirection/"sendrecv"}}, exclude any codecs not included in the intersection of {{RTCRtpSender}}.{{RTCRtpSender/getCapabilities}}(kind).{{RTCRtpCapabilities/codecs}} and {{RTCRtpReceiver}}.{{RTCRtpReceiver/getCapabilities}}(kind).{{RTCRtpCapabilities/codecs}}.

      2. If the {{RTCRtpTransceiver/direction}} is {{RTCRtpTransceiverDirection/"sendonly"}}, exclude any codecs not included in {{RTCRtpSender}}.{{RTCRtpSender/getCapabilities}}(kind).{{RTCRtpCapabilities/codecs}}.

      3. If the {{RTCRtpTransceiver/direction}} is {{RTCRtpTransceiverDirection/"recvonly"}}, exclude any codecs not included in {{RTCRtpReceiver}}.{{RTCRtpReceiver/getCapabilities}}(kind).{{RTCRtpCapabilities/codecs}}.

      The filtering MUST NOT change the order of the codec preferences.

    2. If the length of the [[\SendEncodings]] slot of the {{RTCRtpSender}} is larger than 1, then for each encoding given in [[\SendEncodings]] of the {{RTCRtpSender}}, add an a=rid send line to the corresponding media section, and add an a=simulcast:send line giving the RIDs in the same order as given in the {{RTCRtpSendParameters/encodings}} field. No RID restrictions are set.

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

  5. Set the [[\LastCreatedAnswer]] 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.

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

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

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

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

  3. Let sdp be description.{{RTCSessionDescription/sdp}}.

  4. Return the result of [= chaining =] the following steps to connection's [= operations chain =]:

    1. Let type be description.{{RTCSessionDescription/type}} if present, or {{RTCSdpType/"offer"}} if not present and connection's [= signaling state =] is either {{RTCSignalingState/"stable"}}, {{RTCSignalingState/"have-local-offer"}}, or {{RTCSignalingState/"have-remote-pranswer"}}; otherwise {{RTCSdpType/"answer"}}.

    2. If type is {{RTCSdpType/"offer"}}, and sdp is not the empty string and not equal to connection.[[\LastCreatedOffer]], then return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidModificationError}} and abort these steps.

    3. If type is {{RTCSdpType/"answer"}} or {{RTCSdpType/"pranswer"}}, and sdp is not the empty string and not equal to connection.[[\LastCreatedAnswer]], then return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidModificationError}} and abort these steps.

    4. If sdp is the empty string, and type is {{RTCSdpType/"offer"}}, then run the following sub steps:

      1. Set sdp to the value of connection.[[\LastCreatedOffer]].

      2. If sdp is the empty string, or if it no longer accurately represents the [= offerer's system state =] of connection, then let p be the result of [= creating an offer =] with connection, and return the result of [= promise/reacting =] to p with a fulfillment step that [= set a local RTCSessionDescription | sets the local RTCSessionDescription =] indicated by its first argument.

    5. If sdp is the empty string, and type is {{RTCSdpType/"answer"}} or {{RTCSdpType/"pranswer"}}, then run the following sub steps:

      1. Set sdp to the value of connection.[[\LastCreatedAnswer]].

      2. If sdp is the empty string, or if it no longer accurately represents the [= answerer's system state =] of connection, then let p be the result of [= creating an answer =] with connection, and return the result of [= promise/reacting =] to p with the following fulfillment steps:

        1. Let answer be the first argument to these fulfillment steps.

        2. Return the result of [= setting the local RTCSessionDescription =] indicated by {type, answer.{{RTCSessionDescription/sdp}}}.

    6. Return the result of [= setting the local RTCSessionDescription =] indicated by {type, sdp}.

As noted in [[!JSEP]], calling this method may trigger the ICE candidate gathering process by the [= ICE Agent =].

setRemoteDescription

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

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

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

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

  3. If description.{{RTCSessionDescriptionInit/type}} is not present, [= exception/throw =] a {{TypeError}}.

  4. Return the result of [= chaining =] the following steps to connection's [= operations chain =]:

    1. If description.{{RTCSessionDescriptionInit/type}} is {{RTCSdpType/"offer"}} and is invalid for the current [= signaling state =] of connection as described in [[!JSEP]], then run the following sub steps:

      1. Let p be the result of [= setting the local RTCSessionDescription =] indicated by {type: {{RTCSdpType/"rollback"}}}.

      2. Return the result of [= promise/reacting =] to p with a fulfillment step that [= set a remote RTCSessionDescription | sets the remote RTCSessionDescription =] description, and abort these steps.

    2. Return the result of [= setting the remote RTCSessionDescription =] description.

addIceCandidate

The {{addIceCandidate}} method provides a remote candidate to the [= ICE Agent =]. This method can also be used to indicate the end of remote candidates when called with an empty string for the {{RTCIceCandidate/candidate}} member. The only members of the argument used by this method are {{RTCIceCandidate/candidate}}, {{RTCIceCandidate/sdpMid}}, {{RTCIceCandidate/sdpMLineIndex}}, and {{RTCIceCandidate/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 candidate.{{RTCIceCandidate/candidate}} is not an empty string and both candidate.{{RTCIceCandidate/sdpMid}} and candidate.{{RTCIceCandidate/sdpMLineIndex}} are null, return a promise [= rejected =] with a newly [= exception/create | created =] {{TypeError}}.

  4. Return the result of [= chaining =] the following steps to connection's [= operations chain =]:

    1. If {{RTCPeerConnection/remoteDescription}} is null return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.

    2. Let p be a new promise.

    3. If candidate.{{RTCIceCandidate/sdpMid}} is not null, run the following steps:

      1. If candidate.{{RTCIceCandidate/sdpMid}} is not equal to the mid of any media description in {{RTCPeerConnection/remoteDescription}}, [= reject =] p with a newly [= exception/create | created =] {{OperationError}} and abort these steps.

    4. Else, if candidate.{{RTCIceCandidate/sdpMLineIndex}} is not null, run the following steps:

      1. If candidate.{{RTCIceCandidate/sdpMLineIndex}} is equal to or larger than the number of media descriptions in {{RTCPeerConnection/remoteDescription}}, [= reject =] p with a newly [= exception/create | created =] {{OperationError}} and abort these steps.

    5. If either candidate.{{RTCIceCandidate/sdpMid}} or candidate.{{RTCIceCandidate/sdpMLineIndex}} indicate a media description in {{RTCPeerConnection/remoteDescription}} whose associated transceiver is {{RTCRtpTransceiver/ stopped}}, [= resolve =] p with undefined and abort these steps.

    6. If candidate.{{RTCIceCandidate/usernameFragment}} is not 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 [= exception/create | created =] {{OperationError}} and abort these steps.

    7. In parallel, add the ICE candidate candidate as described in [[!JSEP]]. Use candidate.{{RTCIceCandidate/usernameFragment}} to identify the ICE [= generation =]; if {{RTCIceCandidate/usernameFragment}} is null, process the candidate for the most recent ICE [= generation =]. If candidate.{{RTCIceCandidate/candidate}} is an empty string, process candidate as an end-of-candidates indication for the corresponding [= media description =] and ICE candidate [= generation =]. If both candidate.{{RTCIceCandidate/sdpMid}} and candidate.{{RTCIceCandidate/sdpMLineIndex}} are null, then this applies to all [= media description =]s.

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

        1. If connection.[[\IsClosed]] is true, then abort these steps.

        2. [= Reject =] p with a newly [= exception/create | created =] {{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.[[\IsClosed]] is true, then abort these steps.

        2. If connection.[[\PendingRemoteDescription]] is not null, and represents the ICE [= generation =] for which candidate was processed, add candidate to connection.[[\PendingRemoteDescription]].sdp.

        3. If connection.[[\CurrentRemoteDescription]] is not null, and represents the ICE [= generation =] for which candidate was processed, add candidate to connection.[[\CurrentRemoteDescription]].sdp.

        4. [= Resolve =] p with undefined.

    8. Return p.

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

restartIce

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

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

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

  2. Empty connection.[[\LocalIceCredentialsToReplace]], and populate it with all ICE credentials (ice-ufrag and ice-pwd as defined in section 15.4 of [[!ICE]]) found in connection.[[\CurrentLocalDescription]], as well as all ICE credentials found in connection.[[\PendingLocalDescription]].

  3. [= Update the negotiation-needed flag =] for connection.

getConfiguration

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

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

setConfiguration

The {{setConfiguration}} method updates the configuration of this {{RTCPeerConnection}} object. This includes changing the configuration of the [= ICE Agent =]. As noted in [[!JSEP]], 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.[[\IsClosed]] is true, [= exception/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.[[\IsClosed]] is true, abort these steps.

  3. Set connection.[[\IsClosed]] to true.

  4. Set connection's [= signaling state =] to {{RTCSignalingState/"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.[[\Stopped]] is true, abort these sub steps.

    2. [= Stop the RTCRtpTransceiver =] with transceiver and the value true.

  6. Set the [[\ReadyState]] slot of each of connection's {{RTCDataChannel}}s to {{RTCDataChannelState/"closed"}}.

    The {{RTCDataChannel}}s will be closed abruptly and the closing procedure will not be invoked.
  7. If connection.[[\SctpTransport]] is not null, tear down the underlying SCTP association by sending an SCTP ABORT chunk and set the [[\SctpTransportState]] to {{RTCSctpTransportState/"closed"}}.

  8. Set the [[\DtlsTransportState]] slot of each of connection's {{RTCDtlsTransport}}s to {{RTCDtlsTransportState/"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 {{RTCIceTransportState/"closed"}}.

  11. Set connection's [= ICE connection state =] to {{RTCIceConnectionState/"closed"}}. This does not fire any event.

  12. Set connection's [= connection state =] to {{RTCPeerConnectionState/"closed"}}.

Legacy Interface Extensions

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

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

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));
};

Method extensions

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 {{RTCPeerConnection/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 {{RTCPeerConnection/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
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 {{RTCPeerConnection/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 {{RTCPeerConnection/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 {{RTCPeerConnection/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 = void (RTCSessionDescriptionInit description);

Callback {{RTCSessionDescriptionCallback}} Parameters

description of type {{RTCSessionDescriptionInit}}
The object containing the SDP [[!SDP]].

Legacy configuration extensions

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

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

  1. Let options be the methods first argument.

  2. Let connection be the current {{RTCPeerConnection}} object.

  3. For each offerToReceive<Kind> member in options with kind, kind, run the following steps:

    1. If the value of the dictionary member is false,
      1. For each non-stopped {{RTCRtpTransceiverDirection/"sendrecv"}} transceiver of [= transceiver kind =] kind, set transceiver.[[\Direction]] to {{RTCRtpTransceiverDirection/"sendonly"}}.

      2. For each non-stopped {{RTCRtpTransceiverDirection/"recvonly"}} transceiver of [= transceiver kind =] kind, set transceiver.[[\Direction]] to {{RTCRtpTransceiverDirection/"inactive"}}.

      Continue with the next option, if any.

    2. If connection has any non-stopped {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"recvonly"}} transceivers of [= transceiver kind =] kind, continue with the next option, if any.

    3. Let transceiver be the result of invoking the equivalent of connection.{{RTCPeerConnection/addTransceiver}}(kind), except that this operation MUST NOT [= update the negotiation-needed flag =].

    4. If transceiver is unset because the previous operation threw an error, abort these steps.

    5. Set transceiver.[[\Direction]] to {{RTCRtpTransceiverDirection/"recvonly"}}.

  4. Run the steps specified by {{RTCPeerConnection/createOffer}} to create the offer.

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

Attributes

offerToReceiveAudio of type boolean

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

offerToReceiveVideo of type boolean

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

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.

Error Handling

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.

Session Description Model

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 {{RTCSdpType/"offer"}} indicates that a description MUST be treated as an [[!SDP]] offer.

pranswer

An {{RTCSdpType}} of {{RTCSdpType/"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 {{RTCSdpType/"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 {{RTCSdpType/"rollback"}} indicates that a description MUST be treated as canceling the current SDP negotiation and moving the SDP [[!SDP]] offer back to what it was in the previous stable state. Note the local or remote SDP descriptions in the previous stable state could be null if there has not yet been a successful offer-answer negotiation. An {{RTCSdpType/"answer"}} or {{RTCSdpType/"pranswer"}} cannot be rolled back.

RTCSessionDescription Class

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

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

Constructors

constructor()
The RTCSessionDescription() constructor takes a dictionary argument, description, whose content is used to initialize the new {{RTCSessionDescription}} object. This constructor is deprecated; it exists for legacy compatibility reasons only. The constructor MUST [= exception/throw =] a {{TypeError}} if description.{{type}} is not present.

Attributes

type of type {{RTCSdpType}}, readonly
The type of this RTCSessionDescription.
sdp of type DOMString, readonly, defaulting to ""
The string representation of the SDP [[!SDP]].

Methods

toJSON()
When called, run [[!WEBIDL]]'s [= default toJSON operation =].
dictionary RTCSessionDescriptionInit {
  RTCSdpType type;
  DOMString sdp = "";
};

Dictionary RTCSessionDescriptionInit Members

type of type {{RTCSdpType}}
The type of this description. If not present, then {{RTCPeerConnection/setLocalDescription}} will infer the type based on the {{RTCPeerConnection}}'s [= signaling state =], whereas {{RTCPeerConnection/setRemoteDescription}} and the {{RTCSessionDescription}} constructor will [= exception/throw =] a {{TypeError}}, because they require the argument.
sdp of type DOMString
The string representation of the SDP [[!SDP]]; if {{RTCSessionDescription/type}} is {{RTCSdpType/"rollback"}}, this member is unused.

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.

Setting Negotiation-Needed

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 [= update the negotiation-needed flag | updating the negotiation-needed flag =] are specified below.

Clearing Negotiation-Needed

The negotiation-needed flag is cleared when an {{RTCSessionDescription}} of type {{RTCSdpType/"answer"}} [= set an RTCSessionDescription | 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-{{RTCRtpTransceiver/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 [= update the negotiation-needed flag | updating the negotiation-needed flag =] are specified below.

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.[[\IsClosed]] is true, abort these steps.

  2. If connection's [= signaling state =] is not {{RTCSignalingState/"stable"}}, abort these steps.

    The negotiation-needed flag will be updated once the state transitions to {{RTCSignalingState/"stable"}}, as part of the steps for [= setting an RTCSessionDescription =].

  3. If the result of [= check if negotiation is needed | checking if negotiation is needed =] is false, clear the negotiation-needed flag by setting connection.[[\NegotiationNeeded]] to false, and abort these steps.

  4. If connection.[[\NegotiationNeeded]] is already true, abort these steps.

  5. Set connection.[[\NegotiationNeeded]] to true.

  6. [= Chain =] a step to queue a task that runs the following steps, to connection's [= operations chain =]:

    1. If connection.[[\IsClosed]] is true, abort these steps.

    2. If connection.[[\NegotiationNeeded]] is false, abort these steps.

    3. [= Fire an event =] named {{negotiationneeded}} at connection.

    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. If connection.[[\LocalIceCredentialsToReplace]] is not empty, return true.

  3. Let description be connection.[[\CurrentLocalDescription]].

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

  5. For each transceiver in connection's [= set of transceivers =], perform the following checks:

    1. If transceiver.[[\Stopping]] is true and transceiver.[[\Stopped]] is false, return true.

    2. If transceiver isn't {{RTCRtpTransceiver/ stopped}} and isn't yet [= associated =] with an m= section in description, return true.

    3. If transceiver isn't {{RTCRtpTransceiver/ stopped}} and is [= associated =] with an m= section in description then perform the following checks:

      1. If transceiver.[[\Direction]] is {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"sendonly"}}, and the [= associated =] m= section in description either doesn't contain a single a=msid line, or the number of MSIDs from the a=msid lines in this m= section, or the MSID values themselves, differ from what is in transceiver.sender.[[\AssociatedMediaStreamIds]], return true.

      2. If description is of type {{RTCSdpType/"offer"}}, and the direction of the [= associated =] m= section in neither connection.[[\CurrentLocalDescription]] nor connection.[[\CurrentRemoteDescription]] matches transceiver.[[\Direction]], return true. In this step, when the direction is compared with a direction found in [[\CurrentRemoteDescription]], the description's direction must be reversed to represent the peer's point of view.

      3. If description is of type {{RTCSdpType/"answer"}}, and the direction of the [= associated =] m= section in the description does not match transceiver.[[\Direction]] intersected with the offered direction (as described in [[!JSEP]]), return true.

    4. If transceiver is {{RTCRtpTransceiver/ stopped}} and is [= associated =] with an m= section, but the associated m= section is not yet rejected in connection.[[\CurrentLocalDescription]] or connection.[[\CurrentRemoteDescription]], return true.

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

Interfaces for Connectivity Establishment

RTCIceCandidate Interface

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

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

Constructor

constructor()

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

When invoked, run the following steps:

  1. If both the {{RTCIceCandidateInit/sdpMid}} and {{RTCIceCandidateInit/sdpMLineIndex}} members of candidateInitDict are null, [= exception/throw =] a {{TypeError}}.
  2. Return the result of [= creating an RTCIceCandidate =] with candidateInitDict.

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

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

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

To maintain backward compatibility, any error on parsing the candidate attribute is ignored. In such case, the {{candidate}} attribute holds the raw {{RTCIceCandidateInit/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 or a peer reflexive remote candidate, {{candidate}} is an empty string.
sdpMid of type DOMString, readonly, nullable
If not null, this contains the media stream "identification-tag" defined in [[!RFC5888]] for the media component this candidate is associated with.
sdpMLineIndex of type unsigned short, readonly, nullable
If not null, this indicates the index (starting at zero) of the [= media description =] in the SDP this candidate is associated with.
foundation of type DOMString, readonly, nullable
A unique identifier that allows ICE to correlate candidates that appear on multiple {{RTCIceTransport}}s.
component of type {{RTCIceComponent}}, readonly, nullable
The assigned network component of the candidate ({{RTCIceComponent/"rtp"}} or {{RTCIceComponent/"rtcp"}}). This corresponds to the component-id field in [= candidate-attribute =], decoded to the string representation as defined in {{RTCIceComponent}}.
priority of type unsigned long, readonly, nullable
The assigned priority of the candidate.
address of type DOMString, readonly, nullable

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

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

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

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

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

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

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

protocol of type {{RTCIceProtocol}}, readonly, nullable
The protocol of the candidate ({{RTCIceProtocol/"udp"}}/{{RTCIceProtocol/"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 {{RTCIceProtocol/"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}}, {{usernameFragment}}»:
    1. Let value be the result of getting the underlying value of the attribute identified by attr, given this {{RTCIceCandidate}} object.
    2. Set json[attr] to value.
  3. Return json.
dictionary RTCIceCandidateInit {
  DOMString candidate = "";
  DOMString? sdpMid = null;
  unsigned short? sdpMLineIndex = null;
  DOMString? usernameFragment = null;
};

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, nullable, defaulting to null
If not null, this carries the ufrag as defined in section 15.4 of [[!ICE]].

candidate-attribute Grammar

The [= candidate-attribute =] grammar is used to parse the {{RTCIceCandidateInit/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.

RTCIceProtocol Enum

The {{RTCIceProtocol}} represents the protocol of the ICE candidate.

enum RTCIceProtocol {
  "udp",
  "tcp"
};
Enumeration description
udp A UDP candidate, as described in [[!ICE]].
tcp A TCP candidate, as described in [[!RFC6544]].

RTCIceTcpCandidateType Enum

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

enum RTCIceTcpCandidateType {
  "active",
  "passive",
  "so"
};
Enumeration description
active An {{RTCIceTcpCandidateType/"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 {{RTCIceTcpCandidateType/"passive"}} TCP candidate is one for which the transport will receive incoming connection attempts but not attempt a connection.
so An {{RTCIceTcpCandidateType/"so"}} candidate is one for which the transport will attempt to open a connection simultaneously with its peer.

The user agent will typically only gather {{RTCIceTcpCandidateType/active}} ICE TCP candidates.

RTCIceCandidateType Enum

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

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

RTCPeerConnectionIceEvent

The icecandidate event of the {{RTCPeerConnection}} uses the {{RTCPeerConnectionIceEvent}} interface.

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

The {{icecandidate}} event is used for three different types of indications:
  • A candidate has been gathered. The {{RTCPeerConnectionIceEvent/candidate}} member of the event will be populated normally. It should be signaled to the remote peer and passed into {{RTCPeerConnection/addIceCandidate}}.

  • An {{RTCIceTransport}} has finished gathering a [= generation =] of candidates, and is providing an end-of-candidates indication as defined by Section 8.2 of [[TRICKLE-ICE]]. This is indicated by {{RTCPeerConnectionIceEvent/candidate}}.{{RTCIceCandidate/candidate}} being set to an empty string. The {{RTCPeerConnectionIceEvent/candidate}} object should be signaled to the remote peer and passed into {{RTCPeerConnection/addIceCandidate}} like a typical ICE candidate, in order to provide the end-of-candidates indication to the remote peer.

  • All {{RTCIceTransport}}s have finished gathering candidates, and the {{RTCPeerConnection}}'s {{RTCIceGatheringState}} has transitioned to {{RTCIceGatheringState/"complete"}}. This is indicated by the {{RTCPeerConnectionIceEvent/candidate}} member of the event being set to null. This only exists for backwards compatibility, and this event does not need to be signaled to the remote peer. It's equivalent to an {{icegatheringstatechange}} event with the {{RTCIceGatheringState/"complete"}} state.

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

Constructors

RTCPeerConnectionIceEvent.constructor()

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.

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 {{RTCPeerConnectionIceEvent/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.

RTCPeerConnectionIceErrorEvent

The icecandidateerror event of the {{RTCPeerConnection}} uses the {{RTCPeerConnectionIceErrorEvent}} interface.

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

Constructors

RTCPeerConnectionIceErrorEvent.constructor()

Attributes

address of type DOMString, readonly, nullable

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

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

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

port of type unsigned short, readonly, nullable

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

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

url of type DOMString, readonly

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

errorCode of type unsigned short, readonly

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

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

errorText of type USVString, readonly

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

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

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

Dictionary RTCPeerConnectionIceErrorEventInit Members

address of type DOMString, nullable

The local address used to communicate with the STUN or TURN server, or null.

port of type unsigned short, nullable

The local port used to communicate with the STUN or TURN server, or null.

url of type DOMString

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

errorCode of type unsigned short, required

The numeric STUN error code returned by the STUN or TURN server.

statusText of type USVString

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

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 {{RTCPeerConnection/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 {{RTCConfiguration/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 an X.509 certificate [[!X509V3]] and corresponding private key. A handle to information is provided in the form of the {{RTCCertificate}} interface. The returned {{RTCCertificate}} can be used to control the certificate that is offered in the DTLS sessions established by {{RTCPeerConnection}}.

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

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

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

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

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

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

  1. Let keygenAlgorithm be the first argument to {{generateCertificate}}.

  2. Let expires be a {{DOMTimeStamp}} value of 2592000000.

    This means the certificate will by default expire in 30 days from the time of the {{generateCertificate}} call.

  3. If keygenAlgorithm is an object, run the following steps:

    1. Let certificateExpiration be the result of converting the ECMAScript object represented by keygenAlgorithm to an {{RTCCertificateExpiration}} dictionary.

    2. If the conversion fails with an error, return a promise that is [= rejected =] with error.

    3. If certificateExpiration.{{RTCCertificateExpiration/expires}} is not undefined, set expires to certificateExpiration.{{RTCCertificateExpiration/expires}}.

    4. If expires is greater than 31536000000, set expires to 31536000000.

      This means the certificate cannot be valid for longer than 365 days from the time of the {{generateCertificate}} call.

      A user agent MAY further cap the value of expires.

  4. Let normalizedKeygenAlgorithm be the result of normalizing an algorithm with an operation name of generateKey and a supportedAlgorithms value specific to production of certificates for {{RTCPeerConnection}}.

  5. If the above normalization step fails with an error, return a promise that is [= rejected =] with error.

  6. If the normalizedKeygenAlgorithm parameter identifies an algorithm that the user agent cannot or will not use to generate a certificate for {{RTCPeerConnection}}, return a promise that is [= rejected =] with a {{DOMException}} of type {{NotSupportedError}}. In particular, normalizedKeygenAlgorithm MUST be an asymmetric algorithm that can be used to produce a signature used to authenticate DTLS connections.

  7. Let p be a new promise.

  8. Run the following steps in parallel:

    1. Perform the generate key operation specified by normalizedKeygenAlgorithm using keygenAlgorithm.

    2. Let generatedKeyingMaterial and generatedKeyCertificate be the private keying material and certificate generated by the above step.

    3. Let certificate be a new {{RTCCertificate}} object.

    4. Set certificate.[[\Expires]] to the current time plus expires value.

    5. Set certificate.[[\Origin]] to the current settings object's origin.

    6. Store the generatedKeyingMaterial in a secure module, and let handle be a reference identifier to it.

    7. Set certificate.[[\KeyingMaterialHandle]] to handle.

    8. Set certificate.[[\Certificate]] to generatedCertificate.

    9. Resolve p with certificate.

  9. Return p.

RTCCertificateExpiration Dictionary

{{RTCCertificateExpiration}} is used to set an expiration date on certificates generated by {{RTCPeerConnection/generateCertificate}}.

dictionary RTCCertificateExpiration {
  [EnforceRange] DOMTimeStamp expires;
};
expires, of type {{DOMTimeStamp}}

An optional {{expires}} attribute MAY be added to the definition of the algorithm that is passed to {{RTCPeerConnection/generateCertificate}}. If this parameter is present it indicates the maximum time that the {{RTCCertificate}} is valid for relative to the current time.

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 ([[\KeyingMaterialHandle]]), a certificate ([[\Certificate]]) that {{RTCPeerConnection}} uses to authenticate with a peer, and the origin ([[\Origin]]) that created the object.

[Exposed=Window, Serializable]
interface RTCCertificate {
  readonly attribute DOMTimeStamp expires;
  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

getFingerprints

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

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

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

  1. Set serialized.[[\Expires]] to the value of value.{{RTCCertificate/expires}} attribute.
  2. Set serialized.[[\Certificate]] to a copy of the unstructured binary data in value.[[\Certificate]].
  3. Set serialized.[[\Origin]] to a copy of the unstructured binary data in value.[[\Origin]].
  4. Set serialized.[[\KeyingMaterialHandle]] to a serialization of the handle in value.[[\KeyingMaterialHandle]] (not the private keying material itself).

Their deserialization steps, given serialized and value, are:

  1. Initialize value.{{RTCCertificate/expires}} attribute to contain serialized.[[\Expires]].
  2. Set value.[[\Certificate]] to a copy of serialized.[[\Certificate]].
  3. Set value.[[\Origin]] to a copy of serialized.[[\Origin]].
  4. Set value.[[\KeyingMaterialHandle]] to the private keying material handle resulting from deserializing serialized.[[\KeyingMaterialHandle]].

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

RTP Media API

The RTP media API lets a web application send and receive {{MediaStreamTrack}}s 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.

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

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

Following the rules in [[!JSEP]], 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.

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 {{RTCRtpEncodingParameters/ 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. What to transmit if the integer part of the scaled width or height is zero is implementation-specific.

The actual encoding and transmission of {{MediaStreamTrack}}s is managed through objects called {{RTCRtpSender}}s. Similarly, the reception and decoding of {{MediaStreamTrack}}s 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 {{RTCRtpTransceiver}}s, 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 {{RTCPeerConnection/addTrack()}} method, or explicitly when the application uses the {{RTCPeerConnection/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.

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 belong to non-stopped {{RTCRtpTransceiver}} objects currently attached to this {{RTCPeerConnection}} object.

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

We define the CollectSenders algorithm as follows:

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

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

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

  1. Let transceivers be the result of executing the {{CollectTransceivers}} algorithm.
  2. Let receivers be a new empty sequence.
  3. For each transceiver in transceivers,
    1. If transceiver.[[\Stopped]] is false add transceiver.[[\Receiver]] to receivers.
  4. 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 consisting of all {{RTCRtpTransceiver}} objects in this {{RTCPeerConnection}} object's [= set of transceivers =], in insertion order.
  2. Return transceivers.
addTrack

Adds a new track to the {{RTCPeerConnection}}, and indicates that it is contained in the specified {{MediaStream}}s.

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 kind be track.kind.

  4. 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.

  5. If connection.[[\IsClosed]] is true, [= exception/throw =] an {{InvalidStateError}}.

  6. Let senders be the result of executing the {{CollectSenders}} algorithm. If an {{RTCRtpSender}} for track already exists in senders, [= exception/throw =] an {{InvalidAccessError}}.

  7. The steps below describe how to determine if an existing sender can be reused. Doing so will cause future calls to {{RTCPeerConnection/createOffer}} and {{RTCPeerConnection/createAnswer}} to mark the corresponding [= media description =] as sendrecv or sendonly and add the MSID of the sender's streams, as defined in [[!JSEP]].

    If any {{RTCRtpSender}} object in senders matches all the following criteria, let sender be that object, or null otherwise:

    • The sender's track is null.

    • The transceiver kind of the {{RTCRtpTransceiver}}, associated with the sender, matches kind.

    • The [[\Stopping]] slot of the {{RTCRtpTransceiver}} associated with the sender is false.

    • The sender has never been used to send. More precisely, the [[\CurrentDirection]] slot of the {{RTCRtpTransceiver}} associated with the sender has never had a value of {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"sendonly"}}.

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

    1. Set sender.[[\SenderTrack]] to track.

    2. Set sender.[[\AssociatedMediaStreamIds]] to an empty set.

    3. For each stream in streams, add stream.id to [[\AssociatedMediaStreamIds]] if it's not already there.

    4. Let transceiver be the {{RTCRtpTransceiver}} associated with sender.

    5. If transceiver.[[\Direction]] is {{RTCRtpTransceiverDirection/"recvonly"}}, set transceiver.[[\Direction]] to {{RTCRtpTransceiverDirection/"sendrecv"}}.

    6. If transceiver.[[\Direction]] is {{RTCRtpTransceiverDirection/"inactive"}}, set transceiver.[[\Direction]] to {{RTCRtpTransceiverDirection/"sendonly"}}.

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

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

    2. Create an RTCRtpReceiver with kind, and let receiver be the result.

    3. Create an RTCRtpTransceiver with sender, receiver and an {{RTCRtpTransceiverDirection}} value of {{RTCRtpTransceiverDirection/"sendrecv"}}, and let transceiver be the result.

    4. Add transceiver to connection's [= set of transceivers =].

  10. A track could have contents that are inaccessible to the application. This can be due to anything that would make a track CORS cross-origin. These tracks can be supplied to the {{RTCPeerConnection/addTrack()}} method, and have an {{RTCRtpSender}} created for them, but content MUST NOT be transmitted. Silence (audio), black frames (video) or equivalently absent content is sent in place of track content.

    Note that this property can change over time.

  11. [= Update the negotiation-needed flag =] for connection.

  12. 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 {{RTCRtpTransceiverDirection/"recvonly"}} or {{RTCRtpTransceiverDirection/"inactive"}}, as defined in [[!JSEP]].

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

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

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.[[\IsClosed]] is true, [= exception/throw =] an {{InvalidStateError}}.

  4. If sender was not created by connection, [= exception/throw =] an {{InvalidAccessError}}.

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

  6. If sender is not in senders (which indicates its transceiver was stopped or removed due to [= setting an RTCSessionDescription =] of type {{RTCSdpType/"rollback"}}), then abort these steps.

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

  8. Set sender.[[\SenderTrack]] to null.

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

  10. If transceiver.[[\Direction]] is {{RTCRtpTransceiverDirection/"sendrecv"}}, set transceiver.[[\Direction]] to {{RTCRtpTransceiverDirection/"recvonly"}}.

  11. If transceiver.[[\Direction]] is {{RTCRtpTransceiverDirection/"sendonly"}}, set transceiver.[[\Direction]] to {{RTCRtpTransceiverDirection/"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]].

The initial value of {{RTCRtpTransceiver/mid}} is null. Setting a new {{RTCSessionDescription}} may change it to a non-null value, as defined in [[!JSEP]].

The {{RTCRtpTransceiverInit/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.{{RTCRtpTransceiverInit/streams}}.

  3. Let sendEncodings be init.{{RTCRtpTransceiverInit/sendEncodings}}.

  4. Let direction be init.{{RTCRtpTransceiverInit/direction}}.

  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, [= exception/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. If connection.[[\IsClosed]] is true, [= exception/throw =] an {{InvalidStateError}}.

  8. Validate sendEncodings by running the following steps:
    1. Verify that each {{RTCRtpCodingParameters/rid}} value in sendEncodings conforms to the grammar specified in Section 10 of [[!MMUSIC-RID]]. If one of the RIDs does not meet these requirements, [= exception/throw =] a {{TypeError}}.

    2. If any {{RTCRtpEncodingParameters}} dictionary in sendEncodings contains a read-only parameter other than {{RTCRtpCodingParameters/rid}}, [= exception/throw =] an {{InvalidAccessError}}.

    3. Verify that each {{RTCRtpEncodingParameters/scaleResolutionDownBy}} value in sendEncodings is greater than or equal to 1.0. If one of the {{RTCRtpEncodingParameters/scaleResolutionDownBy}} values does not meet this requirement, [= exception/throw =] a {{RangeError}}.

    4. Let maxN be the maximum number of total simultaneous encodings the user agent may support for this kind, at minimum 1.This should be an optimistic number since the codec to be used is not known yet.

    5. If the number of {{RTCRtpEncodingParameters}} stored in sendEncodings exceeds maxN, then trim sendEncodings from the tail until its length is maxN.

    6. If the number of {{RTCRtpEncodingParameters}} now stored in sendEncodings is 1, then remove any {{RTCRtpCodingParameters/rid}} member from the lone entry.

      Providing a single, default {{RTCRtpEncodingParameters}} in sendEncodings allows the application to subsequently set encoding parameters using {{RTCRtpSender/setParameters}}, even when simulcast isn't used.
  9. Create an RTCRtpSender with track, kind, streams and sendEncodings and let sender be the result.

    If sendEncodings is set, then subsequent calls to {{createOffer}} will be configured to send multiple RTP encodings as defined in [[!JSEP]]. When {{RTCPeerConnection/setRemoteDescription}} is called with a corresponding remote description that is able to receive multiple RTP encodings as defined in [[!JSEP]], the {{RTCRtpSender}} may send multiple RTP encodings and the parameters retrieved via the transceiver's {{RTCRtpTransceiver/sender}}.{{RTCRtpSender/getParameters()}} will reflect the encodings negotiated.

  10. Create an RTCRtpReceiver with kind and let receiver be the result.

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

  12. Add transceiver to connection's [= set of transceivers =].

  13. [= Update the negotiation-needed flag =] for connection.

  14. Return transceiver.

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

Dictionary RTCRtpTransceiverInit Members

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

When the remote PeerConnection's track event fires corresponding to the {{RTCRtpReceiver}} being added, these are the streams that will be put in the event.

sendEncodings of type sequence<{{RTCRtpEncodingParameters}}>

A sequence containing parameters for sending RTP encodings of media.

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

Processing Remote MediaStreamTracks

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

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

  1. Set the associated remote streams with transceiver.[[\Receiver]], msids, addList, and removeList.

  2. If direction is {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"recvonly"}} and transceiver.[[\FiredDirection]] is neither {{RTCRtpTransceiverDirection/"sendrecv"}} nor {{RTCRtpTransceiverDirection/"recvonly"}}, or the previous step increased the length of addList, process the addition of a remote track with transceiver and trackEventInits.

  3. If direction is {{RTCRtpTransceiverDirection/"sendonly"}} or {{RTCRtpTransceiverDirection/"inactive"}}, set transceiver.[[\Receptive]] to false.

  4. If direction is {{RTCRtpTransceiverDirection/"sendonly"}} or {{RTCRtpTransceiverDirection/"inactive"}}, and transceiver.[[\FiredDirection]] is either {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"recvonly"}}, process the removal of a remote track for the media description, with transceiver and muteTracks.

  5. Set transceiver.[[\FiredDirection]] to direction.

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

  1. Let receiver be transceiver.[[\Receiver]].

  2. Let track be receiver.[[\ReceiverTrack]].

  3. Let streams be receiver.[[\AssociatedRemoteMediaStreams]].

  4. Create a new {{RTCTrackEventInit}} dictionary with receiver, track, streams and transceiver as members and add it to trackEventInits.

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

  1. Let receiver be transceiver.[[\Receiver]].

  2. Let track be receiver.[[\ReceiverTrack]].

  3. If track.muted is false, add track to muteTracks.

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

  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. Let track be receiver.[[\ReceiverTrack]].

  5. For each stream in receiver.[[\AssociatedRemoteMediaStreams]] that is not present in streams, add stream and track as a pair to removeList.

  6. For each stream in streams that is not present in receiver.[[\AssociatedRemoteMediaStreams]], add stream and track as a pair to addList.

  7. Set receiver.[[\AssociatedRemoteMediaStreams]] to streams.

RTCRtpSender Interface

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

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

  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 [[\LastStableStateSenderTransport]] internal slot initialized to null.

  5. Let sender have a [[\Dtmf]] internal slot initialized to null.

  6. If kind is "audio" then create an RTCDTMFSender dtmf and set the [[\Dtmf]] internal slot to dtmf.

  7. Let sender have an [[\AssociatedMediaStreamIds]] internal slot, representing a list of Ids of {{MediaStream}} objects that this sender is to be associated with. The [[\AssociatedMediaStreamIds]] slot is used when sender is represented in SDP as described in [[!JSEP]].

  8. Set sender.[[\AssociatedMediaStreamIds]] to an empty set.

  9. For each stream in streams, add stream.id to [[\AssociatedMediaStreamIds]] if it's not already there.

  10. Let sender have a [[\SendEncodings]] internal slot, representing a list of {{RTCRtpEncodingParameters}} dictionaries.

  11. 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 {{RTCRtpEncodingParameters/active}} set to true.

  12. Let sender have a [[\SendCodecs]] internal slot, representing a list of {{RTCRtpCodecParameters}} dictionaries, and initialized to an empty list.

  13. Let sender have a [[\LastReturnedParameters]] internal slot, which will be used to match {{RTCRtpSender/getParameters}} and {{RTCRtpSender/setParameters}} transactions.

  14. Return sender.

[Exposed=Window]
interface RTCRtpSender {
  readonly attribute MediaStreamTrack? track;
  readonly attribute RTCDtlsTransport? transport;
  static RTCRtpCapabilities? getCapabilities(DOMString kind);
  Promise<void> setParameters(RTCRtpSendParameters parameters);
  RTCRtpSendParameters getParameters();
  Promise<void> replaceTrack(MediaStreamTrack? withTrack);
  void setStreams(MediaStream... streams);
  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 the track's output is disabled, i.e. the track is disabled and/or muted, the {{RTCRtpSender}} MUST send black frames (video) and MUST NOT send (audio). In the case of video, the {{RTCRtpSender}} SHOULD send one black frame per second. If {{track}} is null then the {{RTCRtpSender}} does not send. On getting, the attribute MUST return the value of the [[\SenderTrack]] slot.

transport of type {{RTCDtlsTransport}}, readonly, nullable

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

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

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.

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.[[\Sender]]).
  4. If transceiver.[[\Stopped]] is true, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.
  5. If sender.[[\LastReturnedParameters]] is null, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.
  6. Validate parameters by running the following steps:
    1. Let encodings be parameters.{{RTCRtpSendParameters/encodings}}.
    2. Let codecs be parameters.{{RTCRtpParameters/codecs}}.
    3. Let N be the number of {{RTCRtpEncodingParameters}} stored in sender.[[\SendEncodings]].
    4. If any of the following conditions are met, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidModificationError}}:
      1. encodings.length is different from N.
      2. encodings has been re-ordered.
      3. Any parameter in parameters is marked as a Read-only parameter (such as RID) and has a value that is different from the corresponding parameter value in sender.[[\LastReturnedParameters]]. Note that this also applies to transactionId.
    5. Verify that each {{RTCRtpEncodingParameters/scaleResolutionDownBy}} value in encodings is greater than or equal to 1.0. If one of the {{RTCRtpEncodingParameters/scaleResolutionDownBy}} values does not meet this requirement, return a promise [= rejected =] with a newly [= exception/create | created =] {{RangeError}}.

  7. Let p be a new promise.
  8. In parallel, configure the media stack to use parameters to transmit sender.[[\SenderTrack]].
    1. If the media stack is successfully configured with parameters, queue a task to run the following steps:
      1. Set sender.[[\LastReturnedParameters]] to null.
      2. Set sender.[[\SendEncodings]] to parameters.{{RTCRtpSendParameters/encodings}}.
      3. [= Resolve =] p with undefined.
    2. If any error occurred while configuring the media stack, queue a task to run the following steps:
      1. If an error occurred due to hardware resources not being available, [= reject =] p with a newly created {{RTCError}} whose {{RTCError/errorDetail}} is set to {{RTCErrorDetailType/"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 {{RTCError/errorDetail}} is set to {{RTCErrorDetailType/"hardware-encoder-error"}} and abort these steps.
      3. For all other errors, [= reject =] p with a newly [= exception/create | created =] {{OperationError}}.
  9. Return p.

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

getParameters

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

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

  1. Let sender be the {{RTCRtpSender}} object on which the getter was invoked.

  2. If sender.[[\LastReturnedParameters]] is not null, return sender.[[\LastReturnedParameters]], and abort these steps.

  3. Let result be a new {{RTCRtpSendParameters}} dictionary constructed as follows:

    • {{RTCRtpSendParameters/transactionId}} is set to a new unique identifier.
    • {{RTCRtpSendParameters/encodings}} is set to the value of the [[\SendEncodings]] internal slot.
    • The {{RTCRtpParameters/headerExtensions}} sequence is populated based on the header extensions that have been negotiated for sending.
    • {{RTCRtpParameters/codecs}} is set to the value of the [[\SendCodecs]] internal slot.
    • {{RTCRtpParameters/rtcp}}.{{RTCRtcpParameters/cname}} is set to the CNAME of the associated {{RTCPeerConnection}}. {{RTCRtpParameters/rtcp}}.{{RTCRtcpParameters/reducedSize}} is set to true if reduced-size RTCP has been negotiated for sending, and false otherwise.
  4. Set sender.[[\LastReturnedParameters]] to result.

  5. Queue a task that sets sender.[[\LastReturnedParameters]] to null.

  6. Return result.

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

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

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

replaceTrack

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

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

  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 associated with sender.

  4. Let withTrack be the argument to this method.

  5. If withTrack is non-null and withTrack.kind differs from the transceiver kind of transceiver, return a promise [= rejected =] with a newly [= exception/create | created =] {{TypeError}}.

  6. Return the result of [= chaining =] the following steps to connection's [= operations chain =]:

    1. If transceiver.[[\Stopped]] is true, return a promise [= rejected =] with a newly [= exception/create | created =] {{InvalidStateError}}.

    2. Let p be a new promise.

    3. Let sending be true if transceiver.[[\CurrentDirection]] is {{RTCRtpTransceiverDirection/"sendrecv"}} or {{RTCRtpTransceiverDirection/"sendonly"}}, and false otherwise.

    4. Run the following steps in parallel:

      1. If sending is true, and withTrack is null, have the sender stop sending.

      2. If sending is true, and withTrack is not null, determine if withTrack can be sent immediately by the sender without violating the sender's already-negotiated envelope, and if it cannot, then [= reject =] p with a newly [= exception/create | created =] {{InvalidModificationError}}, and abort these steps.

      3. If sending is true, and withTrack is not null, have the sender switch seamlessly to transmitting withTrack instead of the sender's existing track.

      4. Queue a task that runs the following steps:

        1. If connection.[[\IsClosed]] is true, abort these steps.

        2. Set sender.[[\SenderTrack]] to withTrack.

        3. [= Resolve =] p with undefined.

    5. Return p.

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.
setStreams

Sets the {{MediaStream}}s to be associated with this sender's track.

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

  1. Let sender be the {{RTCRtpSender}} object on which this method was invoked.

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

  3. If connection.[[\IsClosed]] is true, [= exception/throw =] an {{InvalidStateError}}.

  4. Let streams be a list of {{MediaStream}} objects constructed from the method's arguments, or an empty list if the method was called without arguments.

  5. Set sender.[[\AssociatedMediaStreamIds]] to an empty set.

  6. For each stream in streams, add stream.id to [[\AssociatedMediaStreamIds]] if it's not already there.

  7. [= Update the negotiation-needed flag =] for connection.

getStats

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

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

  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.

RTCRtpParameters Dictionary

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

Dictionary {{RTCRtpParameters}} Members

headerExtensions of type sequence<{{RTCRtpHeaderExtensionParameters}}>, required

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

rtcp of type {{RTCRtcpParameters}}, required

Parameters used for RTCP. Read-only parameter.

codecs of type sequence<{{RTCRtpCodecParameters}}>, required

A sequence containing the media codecs that an {{RTCRtpSender}} will choose from, as well as entries for RTX, RED and FEC mechanisms. Corresponding to each media codec where retransmission via RTX is enabled, there will be an entry in {{codecs}} with a {{RTCRtpCodecParameters/mimeType}} attribute indicating retransmission via audio/rtx or video/rtx, and an {{RTCRtpCodecParameters/sdpFmtpLine}} attribute (providing the "apt" and "rtx-time" parameters). Read-only parameter.

RTCRtpSendParameters Dictionary

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

Dictionary {{RTCRtpSendParameters}} Members

transactionId of type DOMString, required

A unique identifier for the last set of parameters applied. Ensures that {{RTCRtpSender/setParameters}} can only be called based on a previous {{RTCRtpSender/getParameters}}, and that there are no intervening changes. [= Read-only parameter =].

encodings of type sequence<{{RTCRtpEncodingParameters}}>, required

A sequence containing parameters for RTP encodings of media.

RTCRtpReceiveParameters Dictionary

dictionary RTCRtpReceiveParameters : RTCRtpParameters {
};

RTCRtpCodingParameters Dictionary

dictionary RTCRtpCodingParameters {
  DOMString rid;
};

Dictionary {{RTCRtpCodingParameters}} Members

rid of type DOMString

If set, this RTP encoding will be sent with the RID header extension as defined by [[!JSEP]]. The RID is not modifiable via {{RTCRtpSender/setParameters}}. It can only be set or modified in {{RTCPeerConnection/addTransceiver}} on the sending side. Read-only parameter.

RTCRtpDecodingParameters Dictionary

dictionary RTCRtpDecodingParameters : RTCRtpCodingParameters {};

RTCRtpEncodingParameters Dictionary

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

Dictionary {{RTCRtpEncodingParameters}} Members

active of type boolean, defaulting to true

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

maxBitrate of type unsigned long

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

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

scaleResolutionDownBy of type double

This member is only present if the sender's kind is "video". The video's resolution will be scaled down in each dimension by the given value before sending. For example, if the value is 2.0, the video will be scaled down by a factor of 2 in each dimension, resulting in sending a video of one quarter the size. If the value is 1.0, the video will not be affected. The value must be greater than or equal to 1.0. By default, the sender will not apply any scaling, (i.e., {{RTCRtpEncodingParameters/scaleResolutionDownBy}} will be 1.0).

RTCRtcpParameters Dictionary

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.

RTCRtpHeaderExtensionParameters Dictionary

dictionary RTCRtpHeaderExtensionParameters {
  required DOMString uri;
  required unsigned short id;
  boolean encrypted = false;
};

Dictionary {{RTCRtpHeaderExtensionParameters}} Members

uri of type DOMString, required

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

id of type unsigned short, required

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

encrypted of type boolean

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

The {{RTCRtpHeaderExtensionParameters}} dictionary enables an application to determine whether a header extension is configured for use within an {{RTCRtpSender}} or {{RTCRtpReceiver}}. For an {{RTCRtpTransceiver}} transceiver, an application can determine the "direction" parameter (defined in Section 5 of [[RFC5285]]) of a header extension as follows without having to parse SDP:

  1. sendonly: The header extension is only included in transceiver.{{RTCRtpTransceiver/sender}}.{{RTCRtpSender/getParameters()}}.{{RTCRtpParameters/headerExtensions}}.
  2. recvonly: The header extension is only included in transceiver.{{RTCRtpTransceiver/receiver}}.{{RTCRtpReceiver/getParameters()}}.{{RTCRtpParameters/headerExtensions}}.
  3. sendrecv: The header extension is included in both transceiver.{{RTCRtpTransceiver/sender}}.{{RTCRtpSender/getParameters()}}.{{RTCRtpParameters/headerExtensions}} and transceiver.{{RTCRtpTransceiver/receiver}}.{{RTCRtpReceiver/getParameters()}}.{{RTCRtpParameters/headerExtensions}}.
  4. inactive: The header extension is included in neither transceiver.{{RTCRtpTransceiver/sender}}.{{RTCRtpSender/getParameters()}}.{{RTCRtpParameters/headerExtensions}} nor transceiver.{{RTCRtpTransceiver/receiver}}.{{RTCRtpReceiver/getParameters()}}.{{RTCRtpParameters/headerExtensions}}.

RTCRtpCodecParameters Dictionary

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

Dictionary {{RTCRtpCodecParameters}} Members

payloadType of type octet, required

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

mimeType of type DOMString, required

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, required

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

channels of type unsigned short

When present, indicates 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]]. For an {{RTCRtpSender}}, these parameters come from the remote description, and for an {{RTCRtpReceiver}}, they come from the local description. Read-only parameter.

RTCRtpCapabilities Dictionary

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

Dictionary {{RTCRtpCapabilities}} Members

codecs of type sequence<{{RTCRtpCodecCapability}}>, required

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 {{RTCRtpCodecCapability/sdpFmtpLine}} not present.

headerExtensions of type sequence<{{RTCRtpHeaderExtensionCapability}}>, required

Supported RTP header extensions.

RTCRtpCodecCapability Dictionary

dictionary RTCRtpCodecCapability {
  required DOMString mimeType;
  required 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, required

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

clockRate of type unsigned long, required

The codec clock rate expressed in Hertz.

channels of type unsigned short

If present, indicates the maximum number of channels (mono=1, stereo=2).

sdpFmtpLine of type DOMString

The "format specific parameters" field from the a=fmtp line in the SDP corresponding to the codec, if one exists.

RTCRtpHeaderExtensionCapability Dictionary

dictionary RTCRtpHeaderExtensionCapability {
  DOMString uri;
};

Dictionary {{RTCRtpHeaderExtensionCapability}} Members

uri of type DOMString

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

RTCRtpReceiver Interface

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

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

  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. Note that the track.id is generated by the user agent and does not map to any track IDs on the remote side.

  3. Initialize track.kind to kind.

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

  5. Initialize track.readyState to live.

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

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

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

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

  10. 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.

  11. Let receiver have a [[\LastStableStateAssociatedRemoteMediaStreams]] internal slot and initialize it to an empty list.

  12. Let receiver have a [[\ReceiveCodecs]] internal slot, representing a list of {{RTCRtpCodecParameters}} dictionaries, and initialized to an empty list.

  13. Let receiver have a [[\LastStableStateReceiveCodecs]] internal slot and initialize it to an empty list.

  14. Return receiver.

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

Attributes

track of type {{MediaStreamTrack}}, readonly

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

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

transport of type {{RTCDtlsTransport}}, readonly, nullable

The {{transport}} attribute is the transport over which media for the receiver's {{RTCRtpReceiver/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.

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.

getParameters

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

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

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

    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.
  • {{RTCRtpParameters/rtcp}}.{{RTCRtcpParameters/reducedSize}} is set to true if the receiver is currently prepared to receive reduced-size RTCP packets, and false otherwise. {{RTCRtpParameters/rtcp}}.{{RTCRtcpParameters/cname}} is left out.
getContributingSources

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

getSynchronizationSources

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

getStats

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

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

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

Even if the {{MediaStreamTrack}} is not attached to any sink for playout, {{RTCRtpReceiver/getSynchronizationSources}} and {{RTCRtpReceiver/getContributingSources}} returns up-to-date information as long as the track is not ended; sinks are not a prerequisite for decoding RTP packets.
As stated in the conformance section, requirements phrased as algorithms may be implemented in any manner so long as the end result is equivalent. So, an implementation does not need to literally queue a task for every frame, as long as the end result is that within a single event loop task execution, all returned {{RTCRtpSynchronizationSource}} and {{RTCRtpContributingSource}} dictionaries for a particular {{RTCRtpReceiver}} contain information from a single point in the RTP stream.
dictionary RTCRtpContributingSource {
  required DOMHighResTimeStamp timestamp;
  required unsigned long source;
  double audioLevel;
  required unsigned long rtpTimestamp;
};

Dictionary RTCRtpContributingSource Members

timestamp of type {{DOMHighResTimeStamp}}, required

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

source of type unsigned long, required

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

audioLevel of type double

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

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

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

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

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

rtpTimestamp of type unsigned long, required

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

dictionary RTCRtpSynchronizationSource : RTCRtpContributingSource {
  boolean voiceActivityFlag;
};

Dictionary RTCRtpSynchronizationSource Members

voiceActivityFlag of type boolean

Only present for audio receivers. Whether the last RTP packet, delivered 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 absent.

RTCRtpTransceiver Interface

The {{RTCRtpTransceiver}} interface represents a combination of an {{RTCRtpSender}} and an {{RTCRtpReceiver}} that share a common [= media stream "identification-tag" =]. As defined in [[!JSEP]], an {{RTCRtpTransceiver}} is said to be associated with a [= media description =] if its {{RTCRtpTransceiver/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 [[\Stopping]] internal slot, initialized to false.

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

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

  7. Let transceiver have a [[\Receptive]] internal slot, initialized to false.

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

  9. Let transceiver have a [[\FiredDirection]] internal slot, initialized to null.

  10. Let transceiver have a [[\PreferredCodecs]] internal slot, initialized to an empty list.

  11. Return transceiver.

Creating a transceiver does not create the underlying {{RTCDtlsTransport}} and {{RTCIceTransport}} objects. This will only occur as part of the process of [= set the RTCSessionDescription | setting an RTCSessionDescription =].
[Exposed=Window]
interface RTCRtpTransceiver {
  readonly attribute DOMString? mid;
  [SameObject] readonly attribute RTCRtpSender sender;
  [SameObject] readonly attribute RTCRtpReceiver receiver;
  attribute RTCRtpTransceiverDirection direction;
  readonly attribute RTCRtpTransceiverDirection? currentDirection;
  void stop();
  void setCodecPreferences(sequence<RTCRtpCodecCapability> codecs);
};

Attributes

mid of type DOMString, readonly, nullable

The {{mid}} attribute is the [= media stream "identification-tag" =] negotiated and present in the local and remote descriptions as defined in [[!JSEP]]. 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.

direction of type {{RTCRtpTransceiverDirection}}

As defined in [[!JSEP]], the direction attribute indicates the preferred direction of this transceiver, which will be used in calls to {{RTCPeerConnection/createOffer}} and {{RTCPeerConnection/createAnswer}}. An update of directionality does not take effect immediately. Instead, future calls to {{RTCPeerConnection/createOffer}} and {{RTCPeerConnection/createAnswer}} mark the corresponding [= media description =] as sendrecv, sendonly, recvonly or inactive as defined in [[!JSEP]]

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

  1. Let transceiver be the {{RTCRtpTransceiver}} object on which the getter is invoked.

  2. If transceiver.[[\Stopping]] is true, return {{RTCRtpTransceiverDirection/"stopped"}}.

  3. Otherwise, return the value of the [[\Direction]] slot.

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

  1. Let transceiver be the {{RTCRtpTransceiver}} object on which the setter is invoked.

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

  3. If transceiver.[[\Stopping]] is true, [= exception/throw =] an {{InvalidStateError}}.

  4. Let newDirection be the argument to the setter.

  5. If newDirection is equal to transceiver.[[\Direction]], abort these steps.

  6. If newDirection is equal to {{RTCRtpTransceiverDirection/"stopped"}}, [= exception/throw =] a {{TypeError}}.

  7. Set transceiver.[[\Direction]] to newDirection.

  8. Update the negotiation-needed flag for connection.

currentDirection of type {{RTCRtpTransceiverDirection}}, readonly, nullable

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

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

  1. Let transceiver be the {{RTCRtpTransceiver}} object on which the getter is invoked.

  2. If transceiver.[[\Stopped]] is true, return {{RTCRtpTransceiverDirection/"stopped"}}.

  3. Otherwise, return the value of the [[\CurrentDirection]] slot.

Methods

stop

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

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

A stopped transceiver will cause future calls to {{RTCPeerConnection/createOffer}} or {{RTCPeerConnection/createAnswer}} to generate a zero port in the [= media description =] for the corresponding transceiver, as defined in [[!JSEP]].

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

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

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

  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.[[\IsClosed]] is true, [= exception/throw =] an {{InvalidStateError}}.

  4. If transceiver.[[\Stopping]] is true, abort these steps.

  5. [= Stop sending and receiving =] given transceiver, and update the negotiation-needed flag for connection.

The stop sending and receiving algorithm given a transceiver, is as follows:

  1. Let sender be transceiver.[[\Sender]].

  2. Let receiver be transceiver.[[\Receiver]].

  3. Stop sending media with sender.

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

  5. Stop receiving media with receiver.

  6. Execute the steps for receiver.[[\ReceiverTrack]] to be ended.

  7. Set transceiver.[[\Direction]] to {{RTCRtpTransceiverDirection/"inactive"}}.

  8. Set transceiver.[[\Stopping]] to true.

The stop the RTCRtpTransceiver algorithm given a transceiver, is as follows:

  1. If transceiver.[[\Stopping]] is false, [= stop sending and receiving =] given transceiver.

  2. Set transceiver.[[\Stopped]] to true.

  3. Set transceiver.[[\Receptive]] to false.

  4. Set transceiver.[[\CurrentDirection]] to null.

setCodecPreferences

The {{setCodecPreferences}} method overrides the default codec preferences used by the user agent. When generating a session description using either {{RTCPeerConnection/createOffer}} or {{RTCPeerConnection/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}}.

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

Codec preferences remain in effect for all calls to {{RTCPeerConnection/createOffer}} and {{RTCPeerConnection/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}}.{{RTCRtpSender/getCapabilities}}(kind) or {{RTCRtpReceiver}}.{{RTCRtpReceiver/getCapabilities}}(kind), where kind is the kind of the {{RTCRtpTransceiver}} on which the method is called. Additionally, the {{RTCRtpCodecCapability}} dictionary members cannot be modified. If codecs does not fulfill these requirements, the user agent MUST [= exception/throw =] an {{InvalidModificationError}}.

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

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

  1. Let transceiver be the {{RTCRtpTransceiver}} object this method was invoked on.

  2. Let codecs be the first argument.

  3. If codecs is an empty list, set transceiver.[[\PreferredCodecs]] to codecs and abort these steps.

  4. Remove any duplicate values in codecs. Start at the back of the list such that the priority of the codecs is maintained; the index of the first occurrence of a codec within the list is the same before and after this step.

  5. Let kind be the transceiver's [= transceiver kind =].

  6. If the intersection between codecs and {{RTCRtpSender}}.{{RTCRtpSender/getCapabilities}}(kind).{{RTCRtpParameters/codecs}} or the intersection between codecs and {{RTCRtpReceiver}}.{{RTCRtpReceiver/getCapabilities}}(kind).{{RTCRtpParameters/codecs}} only contains RTX, RED or FEC codecs or is an empty set, throw {{InvalidModificationError}}. This ensures that we always have something to offer, regardless of transceiver.{{RTCRtpTransceiver/direction}}.

  7. Let codecCapabilities be the union of {{RTCRtpSender}}.{{RTCRtpSender/getCapabilities}}(kind).{{RTCRtpParameters/codecs}} and {{RTCRtpReceiver}}.{{RTCRtpReceiver/getCapabilities}}(kind).{{RTCRtpParameters/codecs}}.

  8. For each codec in codecs,

    1. If codec is not in codecCapabilities, throw {{InvalidModificationError}}.
  9. Set transceiver.[[\PreferredCodecs]] to codecs.

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

Simulcast functionality

Simulcast functionality is provided via the {{RTCPeerConnection/addTransceiver}} method of the {{RTCPeerConnection}} object and the {{RTCRtpSender/setParameters}} method of the {{RTCRtpSender}} object.

The {{RTCPeerConnection/addTransceiver}} method establishes the simulcast envelope which includes the maximum number of simulcast streams that can be sent, as well as the ordering of the {{RTCRtpSendParameters/encodings}}. While characteristics of individual simulcast streams can be modified using the {{RTCRtpSender/setParameters}} method, the [= simulcast envelope =] cannot be changed. One of the implications of this model is that the {{RTCPeerConnection/addTrack()}} method cannot provide simulcast functionality since it does not take {{RTCRtpTransceiverInit/sendEncodings}} as an argument, and therefore cannot configure an {{RTCRtpTransceiver}} to send simulcast.

Another implication is that the answerer cannot set the [= simulcast envelope =] directly. Upon calling the {{RTCPeerConnection/setRemoteDescription}} method of the {{RTCPeerConnection}} object, the [= simulcast envelope =] is configured on the {{RTCRtpTransceiver}} to contain the layers described by the specified {{RTCSessionDescription}}. Once the envelope is determined, layers cannot be removed. They can be marked as inactive by setting the {{RTCRtpEncodingParameters/active}} member to false effectively disabling the layer.

While {{RTCRtpSender/setParameters}} cannot modify the [= simulcast envelope =], it is still possible to control the number of streams that are sent and the characteristics of those streams. Using {{RTCRtpSender/setParameters}}, simulcast streams can be made inactive by setting the {{RTCRtpEncodingParameters/active}} member to false, or can be reactivated by setting the {{RTCRtpEncodingParameters/active}} member to true. Using {{RTCRtpSender/setParameters}}, stream characteristics can be changed by modifying attributes such as {{RTCRtpEncodingParameters/maxBitrate}}.

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

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

This specification does not define how to configure {{RTCPeerConnection/createOffer}} to receive multiple RTP encodings. However when {{RTCPeerConnection/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 {{RTCRtpTransceiver/receiver}}.{{RTCRtpReceiver/getParameters()}} will reflect the encodings negotiated.

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

Encoding Parameter Examples

Examples of simulcast scenarios implemented with encoding parameters:


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

"Hold" functionality

Together, the {{RTCRtpTransceiver/direction}} attribute and the {{RTCRtpSender/replaceTrack}} method enable developers to implement "hold" scenarios.

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

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

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

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

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

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

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

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

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 {{RTCPeerConnection/setLocalDescription()}} and {{RTCPeerConnection/setRemoteDescription()}}. Each {{RTCDtlsTransport}} object represents the DTLS transport layer for the RTP or RTCP {{RTCIceTransport/component}} of a specific {{RTCRtpTransceiver}}, or a group of {{RTCRtpTransceiver}}s if such a group has been negotiated via [[!BUNDLE]].

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

An {{RTCDtlsTransport}} has a [[\DtlsTransportState]] internal slot initialized to {{RTCDtlsTransportState/"new"}} and a [[\RemoteCertificates]] slot initialized to an empty list.

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

  1. Let transport be the {{RTCDtlsTransport}} object to receive the state update and error notification.

  2. If the state of transport is already {{RTCDtlsTransportState/"failed"}}, abort these steps.

  3. Set transport.[[\DtlsTransportState]] to {{RTCDtlsTransportState/"failed"}}.

  4. [= Fire an event =] named error using the {{RTCErrorEvent}} interface with its errorDetail attribute set to either {{RTCErrorDetailType/"dtls-failure"}} or {{RTCErrorDetailType/"fingerprint-failure"}}, as appropriate, and other fields set as described under the {{RTCErrorDetailType}} enum description, at transport.

  5. [= Fire an event =] named statechange at transport.

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

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

  2. Let newState be the new state.

  3. Set transport.[[\DtlsTransportState]] to newState.

  4. If newState is {{RTCDtlsTransportState/connected}} then let newRemoteCertificates be the certificate chain in use by the remote side, with each certificate encoded in binary Distinguished Encoding Rules (DER) [[!X690]], and set transport.[[\RemoteCertificates]] to newRemoteCertificates.

  5. [= Fire an event =] named statechange at transport.

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

Attributes

iceTransport of type {{RTCIceTransport}}, readonly

The {{iceTransport}} attribute is the underlying transport that is used to send and receive packets. The underlying transport may not be shared between multiple active {{RTCDtlsTransport}} objects.

state of type {{RTCDtlsTransportState}}, readonly

The {{state}} attribute MUST, on getting, return the value of the [[\DtlsTransportState]] slot.

onstatechange of type EventHandler
The event type of this event handler is statechange.
onerror of type EventHandler
The event type of this event handler is {{error}}.

Methods

getRemoteCertificates

Returns the value of [[\RemoteCertificates]].

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 and verifying the remote fingerprint.
connected DTLS has completed negotiation of a secure connection and verified the remote fingerprint.
closed The transport has been closed intentionally as the result of receipt of a close_notify alert, or calling {{RTCPeerConnection/close()}}.
failed The transport has failed as the result of an error (such as receipt of an error alert or failure to validate the remote fingerprint).

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 [[!IANA-HASH-FUNCTION]].

value of type DOMString

The value of the certificate fingerprint in lowercase hex string as expressed utilizing the syntax of 'fingerprint' in [[!RFC4572]] Section 5.

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 {{RTCPeerConnection/setLocalDescription()}} and {{RTCPeerConnection/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 {{RTCIceTransport/component}} of a specific {{RTCRtpTransceiver}}, or a group of {{RTCRtpTransceiver}}s if such a group has been negotiated via [[!BUNDLE]].

An ICE restart for an existing {{RTCRtpTransceiver}} will be represented by an existing {{RTCIceTransport}} object, whose {{RTCIceTransport/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.[[\IsClosed]] is true, abort these steps.

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

  4. Set transport.[[\IceGathererState]] to {{RTCIceGathererState/gathering}}.

  5. [= Fire an event =] named {{gatheringstatechange}} at transport.

  6. Update the ICE gathering state of connection.

When the [= ICE Agent =] is finished gathering a [= generation =] of candidates for an {{RTCIceTransport}}, and those candidates have been surfaced to the application, 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.[[\IsClosed]] is true, abort these steps.

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

  4. Let newCandidate be the result of [= creating an RTCIceCandidate =] with a new dictionary whose {{RTCIceCandidateInit/sdpMid}} and {{RTCIceCandidateInit/sdpMLineIndex}} are set to the values associated with this {{RTCIceTransport}}, {{RTCIceCandidateInit/usernameFragment}} is set to the username fragment of the [= generation =] of candidates for which gathering finished, and {{RTCIceCandidateInit/candidate}} is set to an empty string.

  5. [= Fire an event =] named {{icecandidate}} using the {{RTCPeerConnectionIceEvent}} interface with the candidate attribute set to newCandidate at connection.

  6. If another [= generation =] of candidates is still being gathered, abort these steps.

    This may occur if an ICE restart is initiated while the ICE agent is still gathering the previous [= generation =] of candidates.
  7. Set transport.[[\IceGathererState]] to {{RTCIceGathererState/complete}}.

  8. [= Fire an 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 size | ICE candidate pool =] or gathering it from scratch, the user agent MUST queue a task that runs the following steps:

  1. Let candidate be the available ICE candidate.

  2. Let connection be the {{RTCPeerConnection}} object associated with this [= ICE Agent =].

  3. If connection.[[\IsClosed]] is true, abort these steps.

  4. If either connection.[[\PendingLocalDescription]] or connection.[[\CurrentLocalDescription]] are not null, and represent the ICE [= generation =] for which candidate was gathered, [= surface the candidate =] with candidate and connection, and abort these steps.

  5. Otherwise, append candidate to connection.[[\EarlyCandidates]].

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

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

  1. For each candidate, candidate, in connection.[[\EarlyCandidates]], queue a task to [= surface the candidate =] with candidate and connection.

  2. Set connection.[[\EarlyCandidates]] to an empty list.

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

  1. If connection.[[\IsClosed]] is true, abort these steps.

  2. Let transport be the {{RTCIceTransport}} for which candidate is being made available.

  3. If connection.[[\PendingLocalDescription]] is not null, and represents the ICE [= generation =] for which candidate was gathered, add candidate to connection.[[\PendingLocalDescription]].sdp.

  4. If connection.[[\CurrentLocalDescription]] is not null, and represents the ICE [= generation =] for which candidate was gathered, add candidate to connection.[[\CurrentLocalDescription]].sdp.

  5. Let newCandidate be the result of [= creating an RTCIceCandidate =] with a new dictionary whose {{RTCIceCandidateInit/sdpMid}} and {{RTCIceCandidateInit/sdpMLineIndex}} are set to the values associated with this {{RTCIceTransport}}, {{RTCIceCandidateInit/usernameFragment}} is set to the username fragment of the candidate, and {{RTCIceCandidateInit/candidate}} is set to a string encoded using the [= candidate-attribute =] grammar to represent candidate.

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

  7. [= Fire an event =] named {{icecandidate}} using the {{RTCPeerConnectionIceEvent}} interface with the candidate attribute set to newCandidate at connection.

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

When the [= ICE Agent =] indicates that an {{RTCIceTransport}} has changed either the selected candidate pair, the {{RTCIceTransportState}} or both, the user agent MUST queue a task that runs the following steps:

  1. Let connection be the {{RTCPeerConnection}} object associated with this [= ICE Agent =].

  2. If connection.[[\IsClosed]] is true, abort these steps.

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

  4. Let selectedCandidatePairChanged be false.

  5. Let transportIceConnectionStateChanged be false.

  6. Let connectionIceConnectionStateChanged be false.

  7. Let connectionStateChanged be false.

  8. If transport's selected candidate pair was changed, run the following steps:

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

    2. Set transport.[[\SelectedCandidatePair]] to newCandidatePair.

    3. Set selectedCandidatePairChanged to true.

  9. If transport's {{RTCIceTransportState}} was changed, run the following steps:

    1. Set transport.[[\IceTransportState]] to the new indicated {{RTCIceTransportState}}.

    2. Set transportIceConnectionStateChanged to true.

    3. Set connection's [= ICE connection state =] to the value of deriving a new state value as described by the {{RTCIceConnectionState}} enum.

    4. If the ice connection state changed in the previous step, set connectionIceConnectionStateChanged to true.

    5. Set connection's [= connection state =] to the value of deriving a new state value as described by the {{RTCPeerConnectionState}} enum.

    6. If the [= connection state =] changed in the previous step, set connectionStateChanged to true.

  10. If selectedCandidatePairChanged is true, [= fire an event =] named {{selectedcandidatepairchange}} at transport.

  11. If transportIceConnectionStateChanged is true, [= fire an event =] named {{statechange}} at transport.

  12. If connectionIceConnectionStateChanged is true, [= fire an event =] named {{iceconnectionstatechange}} at connection.

  13. If connectionStateChanged is true, [= fire an event =] named {{connectionstatechange}} at connection.

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, on getting, return the value of the [[\IceRole]] internal slot.

component of type {{RTCIceComponent}}, readonly

The {{component}} attribute MUST return the ICE component of the transport. When RTCP mux is used, a single {{RTCIceTransport}} transports both RTP and RTCP and {{component}} is set to {{RTCIceComponent/"rtp"}}.

state of type {{RTCIceTransportState}}, readonly

The {{state}} attribute MUST, on getting, return the value of the [[\IceTransportState]] slot.

gatheringState of type {{RTCIceGathererState}}, readonly

The {{gatheringState}} attribute MUST, on getting, return the value of the [[\IceGathererState]] slot.

onstatechange of type EventHandler
This event handler, of event handler event type {{statechange}}, MUST be fired any time the {{RTCIceTransport}} {{RTCIceTransport/state}} changes.
ongatheringstatechange of type EventHandler
This event handler, of event handler event type {{gatheringstatechange}}, MUST be fired any time the {{RTCIceTransport}} [= ICE gathering 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 {{RTCPeerConnection/onicecandidate}}.

getRemoteCandidates

Returns a sequence describing the remote ICE candidates received by this {{RTCIceTransport}} via {{RTCPeerConnection/addIceCandidate()}}.

{{getRemoteCandidates}} will not expose peer reflexive candidates since they are not received via {{RTCPeerConnection/addIceCandidate()}}.
getSelectedCandidatePair

Returns the selected candidate pair on which packets are sent. This method MUST return the value of the [[\SelectedCandidatePair]] slot. When {{RTCIceTransport}}.{{RTCIceTransport/state}} is {{RTCIceTransportState/"new"}} or {{RTCIceTransportState/"closed"}} {{getSelectedCandidatePair}} returns null.

getLocalParameters

Returns the local ICE parameters received by this {{RTCIceTransport}} via {{RTCPeerConnection/setLocalDescription}}, or null if the parameters have not yet been received.

getRemoteParameters

Returns the remote ICE parameters received by this {{RTCIceTransport}} via {{RTCPeerConnection/setRemoteDescription}} or null if the parameters have not yet been received.

RTCIceParameters Dictionary

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.

RTCIceCandidatePair Dictionary

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"
};
{{RTCIceGathererState}} 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",
  "disconnected",
  "failed",
  "closed"
};
{{RTCIceTransportState}} 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 {{RTCIceTransportState/"checking"}} (if there are candidate pairs remaining to be checked) or {{RTCIceTransportState/"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 {{RTCIceTransportState/"failed"}}.
disconnected The [= ICE Agent =] has determined that connectivity is currently lost for this {{RTCIceTransport}}. This is a transient state that may trigger intermittently (and resolve itself without action) on a flaky network. The way this state is determined is implementation dependent. Examples include:
  • Losing the network interface for the connection in use.
  • Repeatedly failing to receive a response to STUN requests.
Alternatively, the {{RTCIceTransport}} has finished checking all existing candidates pairs and not found a connection (or consent checks [[!RFC7675]] once successful, have now failed), but it is still gathering and/or waiting for additional remote candidates.
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. This is a terminal state until ICE is restarted. Since an ICE restart may cause connectivity to resume, entering the {{RTCIceTransportState/"failed"}} state does not cause DTLS transports, SCTP associations or the data channels that run over them to close, or tracks to mute.
closed The {{RTCIceTransport}} has shut down and is no longer responding to STUN requests.

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

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

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

Some example state transitions are:

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

RTCIceRole Enum

enum RTCIceRole {
  "unknown",
  "controlling",
  "controlled"
};
{{RTCIceRole}} Enumeration description
unknown An agent whose role as defined by [[!ICE]], Section 3, has not yet been determined.
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"
};
{{RTCIceComponent}} 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 =].

RTCTrackEvent

The {{track}} event uses the {{RTCTrackEvent}} interface.

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

Constructors

RTCTrackEvent.constructor()

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 {{MediaStream}}s 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}} member represents the {{RTCRtpReceiver}} object associated with the event.

track of type {{MediaStreamTrack}}, required

The {{track}} member represents the {{MediaStreamTrack}} object that is associated with the {{RTCRtpReceiver}} identified by {{RTCTrackEventInit/receiver}}.

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

The {{streams}} member is an array of {{MediaStream}} objects representing the {{MediaStream}}s 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.

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 Web Sockets.

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.[[\IsClosed]] is true, [= exception/throw =] an {{InvalidStateError}}.

  3. [= Create an RTCDataChannel =], channel.

  4. Initialize channel.[[\DataChannelLabel]] to the value of the first argument.

  5. If the UTF-8 representation of [[\DataChannelLabel]] is longer than 65535 bytes, [= exception/throw =] a {{TypeError}}.

  6. Let options be the second argument.

  7. Initialize channel.[[\MaxPacketLifeTime]] to option.{{RTCDataChannelInit/maxPacketLifeTime}}, if present, otherwise null.

  8. Initialize channel.[[\MaxRetransmits]] to option.{{RTCDataChannelInit/maxRetransmits}}, if present, otherwise null.

  9. Initialize channel.[[\Ordered]] to option.{{RTCDataChannelInit/ordered}}.

  10. Initialize channel.[[\DataChannelProtocol]] to option.{{RTCDataChannelInit/protocol}}.

  11. If the UTF-8 representation of [[\DataChannelProtocol]] is longer than 65535 bytes, [= exception/throw =] a {{TypeError}}.

  12. Initialize channel.[[\Negotiated]] to option.{{RTCDataChannelInit/negotiated}}.

  13. Initialize channel.[[\DataChannelId]] to the value of option.{{RTCDataChannelInit/id}}, if it is present and [[\Negotiated]] is true, otherwise null.

    This means the {{RTCDataChannelInit/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]].
  14. If [[\Negotiated]] is true and [[\DataChannelId]] is null, [= exception/throw =] a {{TypeError}}.

  15. If both [[\MaxPacketLifeTime]] and [[\MaxRetransmits]] attributes are set (not null), [= exception/throw =] a {{TypeError}}.

  16. 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.

  17. If [[\DataChannelId]] is equal to 65535, which is greater than the maximum allowed ID of 65534 but still qualifies as an unsigned short, [= exception/throw =] a {{TypeError}}.

  18. 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}}, [= exception/throw =] an {{OperationError}} exception.

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

    If the [[\DataChannelId]] slot is not null, transport is in the {{RTCSctpTransportState/"connected"}} state and [[\DataChannelId]] is greater or equal to transport.[[\MaxChannels]], [= exception/throw =] an {{OperationError}}.

  20. If channel is the first {{RTCDataChannel}} created on connection, [= update the negotiation-needed flag =] for connection.

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

  22. Create channel's associated [= underlying data transport =] and configure it according to the relevant properties of channel.

RTCSctpTransport Interface

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

Create an instance

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

  1. Let transport be a new {{RTCSctpTransport}} object.

  2. Let transport have a [[\SctpTransportState]] internal slot initialized to initialState.

  3. Let transport have a [[\MaxMessageSize]] internal slot and run the steps labeled [= update the data max message size =] to initialize it.

  4. Let transport have a [[\MaxChannels]] internal slot initialized to null.

  5. Return transport.

Update max message size

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

  1. Let transport be the {{RTCSctpTransport}} object to be updated.

  2. Let remoteMaxMessageSize be the value of the max-message-size SDP attribute read from the remote description, as described in [[!SCTP-SDP]] (section 6), or 65536 if the attribute is missing.

  3. Let canSendSize be the number of bytes that this client can send (i.e. the size of the local send buffer) or 0 if the implementation can handle messages of any size.

  4. If both remoteMaxMessageSize and canSendSize are 0, set [[\MaxMessageSize]] to the positive Infinity value.

  5. Else, if either remoteMaxMessageSize or canSendSize is 0, set [[\MaxMessageSize]] to the larger of the two.

  6. Else, set [[\MaxMessageSize]] to the smaller of remoteMaxMessageSize or canSendSize.

Connected procedure

Once an SCTP transport is connected, meaning the SCTP association of an {{ RTCSctpTransport}} has been established, run the following steps:

  1. Let transport be the {{RTCSctpTransport}} object.

  2. Let connection be the {{RTCPeerConnection}} object associated with transport.

  3. Set [[\MaxChannels]] to the minimum of the negotiated amount of incoming and outgoing SCTP streams.

  4. For each of connection's {{RTCDataChannel}}:

    1. Let channel be the {{RTCDataChannel}} object.

    2. If channel.[[\DataChannelId]] is null, initialize [[\DataChannelId]] to the value generated by the underlying sctp data channel, according to [[!RTCWEB-DATA-PROTOCOL]].

    3. If channel.[[\DataChannelId]] is greater or equal to transport.[[\MaxChannels]], or the previous step failed to assign an id, [= unable to create an RTCDataChannel | close =] the channel due to a failure. Otherwise, [= announce the rtcdatachannel as open | announce the channel as open =].

  5. [= Fire an event =] named statechange at transport.

    This event is fired before the open events fired by [= announce the rtcdatachannel as open | announcing the channel as open =]; the open events are fired from a queued task.

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

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, this attribute MUST return the value of the [[\SctpTransportState]] slot.

maxMessageSize of type unrestricted double, readonly

The maximum size of data that can be passed to {{RTCDataChannel}}'s {{RTCDataChannel/send()}} method. The attribute MUST, on getting, return the value of the [[\MaxMessageSize]] slot.

maxChannels of type unsigned short , readonly, nullable

The maximum amount of {{RTCDataChannel}}'s that can be used simultaneously. The attribute MUST, on getting, return the value of the [[\MaxChannels]] slot.

This attribute's value will be null until the SCTP transport goes into the {{RTCSctpTransportState/"connected"}} state.
onstatechange of type EventHandler

The event type of this event handler is statechange.

RTCSctpTransportState Enum

{{RTCSctpTransportState}} indicates the state of the SCTP transport.

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

The {{RTCSctpTransport}} is in the process of negotiating an association. This is the initial state of the [[\SctpTransportState]] slot when an {{RTCSctpTransport}} is created.

connected

When the negotiation of an association is completed, a task is queued to update the [[\SctpTransportState]] slot to {{RTCSctpTransportState/"connected"}}.

closed

A task is queued to update the [[\SctpTransportState]] slot to {{RTCSctpTransportState/"closed"}} when:

  • a SHUTDOWN or ABORT chunk is received.
  • the SCTP association has been closed intentionally, such as by closing the peer connection or applying a remote description that rejects data or changes the SCTP port.
  • the underlying DTLS association has transitioned to {{RTCDtlsTransportState/"closed"}} state.

Note that the last transition is logical due to the fact that an SCTP association requires an established DTLS connection - [[RFC8261]] section 6.1 specifies that SCTP over DTLS is single-homed - and that no way of of switching to an alternate transport is defined in this API.

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 {{RTCDataChannelInit/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 {{RTCDataChannelInit/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 {{RTCDataChannelInit/negotiated}} {{RTCDataChannelInit}} dictionary member set to true and the same {{RTCDataChannel/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 {{RTCDataChannelInit/id}}s.

Each {{RTCDataChannel}} has an associated underlying data transport that is used to transport actual data to the other peer. In the case of SCTP data channels utilizing an {{RTCSctpTransport}} (which represents the state of the SCTP association), the underlying data transport is the SCTP stream pair. The transport properties of the [= underlying data transport =], such as in order delivery settings and reliability mode, are configured by the peer as the channel is created. The properties of a channel cannot change after the channel has been created. The actual wire protocol between the peers is specified by the WebRTC DataChannel Protocol specification [[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 ( {{RTCDataChannelInit/maxRetransmits}} ) or set a time during which transmissions (including retransmissions) are allowed ( {{RTCDataChannelInit/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 {{RTCPeerConnection/createDataChannel}} or dispatched via an {{RTCDataChannelEvent}}, MUST initially be in the {{RTCDataChannelState/"connecting"}} state. When the {{RTCDataChannel}} object's [= underlying data transport =] is ready, the user agent MUST [= announce the RTCDataChannel as open =].

Creating a data channel

To create an {{RTCDataChannel}}, run the following steps:

  1. Let channel be a newly created {{RTCDataChannel}} object.

  2. Let channel have a [[\ReadyState]] internal slot initialized to {{RTCDataChannelState/"connecting"}}.

  3. Let channel have a [[\BufferedAmount]] internal slot initialized to 0.

  4. Let channel have internal slots named [[\DataChannelLabel]], [[\Ordered]], [[\MaxPacketLifeTime]], [[\MaxRetransmits]], [[\DataChannelProtocol]], [[\Negotiated]], and [[\DataChannelId]].

  5. Return channel.

Announcing a data channel 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. If channel.[[\ReadyState]] is {{RTCDataChannelState/"closing"}} or {{RTCDataChannelState/"closed"}}, abort these steps.

  4. Set channel.[[\ReadyState]] to {{RTCDataChannelState/"open"}}.

  5. [= Fire an event =] named {{open}} at channel.

Announcing a data channel instance

When an [= underlying data transport =] is to be announced (the other peer created a channel with {{RTCDataChannelInit/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. Let connection be the {{RTCPeerConnection}} object associated with the [= underlying data transport =].

  2. If connection.[[\IsClosed]] is true, abort these steps.

  3. [= Create an RTCDataChannel =], channel.

  4. 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]].

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

  6. Initialize channel.[[\Negotiated]] to false.

  7. Set channel.[[\ReadyState]] to {{RTCDataChannelState/"open"}} (but do not fire the {{open}} event, yet).

    This allows to start sending messages inside of the {{datachannel}} event handler prior to the {{open}} event being fired.
  8. [= Fire an event =] named {{datachannel}} using the {{RTCDataChannelEvent}} interface with the {{RTCDataChannelEvent/channel}} attribute set to channel at connection.

  9. [= announce the rtcdatachannel as open | Announce the data channel as open =].

Closing procedure

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

  1. Let channel be the {{RTCDataChannel}} object whose [= underlying data transport =] was closed.

  2. Unless the procedure was initiated by channel.{{RTCDataChannel/close}}, set channel.[[\ReadyState]] to {{RTCDataChannelState/"closing"}}.

  3. Run the following steps in parallel:

    1. Finish sending all currently pending messages of the channel.

    2. Follow the closing procedure defined for the channel's [= underlying data transport =] :

      1. In the case of an SCTP-based [= underlying data transport | transport =], follow [[!RTCWEB-DATA]], section 6.7.

    3. Render the channel's [= data transport =] {{closed}} by following the associated procedure.

Announcing a data channel as 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 [= underlying data transport =] was closed.

  2. Set channel.[[\ReadyState]] to {{RTCDataChannelState/"closed"}}.

  3. If the [= underlying data transport | transport =] was closed with an error, [= fire an event =] named {{error}} using the {{RTCErrorEvent}} interface with its {{RTCError/errorDetail}} attribute set to {{RTCErrorDetailType/"sctp-failure"}} at channel.

  4. [= Fire an event =] named {{close}} at channel.

Error on creating data channels

In some cases, the user agent may be unable to create an {{RTCDataChannel}} 's [= underlying data transport =]. For example, the data channel's {{RTCDataChannel/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.[[\ReadyState]] to {{RTCDataChannelState/"closed"}}.

  3. [= Fire an event =] named {{RTCDataChannel/error}} using the {{RTCErrorEvent}} interface with the {{RTCError/errorDetail}} attribute set to {{RTCErrorDetailType/"data-channel-failure"}} at channel.

  4. [= Fire an event =] named {{close}} at channel.

Receiving messages on a data channel

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

  1. Let channel be the {{RTCDataChannel}} object for which the user agent has received a message.

  2. Let connection be the {{RTCPeerConnection}} object associated with channel.

  3. If channel.[[\ReadyState]] is not {{RTCDataChannelState/"open"}}, abort these steps and discard rawData.

  4. Execute the sub step by switching on type and channel.{{RTCDataChannel/binaryType}}:

    • If type indicates that rawData is a string:

      Let data be a DOMString that represents the result of decoding rawData as UTF-8.

    • If type indicates that rawData is binary and {{RTCDataChannel/binaryType}} is "blob":

      Let data be a new {{Blob}} object containing rawData as its raw data source.

    • If type indicates that rawData is binary and {{RTCDataChannel/binaryType}} is "arraybuffer":

      Let data be a new {{ArrayBuffer}} object containing rawData as its raw data source.

  5. [= Fire an event =] named {{message}} using the {{MessageEvent}} interface with its origin attribute initialized to the serialization of an origin of connection.[[\DocumentOrigin]], and the data attribute initialized to data at channel.

[Exposed=Window]
interface RTCDataChannel : EventTarget {
  readonly attribute USVString label;
  readonly attribute boolean ordered;
  readonly attribute unsigned short? maxPacketLifeTime;
  readonly attribute unsigned short? maxRetransmits;
  readonly attribute USVString protocol;
  readonly attribute boolean negotiated;
  readonly attribute unsigned short? id;
  readonly attribute RTCDataChannelState readyState;
  readonly attribute unsigned long bufferedAmount;
  [EnforceRange] attribute unsigned long bufferedAmountLowThreshold;
  attribute EventHandler onopen;
  attribute EventHandler onbufferedamountlow;
  attribute EventHandler onerror;
  attribute EventHandler onclosing;
  attribute EventHandler onclose;
  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 out of order delivery is allowed. On getting, the attribute MUST return the value of the [[\Ordered]] slot.

maxPacketLifeTime of type unsigned short, readonly, nullable

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

maxRetransmits of type unsigned short, readonly, nullable

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

protocol of type USVString, readonly

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

negotiated of type boolean, readonly

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

id of type unsigned short, readonly, nullable

The {{id}} attribute returns the ID for this {{RTCDataChannel}}. The value is initially null, which is what will be returned if the ID was not provided at channel creation time, and the DTLS role of the SCTP transport has not yet been negotiated. Otherwise, it will return the ID that was either selected by the script or generated by the user agent according to [[!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.

readyState of type {{RTCDataChannelState}}, readonly

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

bufferedAmount of type unsigned long, readonly

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

bufferedAmountLowThreshold of type unsigned long

The {{bufferedAmountLowThreshold}} attribute sets the threshold at which the {{RTCDataChannel/bufferedAmount}} is considered to be low. When the {{RTCDataChannel/bufferedAmount}} decreases from above this threshold to equal or below it, the {{bufferedamountlow}} event fires. The {{RTCDataChannel/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}}. {{RTCError/errorDetail}} contains "sctp-failure", {{RTCError/sctpCauseCode}} contains the SCTP Cause Code value, and {{DOMException/message}} contains the SCTP Cause-Specific-Information, possibly with additional text.

onclosing of type EventHandler

The event type of this event handler is {{Event}}.

onclose of type EventHandler

The event type of this event handler is {{Event}}.

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, [= exception/throw =] a {{SyntaxError}}. When an {{RTCDataChannel}} object is created, the {{RTCDataChannel/binaryType}} attribute MUST be initialized to the string "blob".

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

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.[[\ReadyState]] is {{RTCDataChannelState/"closing"}} or {{RTCDataChannelState/"closed"}}, then abort these steps.

  3. Set channel.[[\ReadyState]] to {{RTCDataChannelState/"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.

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.[[\ReadyState]] is not {{RTCDataChannelState/"open"}}, [= exception/throw =] an {{InvalidStateError}}.

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

    • string object:

      Let data be a byte buffer that represents the result of encoding the method's argument as UTF-8.

    • {{Blob}} object:

      Let data be the raw data represented by the {{Blob}} object.

      Although the actual retrieval of data from a {{Blob}} object can happen asynchronously, the user agent will make sure to queue the data on the channel's [= underlying data transport =] in the same order as the send method is called. The byte size of data needs to be known synchronously.
    • {{ArrayBuffer}} object:

      Let data be the data stored in the buffer described by the {{ArrayBuffer}} object.

    • {{ArrayBufferView}} object:

      Let data be the data stored in the section of the buffer described by the {{ArrayBuffer}} object that the {{ArrayBufferView}} object references.

    Any data argument type this method has not been overloaded with will result in a {{TypeError}}. This includes null and undefined.
  4. If the byte size of data exceeds the value of {{RTCSctpTransport/maxMessageSize}} on channel's associated {{RTCSctpTransport}}, [= exception/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, [= exception/throw =] an {{OperationError}}.

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

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

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 {{RTCDataChannel/id}} at the other peer.

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

Sets the channel ID when {{RTCDataChannelInit/negotiated}} is true. Ignored when {{RTCDataChannelInit/negotiated}} is false.

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, whether created with {{RTCPeerConnection/createDataChannel}}, or dispatched as a part of an {{RTCDataChannelEvent}}.

open

The [= underlying data transport =] is established and communication is possible.

closing

The [= closing procedure | procedure =] to close down the [= underlying data transport =] has started.

closed

The [= underlying data transport =] has been {{closed}} or could not be established.

RTCDataChannelEvent

The {{datachannel}} event uses the {{RTCDataChannelEvent}} interface.

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

Constructors

RTCDataChannelEvent.constructor()

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.

Garbage Collection

An {{RTCDataChannel}} object MUST not be garbage collected if its

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]].

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

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

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 [[\Duration]] internal slot.

  3. Let dtmf have a [[\InterToneGap]] internal slot.

  4. 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}} dtmfSender is capable of sending DTMF. On getting, the user agent MUST return the result of running [= determine if DTMF can be sent =] for dtmfSender.

toneBuffer of type DOMString, readonly

The {{toneBuffer}} attribute MUST return a list of the tones remaining to be played out. For the syntax, content, and interpretation of this list, see {{insertDTMF}}.

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. Let dtmf be the {{RTCDTMFSender}} associated with sender.
  4. If [= determine if DTMF can be sent =] for dtmf returns false, [= exception/throw =] an {{InvalidStateError}}.
  5. Let tones be the method's first argument.
  6. Let duration be the method's second argument.
  7. Let interToneGap be the method's third argument.
  8. If tones contains any {{unrecognized}} characters, [= exception/throw =] an {{InvalidCharacterError}}.
  9. Set the object's [[\ToneBuffer]] slot to tones.
  10. Set dtmf.[[\Duration]] to the value of duration.
  11. Set dtmf.[[\InterToneGap]] to the value of interToneGap.
  12. If the value of duration is less than 40 ms, set dtmf.[[\Duration]] to 40 ms.
  13. If the value of duration parameter is greater than 6000 ms, set dtmf.[[\Duration]] to 6000 ms.
  14. If the value of interToneGap is less than 30 ms, set dtmf.[[\InterToneGap]] to 30 ms.
  15. If the value of interToneGap is greater than 6000 ms, set dtmf.[[\InterToneGap]] 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.[[\CurrentDirection]] is neither {{RTCRtpTransceiverDirection/"sendrecv"}} nor {{RTCRtpTransceiverDirection/"sendonly"}}, abort these steps.
    2. If the [[\ToneBuffer]] slot contains the empty string, [= fire an event =] named {{tonechange}} using the {{RTCDTMFToneChangeEvent}} interface with the {{RTCDTMFToneChangeEvent/tone}} attribute set to an empty string at the {{RTCDTMFSender}} object and abort these steps.
    3. Remove the first character from the [[\ToneBuffer]] slot and let that character be tone.
    4. 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.
    5. 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.
    6. [= Fire an event =] named {{tonechange}} using the {{RTCDTMFToneChangeEvent}} interface with the {{RTCDTMFToneChangeEvent/tone}} attribute set to tone at the {{RTCDTMFSender}} object.

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

canInsertDTMF algorithm

To determine if DTMF can be sent for an {{RTCDTMFSender}} instance dtmfSender, the user agent MUST queue a task that runs the following steps:

  1. Let sender be the {{RTCRtpSender}} associated with dtmfSender.
  2. Let transceiver be the {{RTCRtpTransceiver}} associated with sender.
  3. Let connection be the {{RTCPeerConnection}} associated with transceiver.
  4. If connection's {{RTCPeerConnectionState}} is not {{RTCPeerConnectionState/"connected"}} return false.
  5. If sender.[[\SenderTrack]] is null return false.
  6. If transceiver.[[\CurrentDirection]] is neither {{RTCRtpTransceiverDirection/"sendrecv"}} nor {{RTCRtpTransceiverDirection/"sendonly"}} return false.
  7. If sender.[[\SendEncodings]][0].{{RTCRtpEncodingParameters/active}} is false return false.
  8. If no codec with mimetype "audio/telephone-event" has been negotiated for sending with this sender, return false.
  9. Otherwise, return true.

RTCDTMFToneChangeEvent

The {{tonechange}} event uses the {{RTCDTMFToneChangeEvent}} interface.

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

Constructors

RTCDTMFToneChangeEvent.constructor()

Attributes

tone of type DOMString, readonly

The {{tone}} attribute contains the character for the tone (including ",") that has just begun playout (see {{RTCDTMFSender/insertDTMF}} ). If the value is the empty string, it indicates that the [[\ToneBuffer]] slot is an empty string and that the previous tones have completed playback.

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

Dictionary RTCDTMFToneChangeEventInit Members

tone of type DOMString, defaulting to ""

The {{tone}} attribute contains the character for the tone (including ",") that has just begun playout (see {{RTCDTMFSender/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.

Statistics Model

Introduction

The basic statistics model is that the browser maintains a set of statistics for [= monitored object =]s, in the form of [= stats object =]s.

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

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

With a few exceptions, [= monitored object =]s, once created, exist for the duration of their associated {{RTCPeerConnection}}. This ensures statistics from them are available in the result from {{RTCPeerConnection/getStats()}} even past the associated peer connection being {{RTCPeerConnection/close}}d.

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

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 {{RTCRtpSender/track}} attribute matches selectorArg. If no such sender or receiver exists, or if more than one sender or receiver fit this criteria, return a promise [= rejected =] with a newly [= exception/create | 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.

RTCStatsReport Object

The {{RTCPeerConnection/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 ({{RTCStats/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 {{RTCRtpStreamStats/ssrc}} stats attribute).

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

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

RTCStats Dictionary

An {{RTCStats}} dictionary represents the [= stats object =] constructed by inspecting a specific [= monitored object =]. The {{RTCStats}} dictionary is a base type that specifies as set of default attributes, such as {{RTCStats/timestamp}} and {{RTCStats/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 {{RTCSentRtpStreamStats/bytesSent}} and {{RTCSentRtpStreamStats/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}}, 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.

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

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

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

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

The stats selection algorithm

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.

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 {{RTCPeerConnection}}, 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.

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:

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

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

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

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

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

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

Media Stream API Extensions for Network Use

Introduction

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

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

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

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

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

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

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

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

MediaStream

id

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

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

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

MediaStreamTrack

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

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

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

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

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

MediaTrackSupportedConstraints, MediaTrackCapabilities, MediaTrackConstraints and MediaTrackSettings

The concept of constraints and constrainable properties, including {{MediaTrackConstraints}} ({{MediaStreamTrack}}.getConstraints(), {{MediaStreamTrack}}.applyConstraints()), and {{MediaTrackSettings}} ({{MediaStreamTrack}}.getSettings()) are outlined in [[!GETUSERMEDIA]]. However, the constrainable properties of tracks sourced from a peer connection are different than those sourced by getUserMedia(); the constraints and settings applicable to {{MediaStreamTrack}}s sourced from a [= remote source =] are defined here. The settings of a remote track represent the latest frame received. {{MediaStreamTrack}}.getCapabilities() MUST always return the empty set and {{MediaStreamTrack}}.applyConstraints() MUST always reject with OverconstrainedError on remote tracks for constraints defined here.

The following constrainable properties are defined to apply to video {{MediaStreamTrack}}s sourced from a [= remote source =]:

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

This document does not define any constrainable properties to apply to audio {{MediaStreamTrack}}s sourced from a [= remote source =].

Examples and Call Flows

Simple Peer-to-peer Example

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

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

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

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

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

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

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

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

Advanced Peer-to-peer Example with Warm-up

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

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

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

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

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

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

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

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

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

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

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

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

Simulcast Example

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

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

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

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

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

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

Peer-to-peer Data Example

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

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

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

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

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

  // create data channel and setup chat using "negotiated" pattern
  channel = pc.createDataChannel('chat', {negotiated: true, id: 0});
  channel.onopen = () => input.disabled = false;
  channel.onmessage = ({data}) => showChatMessage(data);

  input.onkeypress = ({keyCode}) => {
    // only send when user presses enter
    if (keyCode != 13) return;
    channel.send(input.value);
  }
}

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

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

Call Flow Browser to Browser

This shows an example of one possible call flow between two browsers. This does not show the procedure to get access to local media or every callback that gets fired but instead tries to reduce it down to only show the key events and messages.

A message sequence chart detailing a call flow between two browsers

DTMF Example

Examples assume that sender is an {{RTCRtpSender}}.

Sending the DTMF signal "1234" with 500 ms duration per tone:

if (sender.dtmf.canInsertDTMF) {
  const duration = 500;
  sender.dtmf.insertDTMF('1234', duration);
} else {
  console.log('DTMF function not available');
}
      

Send the DTMF signal "123" and abort after sending "2".

async function sendDTMF() {
  if (sender.dtmf.canInsertDTMF) {
    sender.dtmf.insertDTMF('123');
    await new Promise(r => sender.dtmf.ontonechange = e => e.tone == '2' && r());
    // empty the buffer to not play any tone after "2"
    sender.dtmf.insertDTMF('');
  } else {
    console.log('DTMF function not available');
  }
}
      

Send the DTMF signal "1234", and light up the active key using lightKey(key) while the tone is playing (assuming that lightKey("") will darken all the keys):

const wait = ms => new Promise(resolve => setTimeout(resolve, ms));

if (sender.dtmf.canInsertDTMF) {
  const duration = 500; // ms
  sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '1234', duration);
  sender.dtmf.ontonechange = async ({tone}) => {
    if (!tone) return;
    lightKey(tone); // light up the key when playout starts
    await wait(duration);
    lightKey(''); // turn off the light after tone duration
  };
} else {
  console.log('DTMF function not available');
}
      

It is always safe to append to the tone buffer. This example appends before any tone playout has started as well as during playout.

if (sender.dtmf.canInsertDTMF) {
  sender.dtmf.insertDTMF('123');
  // append more tones to the tone buffer before playout has begun
  sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '456');

  sender.dtmf.ontonechange = ({tone}) => {
    // append more tones when playout has begun
    if (tone != '1') return;
    sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '789');
  };
} else {
  console.log('DTMF function not available');
}
      

Send a 1-second "1" tone followed by a 2-second "2" tone:

if (sender.dtmf.canInsertDTMF) {
  sender.dtmf.ontonechange = ({tone}) => {
    if (tone == '1') {
      sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '2', 2000);
    }
  };
  sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '1', 1000);
} else {
  console.log('DTMF function not available');
}
      

Perfect Negotiation Example

Perfect negotiation is a recommended pattern to manage negotiation transparently, abstracting this asymmetric task away from the rest of an application. This pattern has advantages over one side always being the offerer, as it lets applications operate on both peer connection objects simultaneously without risk of glare (an offer coming in outside of {{RTCSignalingState/"stable"}} state). The rest of the application may use any and all modification methods and attributes, without worrying about signaling state races.

It designates different roles to the two peers, with behavior to resolve signaling collisions between them:

  1. The polite peer uses rollback to avoid collision with an incoming offer.

  2. The impolite peer ignores an incoming offer when this would collide with its own.

Together, they manage signaling for the rest of the application in a manner that doesn't deadlock. The example assumes a polite boolean variable indicating the designated role:

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

// call start() anytime on either end to add camera and microphone to connection
async function start() {
  try {
    const stream = await navigator.mediaDevices.getUserMedia(constraints);
    for (const track of stream.getTracks()) {
      pc.addTrack(track, stream);
    }
    selfView.srcObject = stream;
  } catch (err) {
    console.error(err);
  }
}

pc.ontrack = ({streams}) => {
  if (remoteView.srcObject) return;
  remoteView.srcObject = streams[0];
};

// - The perfect negotiation logic, separated from the rest of the application ---

let offering = false, ignoredOffer = false;

pc.onnegotiationneeded = async () => {
  try {
    offering = true;
    await pc.setLocalDescription();
    signaling.send({description: pc.localDescription});
  } catch (err) {
     console.error(err);
  } finally {
    offering = false;
  }
};

signaling.onmessage = async ({data: {description, candidate}}) => {
  try {
    if (description) {
      const collision = pc.signalingState != "stable" || offering;
      if (ignoredOffer = !polite && description.type == "offer" && collision) {
        return;
      }
      await pc.setRemoteDescription(description); // SRD rolls back as needed
      if (description.type == "offer") {
        await pc.setLocalDescription();
        signaling.send({description: pc.localDescription});
      }
    } else if (candidate) {
      try {
        await pc.addIceCandidate(candidate);
      } catch (err) {
        if (!ignoredOffer) throw err; // Suppress ignored offer's candidates
      }
    }
  } catch (err) {
    console.error(err);
  }
}
      

Note that this is timing sensitive, and deliberately uses versions of {{RTCPeerConnection/setLocalDescription}} (without arguments) and {{RTCPeerConnection/setRemoteDescription}} (with implicit rollback) to avoid races with other signaling messages being serviced.

The ignoredOffer variable is needed, because the {{RTCPeerConnection}} object on the impolite side is never told about ignored offers. We must therefore suppress errors from incoming candidates belonging to such offers.

Error Handling

Some operations throw or fire {{RTCError}}. This is an extension of {{DOMException}} that carries additional WebRTC-specific information.

RTCError Interface

[Exposed=Window]
interface RTCError : DOMException {
  constructor(RTCErrorInit init, optional DOMString message = "");
  readonly attribute RTCErrorDetailType errorDetail;
  readonly attribute long? sdpLineNumber;
  readonly attribute long? sctpCauseCode;
  readonly attribute unsigned long? receivedAlert;
  readonly attribute unsigned long? sentAlert;
};

Constructors

constructor()

Run the following steps:

  1. Let init be the constructor's first argument.

  2. Let message be the constructor's second argument.

  3. Let e be a new {{RTCError}} object.

  4. Invoke the {{DOMException}} constructor of e with the {{DOMException/message}} argument set to message and the {{DOMException/name}} argument set to "OperationError".

    This name does not have a mapping to a legacy code so e.{{DOMException/code}} will return 0.

  5. Set all {{RTCError}} attributes of e to the value of the corresponding attribute in init if it is present, otherwise set it to null.

  6. Return e.

Attributes

errorDetail of type RTCErrorDetailType, readonly

The WebRTC-specific error code for the type of error that occurred.

sdpLineNumber of type long, readonly, nullable

If {{RTCError/errorDetail}} is {{RTCErrorDetailType/"sdp-syntax-error"}} this is the line number where the error was detected (the first line has line number 1).

sctpCauseCode of type long, readonly, nullable

If {{RTCError/errorDetail}} is {{RTCErrorDetailType/"sctp-failure"}} this is the SCTP cause code of the failed SCTP negotiation.

receivedAlert of type unsigned long, readonly, nullable

If {{RTCError/errorDetail}} is {{RTCErrorDetailType/"dtls-failure"}} and a fatal DTLS alert was received, this is the value of the DTLS alert received.

sentAlert of type unsigned long, readonly, nullable

If {{RTCError/errorDetail}} is {{RTCErrorDetailType/"dtls-failure"}} and a fatal DTLS alert was sent, this is the value of the DTLS alert sent.

All attributes defined in {{RTCError}} are marked at risk due to lack of implementation ({{errorDetail}}, {{sdpLineNumber}}, {{sctpCauseCode}}, {{receivedAlert}} and {{sentAlert}}). This does not include attributes inherited from {{DOMException}}.

RTCErrorInit Dictionary

dictionary RTCErrorInit {
  required RTCErrorDetailType errorDetail;
  long sdpLineNumber;
  long sctpCauseCode;
  unsigned long receivedAlert;
  unsigned long sentAlert;
};

The errorDetail, sdpLineNumber, sctpCauseCode, receivedAlert and sentAlert members of {{RTCErrorInit}} have the same definitions as the attributes of the same name of {{RTCError}}.

RTCErrorDetailType Enum

enum RTCErrorDetailType {
  "data-channel-failure",
  "dtls-failure",
  "fingerprint-failure",
  "sctp-failure",
  "sdp-syntax-error",
  "hardware-encoder-not-available",
  "hardware-encoder-error"
};
Enumeration description
data-channel-failure The data channel has failed.
dtls-failure The DTLS negotiation has failed or the connection has been terminated with a fatal error. The {{DOMException/message}} contains information relating to the nature of error. If a fatal DTLS alert was received, the {{RTCError/receivedAlert}} attribute is set to the value of the DTLS alert received. If a fatal DTLS alert was sent, the {{RTCError/sentAlert}} attribute is set to the value of the DTLS alert sent.
fingerprint-failure The {{RTCDtlsTransport}}'s remote certificate did not match any of the fingerprints provided in the SDP. If the remote peer cannot match the local certificate against the provided fingerprints, this error is not generated. Instead a "bad_certificate" (42) DTLS alert might be received from the remote peer, resulting in a {{RTCErrorDetailType/"dtls-failure"}}.
sctp-failure The SCTP negotiation has failed or the connection has been terminated with a fatal error. The {{RTCError/sctpCauseCode}} attribute is set to the SCTP cause code.
sdp-syntax-error The SDP syntax is not valid. The {{RTCError/sdpLineNumber}} attribute is set to the line number in the SDP where the syntax error was detected.
hardware-encoder-not-available The hardware encoder resources required for the requested operation are not available.
hardware-encoder-error The hardware encoder does not support the provided parameters.

RTCErrorEvent Interface

The {{RTCErrorEvent}} interface is defined for cases when an {{RTCError}} is raised as an event:

[Exposed=Window]
interface RTCErrorEvent : Event {
  constructor(DOMString type, RTCErrorEventInit eventInitDict);
  [SameObject] readonly attribute RTCError error;
};

Constructors

constructor()

Constructs a new {{RTCErrorEvent}}.

Attributes

error of type {{RTCError}}, readonly

The {{RTCError}} describing the error that triggered the event.

RTCErrorEventInit Dictionary

dictionary RTCErrorEventInit : EventInit {
  required RTCError error;
};

Dictionary RTCErrorEventInit Members

error of type {{RTCError}}

The {{RTCError}} describing the error associated with the event (if any).

Event summary

The following events fire on {{RTCDataChannel}} objects:

Event name Interface Fired when...
open {{Event}} The {{RTCDataChannel}} object's [= underlying data transport =] has been established (or re-established).
message {{MessageEvent}} [[html]] A message was successfully received.
bufferedamountlow {{Event}} The {{RTCDataChannel}} object's {{RTCDataChannel/bufferedAmount}} decreases from above its {{RTCDataChannel/bufferedAmountLowThreshold}} to less than or equal to its {{RTCDataChannel/bufferedAmountLowThreshold}}.
error {{RTCErrorEvent}} An error occurred on the data channel.
closing {{Event}} The {{RTCDataChannel}} object transitions to the {{RTCDataChannelState/"closing"}} state
close {{Event}} The {{RTCDataChannel}} object's [= underlying data transport =] has been closed.

The following events fire on {{RTCPeerConnection}} objects:

Event name Interface Fired when...
track {{RTCTrackEvent}} New incoming media has been negotiated for a specific {{RTCRtpReceiver}}, and that receiver's {{RTCRtpReceiver/track}} has been added to any associated remote {{MediaStream}}s.
negotiationneeded {{Event}} The browser wishes to inform the application that session negotiation needs to be done (i.e. a createOffer call followed by setLocalDescription).
signalingstatechange {{Event}} The [= signaling state =] has changed. This state change is the result of either {{RTCPeerConnection/setLocalDescription}} or {{RTCPeerConnection/setRemoteDescription}} being invoked.
iceconnectionstatechange {{Event}} The {{RTCPeerConnection}}'s [= ICE connection state =] has changed.
icegatheringstatechange {{Event}} The {{RTCPeerConnection}}'s [= ICE gathering state =] has changed.
icecandidate {{RTCPeerConnectionIceEvent}} A new {{RTCIceCandidate}} is made available to the script.
connectionstatechange {{Event}} The {{RTCPeerConnection}}.{{RTCPeerConnection/connectionState}} has changed.
icecandidateerror {{RTCPeerConnectionIceErrorEvent}} A failure occured when gathering ICE candidates.
datachannel {{RTCDataChannelEvent}} A new {{RTCDataChannel}} is dispatched to the script in response to the other peer creating a channel.

The following events fire on {{RTCDTMFSender}} objects:

Event name Interface Fired when...
tonechange {{RTCDTMFToneChangeEvent}} The {{RTCDTMFSender}} object has either just begun playout of a tone (returned as the {{RTCDTMFToneChangeEvent/tone}} attribute) or just ended the playout of tones in the {{RTCDTMFSender/toneBuffer}} (returned as an empty value in the {{RTCDTMFToneChangeEvent/tone}} attribute).

The following events fire on {{RTCIceTransport}} objects:

Event name Interface Fired when...
statechange {{Event}} The {{RTCIceTransport}} state changes.
gatheringstatechange {{Event}} The {{RTCIceTransport}} gathering state changes.
selectedcandidatepairchange {{Event}} The {{RTCIceTransport}}'s selected candidate pair changes.

The following events fire on {{RTCDtlsTransport}} objects:

Event name Interface Fired when...
statechange {{Event}} The {{RTCDtlsTransport}} state changes.
error {{RTCErrorEvent}} An error occurred on the {{RTCDtlsTransport}} (either {{RTCErrorDetailType/"dtls-failure"}} or {{RTCErrorDetailType/"fingerprint-failure"}}).

The following events fire on {{RTCSctpTransport}} objects:

Event name Interface Fired when...
statechange {{Event}} The {{RTCSctpTransport}} state changes.

Privacy and Security Considerations

This section is non-normative; it specifies no new behaviour, but instead summarizes information already present in other parts of the specification. The overall security considerations of the general set of APIs and protocols used in WebRTC are described in [[?RTCWEB-SECURITY-ARCH]].

Impact on same origin policy

This document extends the Web platform with the ability to set up real time, direct communication between browsers and other devices, including other browsers.

This means that data and media can be shared between applications running in different browsers, or between an application running in the same browser and something that is not a browser, something that is an extension to the usual barriers in the Web model against sending data between entities with different origins.

The WebRTC specification provides no user prompts or chrome indicators for communication; it assumes that once the Web page has been allowed to access media, it is free to share that media with other entities as it chooses. Peer-to-peer exchanges of data view WebRTC datachannels can thus occur without any user explicit consent or involvement, similarly as a server-mediated exchange (e.g. via Web Sockets) could occur without user involvement.

Revealing IP addresses

Even without WebRTC, the Web server providing a Web application will know the public IP address to which the application is delivered. Setting up communications exposes additional information about the browser’s network context to the web application, and may include the set of (possibly private) IP addresses available to the browser for WebRTC use. Some of this information has to be passed to the corresponding party to enable the establishment of a communication session.

Revealing IP addresses can leak location and means of connection; this can be sensitive. Depending on the network environment, it can also increase the fingerprinting surface and create persistent cross-origin state that cannot easily be cleared by the user.

A connection will always reveal the IP addresses proposed for communication to the corresponding party. The application can limit this exposure by choosing not to use certain addresses using the settings exposed by the {{RTCIceTransportPolicy}} dictionary, and by using relays (for instance TURN servers) rather than direct connections between participants. One will normally assume that the IP address of TURN servers is not sensitive information. These choices can for instance be made by the application based on whether the user has indicated consent to start a media connection with the other party.

Mitigating the exposure of IP addresses to the application itself requires limiting the IP addresses that can be used, which will impact the ability to communicate on the most direct path between endpoints. Browsers are encouraged to provide appropriate controls for deciding which IP addresses are made available to applications, based on the security posture desired by the user. The choice of which addresses to expose is controlled by local policy (see [[RTCWEB-IP-HANDLING]] for details).

Impact on local network

Since the browser is an active platform executing in a trusted network environment (inside the firewall), it is important to limit the damage that the browser can do to other elements on the local network, and it is important to protect data from interception, manipulation and modification by untrusted participants.

Mitigations include:

These measures are specified in the relevant IETF documents.

Confidentiality of Communications

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

Communication certificates may be opaquely shared using {{MessagePort/postMessage()}} in anticipation of future needs. User agents are strongly encouraged to isolate the private keying material these objects hold a handle to, from the processes that have access to the {{RTCCertificate}} objects, to reduce memory attack surface.

Persistent information exposed by WebRTC

As described above, the list of IP addresses exposed by the WebRTC API can be used as a persistent cross-origin state.

Beyond IP addresses, the WebRTC API exposes information about the underlying media system via the {{RTCRtpSender}}.{{RTCRtpSender/getCapabilities}} and {{RTCRtpReceiver}}.{{RTCRtpReceiver/getCapabilities}} methods, including detailed and ordered information about the codecs that the system is able to produce and consume. A subset of that information is likely to be represented in the SDP session descriptions generated, exposed and transmitted during session negotiation. That information is in most cases persistent across time and origins, and increases the fingerprint surface of a given device.

When establishing DTLS connections, the WebRTC API can generate certificates that can be persisted by the application (e.g. in IndexedDB). These certificates are not shared across origins, and get cleared when persistent storage is cleared for the origin.

Setting SDP from remote endpoints

{{RTCPeerConnection/setRemoteDescription}} guards against malformed and invalid SDP by throwing exceptions, but makes no attempt to guard against SDP that might be unexpected by the application. Setting the remote description can cause significant resources to be allocated (including image buffers and network ports), media to start flowing (which may have privacy and bandwidth implications) among other things. An application that does not guard against malicious SDP could be at risk of resource deprivation, unintentionally allowing incoming media or at risk of not having certain events fire like {{RTCPeerConnection/ontrack}} if the other endpoint does not negotiate sending. Applications need to be on guard against malevolent SDP.

Accessibility Considerations

The WebRTC 1.0 specification exposes an API to control protocols (defined within the IETF) necessary to establish real-time audio, video and data exchange.

The Telecommunications Device for the Deaf (TDD/TTY) enables individuals who are hearing or speech impaired (among others) to communicate over telephone lines. Real-time Text, defined in [[RFC4103]], utilizes T.140 encapsulated in RTP to enable the transition from TDD/TTY devices to IP-based communications, including emergency communication with Public Safety Access Points (PSAP).

Since Real-time Text requires the ability to send and receive data in near real time, it can be best supported via the WebRTC 1.0 data channel API. As defined by the IETF, the data channel protocol utilizes the SCTP/DTLS/UDP protocol stack, which supports both reliable and unreliable data channels. The IETF chose to standardize SCTP/DTLS/UDP over proposals for an RTP data channel which relied on SRTP key management and were focused on unreliable communications.

Since the IETF chose a different approach than the RTP data channel as part of the WebRTC suite of protocols, as of the time of this publication there is no standardized way for the WebRTC APIs to directly support Real-time Text as defined at IETF and implemented in U.S. (FCC) regulations. The WebRTC working Group will evaluate whether the developing IETF protocols in this space warrant direct exposure in the browser APIs and is looking for input from the relevant user communities on this potential gap.

Within the IETF MMUSIC Working Group, work is ongoing to enable Real-time text to be sent over the WebRTC data channel, allowing gateways to be deployed to translate between the SCTP data channel protocol and RFC 4103 Real-time text. This work, once completed, is expected to enable a unified and interoperable approach for integrating real-time text in WebRTC user-agents (including browsers) - through a gateway or otherwise.

At the time of this publication, gateways that enable effective RTT support in WebRTC clients can be developed e.g. through a custom WebRTC data channel. This is deemed sufficient until such time as future standardized gateways are enabled via IETF protocols such as the SCTP data channel protocol and RFC 4103 Real-time text. This will need to be defined at IETF in conjunction with related work at W3C groups to effectively and consistently standardise RTT support internationally.

Acknowledgements

The editors wish to thank the Working Group chairs and Team Contact, Harald Alvestrand, Stefan Håkansson, Erik Lagerway and Dominique Hazaël-Massieux, for their support. Substantial text in this specification was provided by many people including Martin Thomson, Harald Alvestrand, Justin Uberti, Eric Rescorla, Peter Thatcher, Jan-Ivar Bruaroey and Peter Saint-Andre. Dan Burnett would like to acknowledge the significant support received from Voxeo and Aspect during the development of this specification.

The {{RTCRtpSender}} and {{RTCRtpReceiver}} objects were initially described in the W3C ORTC CG, and have been adapted for use in this specification.