Package-level declarations
Types
The RTCCertificate interface of the WebRTC API provides an object representing a certificate that an RTCPeerConnection uses to authenticate.
The RTCDataChannel interface represents a network channel which can be used for bidirectional peer-to-peer transfers of arbitrary data. Every data channel is associated with an RTCPeerConnection, and each peer connection can have up to a theoretical maximum of 65,534 data channels (the actual limit may vary from browser to browser).
The RTCDataChannelEvent interface represents an event related to a specific RTCDataChannel.
The RTCDtlsTransport interface provides access to information about the Datagram Transport Layer Security (DTLS) transport over which a RTCPeerConnection's RTP and RTCP packets are sent and received by its RTCRtpSender and RTCRtpReceiver objects.
The RTCDTMFSender interface provides a mechanism for transmitting DTMF codes on a WebRTC RTCPeerConnection. You gain access to the connection's RTCDTMFSender through the RTCRtpSender.dtmf property on the audio track you wish to send DTMF with.
The RTCDTMFToneChangeEvent interface represents events sent to indicate that DTMF tones have started or finished playing. This interface is used by the tonechange event.
The RTCEncodedAudioFrame of the WebRTC API represents an encoded audio frame in the WebRTC receiver or sender pipeline, which may be modified using a WebRTC Encoded Transform.
The RTCEncodedVideoFrame of the WebRTC API represents an encoded video frame in the WebRTC receiver or sender pipeline, which may be modified using a WebRTC Encoded Transform.
The RTCError interface describes an error which has occurred while handling WebRTC operations. It's based upon the standard DOMException interface that describes general DOM errors.
The WebRTC API's RTCErrorEvent interface represents an error sent to a WebRTC object. It's based on the standard Event interface, but adds RTC-specific information describing the error, as shown below.
The RTCIceCandidate interface—part of the WebRTC API—represents a candidate Interactive Connectivity Establishment (ICE) configuration which may be used to establish an RTCPeerConnection.
The RTCIceCandidatePair dictionary describes a pair of ICE candidates which together comprise a description of a viable connection between two WebRTC endpoints. It is used as the return value from RTCIceTransport.getSelectedCandidatePair() to identify the currently-selected candidate pair identified by the ICE agent.
The RTCIceTransport interface provides access to information about the ICE transport layer over which the data is being sent and received. This is particularly useful if you need to access state information about the connection.
The RTCPeerConnection interface represents a WebRTC connection between the local computer and a remote peer. It provides methods to connect to a remote peer, maintain and monitor the connection, and close the connection once it's no longer needed.
The RTCPeerConnectionIceErrorEvent interface—based upon the Event interface—provides details pertaining to an ICE error announced by sending an icecandidateerror event to the RTCPeerConnection object.
The RTCPeerConnectionIceEvent interface represents events that occur in relation to ICE candidates with the target, usually an RTCPeerConnection.
The RTCRtpReceiver interface of the WebRTC API manages the reception and decoding of data for a MediaStreamTrack on an RTCPeerConnection.
The RTCRtpScriptTransform interface of the WebRTC API is used to insert a WebRTC Encoded Transform (a TransformStream running in a worker thread) into the WebRTC sender and receiver pipelines.
The RTCRtpScriptTransformer interface of the WebRTC API provides a worker-side Stream API interface that a WebRTC Encoded Transform can use to modify encoded media frames in the incoming and outgoing WebRTC pipelines.
The RTCRtpSender interface provides the ability to control and obtain details about how a particular MediaStreamTrack is encoded and sent to a remote peer.
The WebRTC interface RTCRtpTransceiver describes a permanent pairing of an RTCRtpSender and an RTCRtpReceiver, along with some shared state.
The RTCSctpTransport interface provides information which describes a Stream Control Transmission Protocol (SCTP) transport. This provides information about limitations of the transport, but also provides a way to access the underlying Datagram Transport Layer Security (DTLS) transport over which SCTP packets for all of an RTCPeerConnection's data channels are sent and received.
The RTCSessionDescription interface describes one end of a connection—or potential connection—and how it's configured. Each RTCSessionDescription consists of a description type indicating which part of the offer/answer negotiation process it describes and of the SDP descriptor of the session.
The RTCStatsReport interface of the WebRTC API provides a statistics report for a RTCPeerConnection, RTCRtpSender, or RTCRtpReceiver.
The WebRTC API interface RTCTrackEvent represents the track event, which is sent when a new MediaStreamTrack is added to an RTCRtpReceiver which is part of the RTCPeerConnection.
The RTCTransformEvent of the WebRTC API represent an event that is fired in a dedicated worker when an encoded frame has been queued for processing by a WebRTC Encoded Transform.
Properties
Functions
The addIceCandidate() method of the RTCPeerConnection interface adds a new remote candidate to the connection's remote description, which describes the state of the remote end of the connection.
The createAnswer() method of the RTCPeerConnection interface creates an SDP answer to an offer received from a remote peer during the offer/answer negotiation of a WebRTC connection.
The createOffer() method of the RTCPeerConnection interface initiates the creation of an SDP offer for the purpose of starting a new WebRTC connection to a remote peer.
The generateCertificate() static function of the RTCPeerConnection interface creates an X.509 certificate and corresponding private key, returning a promise that resolves with the new RTCCertificate once it's generated.
The generateKeyFrame() method of the RTCRtpScriptTransformer interface causes a video encoder to generate a key frame.
The getStats() method of the RTCPeerConnection interface returns a promise which resolves with data providing statistics about either the overall connection or about the specified MediaStreamTrack.
The RTCRtpReceiver method getStats() asynchronously requests an RTCStatsReport object which provides statistics about incoming traffic on the owning RTCPeerConnection, returning a Promise whose fulfillment handler will be called once the results are available.
The RTCRtpSender method getStats() asynchronously requests an RTCStatsReport object which provides statistics about outgoing traffic on the RTCPeerConnection which owns the sender, returning a Promise which is fulfilled when the results are available.
The RTCRtpSender method replaceTrack() replaces the track currently being used as the sender's source with a new MediaStreamTrack.
The sendKeyFrameRequest() method of the RTCRtpScriptTransformer interface may be called by a WebRTC Encoded Transform that is processing incoming encoded video frames, in order to request a key frame from the sender.
The setLocalDescription() method of the RTCPeerConnection interface changes the local description associated with the connection. This description specifies the properties of the local end of the connection, including the media format. The method takes a single parameter—the session description—and it returns a Promise which is fulfilled once the description has been changed, asynchronously.
The setParameters() method of the RTCRtpSender interface applies changes the configuration of sender's track, which is the MediaStreamTrack for which the RTCRtpSender is responsible.
The setRemoteDescription() method of the RTCPeerConnection interface sets the specified session description as the remote peer's current offer or answer. The description specifies the properties of the remote end of the connection, including the media format. The method takes a single parameter—the session description—and it returns a Promise which is fulfilled once the description has been changed, asynchronously.