webrtc library Null safety

WebRTC 1.0: Real-Time Communication Between Browsers

https://w3c.github.io/webrtc-pc/

Classes

RTCAnswerOptions
The RTCAnswerOptions dictionary is used to provide optional settings when creating an SDP answer using RTCPeerConnection.createOffer() after receiving an offer from a remote peer. The createOffer() method's options parameter is of this type.
RTCCertificate
The interface of the WebRTC API provides an object represents a certificate that an RTCPeerConnection uses to authenticate.
RTCCertificateExpiration
RTCConfiguration
The dictionary is used to provide configuration options for an RTCPeerConnection. It may be passed into the constructor when instantiating a connection, or used with the RTCPeerConnection.getConfiguration() and RTCPeerConnection.setConfiguration() methods, which allow inspecting and changing the configuration while a connection is established. [...]
RTCDataChannel
RTCDataChannelEvent
RTCDataChannelEventInit
RTCDataChannelInit
RTCDtlsFingerprint
RTCDtlsTransport
RTCDTMFSender
The interface provides a mechanism for transmitting DTMF codes on a WebRTC RTCPeerConnection. You gain access to the connection's through the RTCRtpSender.dtmf property on the audio track you wish to send DTMF with. [...]
RTCDTMFToneChangeEvent
The interface represents events sent to indicate that DTMF tones have started or finished playing. This interface is used by the tonechange event.
RTCDTMFToneChangeEventInit
RTCError
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.
RTCErrorEvent
The WebRTC API's 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.
RTCErrorEventInit
RTCErrorInit
RTCIceCandidate
The interface—part of the WebRTC API—represents a candidate Internet Connectivity Establishment (ICE) configuration which may be used to establish an RTCPeerConnection. [...]
RTCIceCandidateInit
The WebRTC API's dictionary, which contains the information needed to fundamentally describe an RTCIceCandidate. is used when using new RTCIceCandidate() to create a new ICE candidate object. It's also used as the return value from the RTCIceCandidate.toJSON() method, and can be passed directly into RTCPeerConnection.addIceCandidate() to add a candidate to the peer connection.
RTCIceCandidatePair
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.
RTCIceParameters
The RTCIceParameters dictionary specifies the username fragment and password assigned to an ICE session. During ICE negotiation, each peer's username fragment and password are recorded in an object, which can be obtained from the RTCIceTransport by calling its getLocalParameters() or getRemoteParameters() method, depending on which end interests you.
RTCIceServer
The RTCIceServer dictionary defines how to connect to a single ICE server (such as a STUN or TURN server). Objects of this type are provided in the configuration of an RTCPeerConnection, in the iceServers array.
RTCIceTransport
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.
RTCLocalSessionDescriptionInit
RTCOfferAnswerOptions
The WebRTC API's RTCOfferAnswerOptions dictionary is used to specify options that configure and control the process of creating WebRTC offers or answers. It's used as the base type for the options parameter when calling createOffer() or createAnswer() on an RTCPeerConnection. [...]
RTCOfferOptions
The RTCOfferOptions dictionary is used to provide optional settings when creating an RTCPeerConnection offer with the createOffer() method.
RTCPeerConnection
RTCPeerConnectionIceErrorEvent
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.
RTCPeerConnectionIceErrorEventInit
RTCPeerConnectionIceEvent
The interface represents events that occurs in relation to ICE candidates with the target, usually an RTCPeerConnection. Only one event is of this type: icecandidate.
RTCPeerConnectionIceEventInit
RTCRtcpParameters
The RTCRtcpParameters dictionary provides parameters of an RTCP connection. It's used as the value of the rtcp property of the parameters of an RTCRtpSender or RTCRtpReceiver.
RTCRtpCapabilities
The RTCRtpCapabilities dictionary is a data type used to describe the capabilities of an RTCRtpSender or RTCRtpReceiver in response to a call to the RTCRtpSender.getCapabilities() or RTCRtpReceiver.getCapabilities() static functions, both of which return an array of objects. [...]
RTCRtpCodecCapability
The WebRTC API's RTCRtpCodecCapability dictionary provides information describing the capabilities of a single media codec.
RTCRtpCodecParameters
The dictionary, part of the WebRTC API, is used to describe the configuration parameters for a single media codec. In addition to being the type of the RTCRtpParameters.codecs property, it's used when calling RTCRtpTransceiver.setCodecPreferences() to configure a transceiver's codecs before beginning the offer/answer process to establish a WebRTC peer connection. [...]
RTCRtpCodingParameters
RTCRtpContributingSource
The dictionary of the WebRTC API is used by getContributingSources() to provide information about a given contributing source (CSRC), including the most recent time a packet that the source contributed was played out. [...]
RTCRtpDecodingParameters
RTCRtpEncodingParameters
An instance of the WebRTC API's dictionary describes a single configuration of a codec for an RTCRtpSender. It's used in the RTCRtpSendParameters describing the configuration of an RTP sender's encodings; RTCRtpDecodingParameters is used to describe the configuration of an RTP receiver's encodings.
RTCRtpHeaderExtensionCapability
RTCRtpHeaderExtensionParameters
RTCRtpParameters
The RTCRtpParameters dictionary is the basic object describing the parameters of an RTP transport. It is extended separately for senders and receivers in the form of the RTCRtpSendParameters and RTCRtpReceiveParameters dictionaries. [...]
RTCRtpReceiveParameters
The RTCRtpReceiveParameters dictionary, based upon the RTCRtpParameters dictionary, is returned by the RTCRtpReceiver method getParameters(). It describes the parameters being used by the receiver's RTP connection to the remote peer.
RTCRtpReceiver
The interface of the WebRTC API manages the reception and decoding of data for a MediaStreamTrack on an RTCPeerConnection.
RTCRtpSender
The interface provides the ability to control and obtain details about how a particular MediaStreamTrack is encoded and sent to a remote peer. With it, you can configure the encoding used for the corresponding track, get information about the device's media capabilities, and so forth. You can also obtain access to an RTCDTMFSender which can be used to send DTMF codes (to simulate the user pressing buttons on a telephone's dial pad) to the remote peer.
RTCRtpSendParameters
The WebRTC API's RTCRtpSendParameters dictionary is used to specify the parameters for an RTCRtpSender when calling its setParameters() method.
RTCRtpSynchronizationSource
The dictionary of the WebRTC API is used by getSynchronizationSources() to describe a particular synchronization source (SSRC). A synchronization source is a single source that shares timing and sequence number space. Since implements RTCRtpContributingSource, its properties are also available. [...]
RTCRtpTransceiver
The WebRTC interface describes a permanent pairing of an RTCRtpSender and an RTCRtpReceiver, along with some shared state. [...]
RTCRtpTransceiverInit
The dictionary is used when calling the WebRTC function RTCPeerConnection.addTransceiver() to provide configuration options for the new transceiver.
RTCSctpTransport
Experimental This is an experimental technologyCheck the Browser compatibility table carefully before using this in production. The 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. [...]
RTCSessionDescription
Experimental This is an experimental technologyCheck the Browser compatibility table carefully before using this in production. The interface describes one end of a connection—or potential connection—and how it's configured. Each consists of a description type indicating which part of the offer/answer negotiation process it describes and of the SDP descriptor of the session. [...]
RTCSessionDescriptionInit
RTCStats
The dictionary is the basic statistics object used by WebRTC's statistics monitoring model, providing the properties required of all statistics data objects. Specific classes of statistic are defined as dictionaries based on . For example, statistics about a received RTP stream are represented by RTCReceivedRtpStreamStats.
RTCStatsReport
Draft This page is not complete. This page is currently incomplete and under active construction. Please be aware that it's not going to answer all of your questions just yet. The RTCStatsReport interface provides a statistics report obtained by calling one of the RTCPeerConnection.getStats(), RTCRtpReceiver.getStats(), and RTCRtpSender.getStats() methods. This statistics report contains a mapping of statistic category string names to objects containing the corresponding statistics data. [...]
RTCTrackEvent
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 target is the RTCPeerConnection object to which the track is being added. [...]
RTCTrackEventInit
The WebRTC API's RTCTrackEventInit dictionary is used to provide information describing an RTCTrackEvent when instantiating a new track event using new RTCTrackEvent().

Enums

RTCBundlePolicy
RTCDataChannelState
RTCDtlsTransportState
RTCErrorDetailType
RTCIceCandidateType
RTCIceComponent
RTCIceConnectionState
RTCIceCredentialType
RTCIceGathererState
RTCIceGatheringState
RTCIceProtocol
RTCIceRole
RTCIceTcpCandidateType
RTCIceTransportPolicy
RTCIceTransportState
RTCPeerConnectionState
RTCRtcpMuxPolicy
RTCRtpTransceiverDirection
RTCSctpTransportState
RTCSdpType
RTCSignalingState