| Interface | Description |
|---|---|
| GenerateAssertionCallback | |
| 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.
|
| RTCAnswerOptions.Builder |
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.
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| RTCAudioHandlerStats | |
| RTCAudioHandlerStats.Builder | |
| RTCAudioReceiverStats | |
| RTCAudioReceiverStats.Builder | |
| RTCAudioSenderStats | |
| RTCAudioSenderStats.Builder | |
| RTCAudioSourceStats | |
| RTCAudioSourceStats.Builder | |
| RTCCertificateExpiration | |
| RTCCertificateExpiration.Builder | |
| RTCCertificateStats | |
| RTCCertificateStats.Builder | |
| RTCCodecStats | |
| RTCCodecStats.Builder | |
| RTCConfiguration |
The RTCConfiguration dictionary is used to provide configuration options for an RTCPeerConnection.
|
| RTCConfiguration.Builder |
The RTCConfiguration dictionary is used to provide configuration options for an RTCPeerConnection.
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| RTCDataChannelEventHandler |
Handle events of type RTCDataChannelEvent
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| RTCDataChannelEventInit | |
| RTCDataChannelEventInit.Builder | |
| RTCDataChannelEventListener |
Listener for events of type RTCDataChannelEvent
|
| RTCDataChannelInit | |
| RTCDataChannelInit.Builder | |
| RTCDataChannelStats | |
| RTCDataChannelStats.Builder | |
| RTCDtlsFingerprint | |
| RTCDtlsFingerprint.Builder | |
| RTCDTMFToneChangeEventHandler |
Handle events of type RTCDTMFToneChangeEvent
|
| RTCDTMFToneChangeEventInit | |
| RTCDTMFToneChangeEventInit.Builder | |
| RTCDTMFToneChangeEventListener |
Listener for events of type RTCDTMFToneChangeEvent
|
| RTCErrorEventHandler |
Handle events of type RTCErrorEvent
|
| RTCErrorEventInit | |
| RTCErrorEventInit.Builder | |
| RTCErrorEventListener |
Listener for events of type RTCErrorEvent
|
| RTCErrorInit | |
| RTCErrorInit.Builder | |
| RTCIceCandidateInit |
The RTCIceCandidate() constructor creates and returns a new RTCIceCandidate object, which can be configured to represent a single ICE candidate.
|
| RTCIceCandidateInit.Builder |
The RTCIceCandidate() constructor creates and returns a new RTCIceCandidate object, which can be configured to represent a single ICE candidate.
|
| RTCIceCandidatePair |
The RTCIceCandidatePair dictionary describes a pair of ICE candidates which together comprise a description of a viable connection between two WebRTC endpoints.
|
| RTCIceCandidatePair.Builder |
The RTCIceCandidatePair dictionary describes a pair of ICE candidates which together comprise a description of a viable connection between two WebRTC endpoints.
|
| RTCIceCandidatePairStats |
The WebRTC RTCIceCandidatePairStats dictionary reports statistics which provide insight into the quality and performance of an RTCPeerConnection while connected and configured as described by the specified pair of ICE candidates.
|
| RTCIceCandidatePairStats.Builder |
The WebRTC RTCIceCandidatePairStats dictionary reports statistics which provide insight into the quality and performance of an RTCPeerConnection while connected and configured as described by the specified pair of ICE candidates.
|
| RTCIceCandidateStats |
The WebRTC API's RTCIceCandidateStats dictionary provides statistics related to an RTCIceCandidate.
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| RTCIceCandidateStats.Builder |
The WebRTC API's RTCIceCandidateStats dictionary provides statistics related to an RTCIceCandidate.
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| RTCIceParameters |
The RTCIceParameters dictionary specifies the username fragment and password assigned to an ICE session.
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| RTCIceParameters.Builder |
The RTCIceParameters dictionary specifies the username fragment and password assigned to an ICE session.
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| RTCIceServer |
The RTCIceServer dictionary defines how to connect to a single ICE server (such as a STUN or TURN server).
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| RTCIceServer.Builder |
The RTCIceServer dictionary defines how to connect to a single ICE server (such as a STUN or TURN server).
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| RTCIceServerStats | |
| RTCIceServerStats.Builder | |
| RTCIdentityAssertionResult | |
| RTCIdentityAssertionResult.Builder | |
| RTCIdentityProvider | |
| RTCIdentityProvider.Builder | |
| RTCIdentityProviderDetails | |
| RTCIdentityProviderDetails.Builder | |
| RTCIdentityProviderOptions | |
| RTCIdentityProviderOptions.Builder | |
| RTCIdentityValidationResult | |
| RTCIdentityValidationResult.Builder | |
| RTCInboundRtpStreamStats |
The WebRTC API's RTCInboundRtpStreamStats dictionary, based upon RTCReceivedRtpStreamStats and RTCStats, contains statistics related to the receiving end of an RTP stream on the local end of the RTCPeerConnection.
|
| RTCInboundRtpStreamStats.Builder |
The WebRTC API's RTCInboundRtpStreamStats dictionary, based upon RTCReceivedRtpStreamStats and RTCStats, contains statistics related to the receiving end of an RTP stream on the local end of the RTCPeerConnection.
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| RTCLocalSessionDescriptionInit | |
| RTCLocalSessionDescriptionInit.Builder | |
| RTCMediaHandlerStats | |
| RTCMediaHandlerStats.Builder | |
| RTCMediaSourceStats | |
| RTCMediaSourceStats.Builder | |
| RTCMediaStreamStats | |
| RTCMediaStreamStats.Builder | |
| RTCOfferAnswerOptions |
The WebRTC API's RTCOfferAnswerOptions dictionary is used to specify options that configure and control the process of creating WebRTC offers or answers.
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| RTCOfferAnswerOptions.Builder |
The WebRTC API's RTCOfferAnswerOptions dictionary is used to specify options that configure and control the process of creating WebRTC offers or answers.
|
| RTCOfferOptions |
The RTCOfferOptions dictionary is used to provide optional settings when creating an RTCPeerConnection offer with the createOffer() method.
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| RTCOfferOptions.Builder |
The RTCOfferOptions dictionary is used to provide optional settings when creating an RTCPeerConnection offer with the createOffer() method.
|
| RTCOutboundRtpStreamStats |
The RTCOutboundRtpStreamStats dictionary is the RTCStats-based object which provides metrics and statistics related to an outbound RTP stream being sent by an RTCRtpSender.
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| RTCOutboundRtpStreamStats.Builder |
The RTCOutboundRtpStreamStats dictionary is the RTCStats-based object which provides metrics and statistics related to an outbound RTP stream being sent by an RTCRtpSender.
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| RTCPeerConnectionErrorCallback | |
| RTCPeerConnectionIceErrorEventHandler |
Handle events of type RTCPeerConnectionIceErrorEvent
|
| RTCPeerConnectionIceErrorEventInit | |
| RTCPeerConnectionIceErrorEventInit.Builder | |
| RTCPeerConnectionIceErrorEventListener |
Listener for events of type RTCPeerConnectionIceErrorEvent
|
| RTCPeerConnectionIceEventHandler |
Handle events of type RTCPeerConnectionIceEvent
|
| RTCPeerConnectionIceEventInit | |
| RTCPeerConnectionIceEventInit.Builder | |
| RTCPeerConnectionIceEventListener |
Listener for events of type RTCPeerConnectionIceEvent
|
| RTCPeerConnectionStats | |
| RTCPeerConnectionStats.Builder | |
| RTCReceivedRtpStreamStats | |
| RTCReceivedRtpStreamStats.Builder | |
| RTCReceiverAudioTrackAttachmentStats | |
| RTCReceiverAudioTrackAttachmentStats.Builder | |
| RTCReceiverVideoTrackAttachmentStats | |
| RTCReceiverVideoTrackAttachmentStats.Builder | |
| RTCRemoteInboundRtpStreamStats | |
| RTCRemoteInboundRtpStreamStats.Builder | |
| RTCRemoteOutboundRtpStreamStats |
The WebRTC statistics model's RTCRemoteOutboundRtpStreamStats dictionary extends the underlying RTCSentRtpStreamStats dictionary with properties measuring metrics specific to outgoing RTP streams.
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| RTCRemoteOutboundRtpStreamStats.Builder |
The WebRTC statistics model's RTCRemoteOutboundRtpStreamStats dictionary extends the underlying RTCSentRtpStreamStats dictionary with properties measuring metrics specific to outgoing RTP streams.
|
| RTCRtcpParameters |
The RTCRtcpParameters dictionary provides parameters of an RTCP connection.
|
| RTCRtcpParameters.Builder |
The RTCRtcpParameters dictionary provides parameters of an RTCP connection.
|
| 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 RTCRtpCapabilities objects.
|
| RTCRtpCapabilities.Builder |
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 RTCRtpCapabilities objects.
|
| RTCRtpCodecCapability |
The WebRTC API's RTCRtpCodecCapability dictionary provides information describing the capabilities of a single media codec.
|
| RTCRtpCodecCapability.Builder |
The WebRTC API's RTCRtpCodecCapability dictionary provides information describing the capabilities of a single media codec.
|
| RTCRtpCodecParameters |
The RTCRtpCodecParameters dictionary, part of the WebRTC API, is used to describe the configuration parameters for a single media codec.
|
| RTCRtpCodecParameters.Builder |
The RTCRtpCodecParameters dictionary, part of the WebRTC API, is used to describe the configuration parameters for a single media codec.
|
| RTCRtpCodingParameters | |
| RTCRtpCodingParameters.Builder | |
| RTCRtpContributingSource |
The RTCRtpContributingSource 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.
|
| RTCRtpContributingSource.Builder |
The RTCRtpContributingSource 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.
|
| RTCRtpContributingSourceStats | |
| RTCRtpContributingSourceStats.Builder | |
| RTCRtpDecodingParameters | |
| RTCRtpDecodingParameters.Builder | |
| RTCRtpEncodingParameters |
An instance of the WebRTC API's RTCRtpEncodingParameters dictionary describes a single configuration of a codec for an RTCRtpSender.
|
| RTCRtpEncodingParameters.Builder |
An instance of the WebRTC API's RTCRtpEncodingParameters dictionary describes a single configuration of a codec for an RTCRtpSender.
|
| RTCRtpHeaderExtensionCapability | |
| RTCRtpHeaderExtensionCapability.Builder | |
| RTCRtpHeaderExtensionParameters | |
| RTCRtpHeaderExtensionParameters.Builder | |
| RTCRtpParameters |
The RTCRtpParameters dictionary is the basic object describing the parameters of an RTP transport.
|
| RTCRtpParameters.Builder |
The RTCRtpParameters dictionary is the basic object describing the parameters of an RTP transport.
|
| RTCRtpReceiveParameters |
The RTCRtpReceiveParameters dictionary, based upon the RTCRtpParameters dictionary, is returned by the RTCRtpReceiver method getParameters().
|
| RTCRtpReceiveParameters.Builder |
The RTCRtpReceiveParameters dictionary, based upon the RTCRtpParameters dictionary, is returned by the RTCRtpReceiver method getParameters().
|
| RTCRtpSendParameters |
The WebRTC API's RTCRtpSendParameters dictionary is used to specify the parameters for an RTCRtpSender when calling its setParameters() method.
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| RTCRtpSendParameters.Builder |
The WebRTC API's RTCRtpSendParameters dictionary is used to specify the parameters for an RTCRtpSender when calling its setParameters() method.
|
| RTCRtpStreamStats |
The RTCRtpStreamStats dictionary is returned by the RTCPeerConnection.getStats(), RTCRtpSender.getStats(), and RTCRtpReceiver.getStats() methods to provide detailed statistics about WebRTC connectivity.
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| RTCRtpStreamStats.Builder |
The RTCRtpStreamStats dictionary is returned by the RTCPeerConnection.getStats(), RTCRtpSender.getStats(), and RTCRtpReceiver.getStats() methods to provide detailed statistics about WebRTC connectivity.
|
| RTCRtpSynchronizationSource |
The RTCRtpSynchronizationSource dictionary of the WebRTC API is used by getSynchronizationSources() to describe a particular synchronization source (SSRC).
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| RTCRtpSynchronizationSource.Builder |
The RTCRtpSynchronizationSource dictionary of the WebRTC API is used by getSynchronizationSources() to describe a particular synchronization source (SSRC).
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| RTCRtpTransceiverInit |
The RTCRtpTransceiverInit dictionary is used when calling the WebRTC function RTCPeerConnection.addTransceiver() to provide configuration options for the new transceiver.
|
| RTCRtpTransceiverInit.Builder |
The RTCRtpTransceiverInit dictionary is used when calling the WebRTC function RTCPeerConnection.addTransceiver() to provide configuration options for the new transceiver.
|
| RTCRtpTransceiverStats | |
| RTCRtpTransceiverStats.Builder | |
| RTCSctpTransportStats | |
| RTCSctpTransportStats.Builder | |
| RTCSenderAudioTrackAttachmentStats | |
| RTCSenderAudioTrackAttachmentStats.Builder | |
| RTCSenderVideoTrackAttachmentStats | |
| RTCSenderVideoTrackAttachmentStats.Builder | |
| RTCSentRtpStreamStats | |
| RTCSentRtpStreamStats.Builder | |
| RTCSessionDescriptionCallback | |
| RTCSessionDescriptionInit | |
| RTCSessionDescriptionInit.Builder | |
| RTCStats |
The RTCStats dictionary is the basic statistics object used by WebRTC's statistics monitoring model, providing the properties required of all statistics data objects.
|
| RTCStats.Builder |
The RTCStats dictionary is the basic statistics object used by WebRTC's statistics monitoring model, providing the properties required of all statistics data objects.
|
| RTCStatsReport.ForEachCallback | |
| RTCStatsReport.ForEachCallback2 | |
| RTCStatsReport.ForEachCallback3 | |
| RTCTrackEventHandler |
Handle events of type RTCTrackEvent
|
| RTCTrackEventInit |
The WebRTC API's RTCTrackEventInit dictionary is used to provide information describing an RTCTrackEvent when instantiating a new track event using new RTCTrackEvent().
|
| RTCTrackEventInit.Builder |
The WebRTC API's RTCTrackEventInit dictionary is used to provide information describing an RTCTrackEvent when instantiating a new track event using new RTCTrackEvent().
|
| RTCTrackEventListener |
Listener for events of type RTCTrackEvent
|
| RTCTransportStats | |
| RTCTransportStats.Builder | |
| RTCVideoHandlerStats | |
| RTCVideoHandlerStats.Builder | |
| RTCVideoReceiverStats | |
| RTCVideoReceiverStats.Builder | |
| RTCVideoSenderStats | |
| RTCVideoSenderStats.Builder | |
| RTCVideoSourceStats | |
| RTCVideoSourceStats.Builder | |
| ValidateAssertionCallback |
| Class | Description |
|---|---|
| RTCBundlePolicy.Util | |
| RTCCertificate |
The interface of the WebRTC API provides an object represents a certificate that an RTCPeerConnection uses to authenticate.
|
| RTCCodecType.Util | |
| RTCDataChannel |
The RTCDataChannel interface represents a network channel which can be used for bidirectional peer-to-peer transfers of arbitrary data.
|
| RTCDataChannelEvent |
The RTCDataChannelEvent interface represents an event related to a specific RTCDataChannel.
|
| RTCDataChannelState.Util | |
| RTCDtlsTransport |
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.
|
| RTCDtlsTransportState.Util | |
| RTCDTMFSender |
The RTCDTMFSender interface provides a mechanism for transmitting DTMF codes on a WebRTC RTCPeerConnection.
|
| RTCDTMFToneChangeEvent |
The RTCDTMFToneChangeEvent interface represents events sent to indicate that DTMF tones have started or finished playing.
|
| RTCError |
The RTCError interface describes an error which has occurred while handling WebRTC operations.
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| RTCErrorDetailType.Util | |
| RTCErrorDetailTypeIdp.Util | |
| RTCErrorEvent |
The WebRTC API's RTCErrorEvent interface represents an error sent to a WebRTC object.
|
| RTCIceCandidate |
The RTCIceCandidate interface—part of the WebRTC API—represents a candidate Interactive Connectivity Establishment (ICE) configuration which may be used to establish an RTCPeerConnection.
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| RTCIceCandidateType.Util | |
| RTCIceComponent.Util | |
| RTCIceConnectionState.Util | |
| RTCIceCredentialType.Util | |
| RTCIceGathererState.Util | |
| RTCIceGatheringState.Util | |
| RTCIceProtocol.Util | |
| RTCIceRole.Util | |
| RTCIceTcpCandidateType.Util | |
| RTCIceTransport |
The RTCIceTransport interface provides access to information about the ICE transport layer over which the data is being sent and received.
|
| RTCIceTransportPolicy.Util | |
| RTCIceTransportState.Util | |
| RTCIdentityAssertion |
The RTCIdentityAssertion interface of the WebRTC API represents the identity of the a remote peer of the current connection.
|
| RTCIdentityProviderGlobalScope | |
| RTCIdentityProviderRegistrar | |
| RTCPeerConnection |
The RTCPeerConnection interface represents a WebRTC connection between the local computer and a remote peer.
|
| 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.
|
| RTCPeerConnectionIceEvent |
The RTCPeerConnectionIceEvent interface represents events that occurs in relation to ICE candidates with the target, usually an RTCPeerConnection.
|
| RTCPeerConnectionState.Util | |
| RTCPriorityType.Util | |
| RTCQualityLimitationReason.Util | |
| RTCRtcpMuxPolicy.Util | |
| RTCRtpReceiver |
The RTCRtpReceiver interface of the WebRTC API manages the reception and decoding of data for a MediaStreamTrack on an RTCPeerConnection.
|
| RTCRtpSender |
The RTCRtpSender interface provides the ability to control and obtain details about how a particular MediaStreamTrack is encoded and sent to a remote peer.
|
| RTCRtpTransceiver |
The WebRTC interface RTCRtpTransceiver describes a permanent pairing of an RTCRtpSender and an RTCRtpReceiver, along with some shared state.
|
| RTCRtpTransceiverDirection.Util | |
| RTCSctpTransport |
The RTCSctpTransport interface provides information which describes a Stream Control Transmission Protocol (SCTP) transport.
|
| RTCSctpTransportState.Util | |
| RTCSdpType.Util | |
| RTCSessionDescription |
The RTCSessionDescription interface describes one end of a connection—or potential connection—and how it's configured.
|
| RTCSignalingState.Util | |
| RTCStatsIceCandidatePairState.Util | |
| RTCStatsReport |
The RTCStatsReport interface provides a statistics report obtained by calling one of the RTCPeerConnection.getStats(), RTCRtpReceiver.getStats(), and RTCRtpSender.getStats() methods.
|
| RTCStatsReport.Entry | |
| RTCStatsType.Util | |
| 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.
|