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web.rtc.ConstrainDouble

The ConstrainDouble type is used to specify a constraint for property whose value is a double-precision floating-point number. extends the web.streams.DoubleRange dictionary (which provides ability to specify a permitted range of property values) to also an exact value and/or an ideal value the property should take Additionally, you can specify the property's value as a simple value, in which case the user agent does its best to match the once all other more stringent constraints are met.

The ConstrainDouble type is used to specify a constraint for
property whose value is a double-precision floating-point number.
extends the `web.streams.DoubleRange` dictionary (which provides
ability to specify a permitted range of property values) to also
an exact value and/or an ideal value the property should take
Additionally, you can specify the property's value as a simple
value, in which case the user agent does its best to match the
once all other more stringent constraints are met.
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web.rtc.core

web.rtc interfaces.

web.rtc interfaces.
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No vars found in this namespace.

web.rtc.LocalMediaStream

Do not use LocalMediaStream; you need to update any code that use it as soon as possible or your content or application will working. See Stopping a video stream in MediaStreamTrack to learn

Do not use LocalMediaStream; you need to update any code that
use it as soon as possible or your content or application will
working. See Stopping a video stream in MediaStreamTrack to learn
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web.rtc.MediaStreamEvent

The MediaStreamEvent interface represents events that occurs relation to a web.streams.MediaStream. Two events of this type be thrown: addstream and removestream.

The MediaStreamEvent interface represents events that occurs
relation to a `web.streams.MediaStream`. Two events of this type
be thrown: addstream and removestream.
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web.rtc.RTCCertificate

The interface of the the WebRTC API provides an object represents certificate that an web.audio.RTCPeerConnection uses to authenticate.

The interface of the the WebRTC API provides an object represents
certificate that an `web.audio.RTCPeerConnection` uses to authenticate.
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web.rtc.RTCConfiguration

The RTCConfiguration dictionary is used to provide configuration for an web.audio.RTCPeerConnection. It may be passed into the when instantiating a connection, or used with the RTCPeerConnection.getConfiguration() RTCPeerConnection.setConfiguration() methods, which allow inspecting changing the configuration while a connection is established.

The RTCConfiguration dictionary is used to provide configuration
for an `web.audio.RTCPeerConnection`. It may be passed into the
when instantiating a connection, or used with the `RTCPeerConnection.getConfiguration()`
`RTCPeerConnection.setConfiguration()` methods, which allow inspecting
changing the configuration while a connection is established.
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web.rtc.RTCDataChannel

The RTCDataChannel interface represents a network channel which be used for bidirectional peer-to-peer transfers of arbitrary Every data channel is associated with an web.audio.RTCPeerConnection, each peer connection can have up to a theoretical maximum of data channels (the actual limit may vary from browser to browser).

The RTCDataChannel interface represents a network channel which
be used for bidirectional peer-to-peer transfers of arbitrary
Every data channel is associated with an `web.audio.RTCPeerConnection`,
each peer connection can have up to a theoretical maximum of
data channels (the actual limit may vary from browser to browser).
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web.rtc.RTCDataChannelEvent

The RTCDataChannelEvent() constructor returns a new web.rtc.RTCDataChannelEvent which represents a datachannel event. These events sent to web.audio.RTCPeerConnection when its remote peer is asking open an web.rtc.RTCDataChannel between the two peers.

The RTCDataChannelEvent() constructor returns a new `web.rtc.RTCDataChannelEvent`
which represents a `datachannel` event. These events sent to
`web.audio.RTCPeerConnection` when its remote peer is asking
open an `web.rtc.RTCDataChannel` between the two peers.
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web.rtc.RTCDtlsTransport

The RTCDtlsTransport interface provides information which describes Datagram Transport Layer Security (DTLS) transport.

The RTCDtlsTransport interface provides information which describes
Datagram Transport Layer Security (DTLS) transport.
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web.rtc.RTCDTMFToneChangeEvent

The RTCDTMFToneChangeEvent interface represents events sent to that DTMF tones have started or finished playing. This interface used by the tonechange event.

The RTCDTMFToneChangeEvent interface represents events sent to
that DTMF tones have started or finished playing. This interface
used by the tonechange event.
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web.rtc.RTCIceCandidate

The RTCIceCandidate interface—part of the WebRTC API—represents candidate Internet Connectivity Establishment (ICE) configuration may be used to establish an web.audio.RTCPeerConnection.

The RTCIceCandidate interface—part of the WebRTC API—represents
candidate Internet Connectivity Establishment (ICE) configuration
may be used to establish an `web.audio.RTCPeerConnection`.
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web.rtc.RTCIceCandidateInit

The WebRTC API's web.rtc.RTCIceCandidateInit dictionary, which the information needed to fundamentally describe an web.rtc.RTCIceCandidate.

The WebRTC API's `web.rtc.RTCIceCandidateInit` dictionary, which
the information needed to fundamentally describe an `web.rtc.RTCIceCandidate`.
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web.rtc.RTCIceCandidatePair

The RTCIceCandidatePair dictionary describes a pair of ICE candidates together comprise a description of a viable connection between WebRTC endpoints.

The RTCIceCandidatePair dictionary describes a pair of ICE candidates
together comprise a description of a viable connection between
WebRTC endpoints.
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web.rtc.RTCIceCandidatePairStats

The WebRTC RTCIceCandidatePairStats dictionary reports statistics provide insight into the quality and performance of an web.audio.RTCPeerConnection connected and configured as described by the specified pair of candidates.

The WebRTC RTCIceCandidatePairStats dictionary reports statistics
provide insight into the quality and performance of an `web.audio.RTCPeerConnection`
connected and configured as described by the specified pair of
candidates.
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web.rtc.RTCIceParameters

The RTCIceParameters dictionary specifies the username fragment password assigned to an ICE session.

The RTCIceParameters dictionary specifies the username fragment
password assigned to an ICE session.
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web.rtc.RTCIceServer

The RTCIceServer dictionary defines how to connect to a single server (such as a STUN or TURN server). It includes both the and the necessary credentials, if any, to connect to the server.

The RTCIceServer dictionary defines how to connect to a single
server (such as a STUN or TURN server). It includes both the
and the necessary credentials, if any, to connect to the server.
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web.rtc.RTCIdentityAssertion

The RTCIdentityAssertion interface of the the WebRTC API represents identity of the a remote peer of the current connection. If no has yet been set and verified this interface returns null. Once it can't be changed.

The RTCIdentityAssertion interface of the the WebRTC API represents
identity of the a remote peer of the current connection. If no
has yet been set and verified this interface returns null. Once
it can't be changed.
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web.rtc.RTCIdentityErrorEvent

The RTCIdentityErrorEvent interface represents an error associated the identity provider (idP). This is usually for an web.audio.RTCPeerConnection. events are sent with this type: idpassertionerror and idpvalidationerror.

The RTCIdentityErrorEvent interface represents an error associated
the identity provider (idP). This is usually for an `web.audio.RTCPeerConnection`.
events are sent with this type: idpassertionerror and idpvalidationerror.
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web.rtc.RTCIdentityEvent

The RTCIdentityEvent interface represents an identity assertion by an identity provider (idP). This is usually for an web.audio.RTCPeerConnection. only event sent with this type is identityresult..

The RTCIdentityEvent interface represents an identity assertion
by an identity provider (idP). This is usually for an `web.audio.RTCPeerConnection`.
only event sent with this type is identityresult..
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web.rtc.RTCOfferAnswerOptions

The WebRTC API's RTCOfferAnswerOptions dictionary is used to options that configure and control the process of creating WebRTC or answers.

The WebRTC API's RTCOfferAnswerOptions dictionary is used to
options that configure and control the process of creating WebRTC
or answers.
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web.rtc.RTCOfferOptions

The RTCOfferOptions dictionary is used to provide optional settings creating an web.audio.RTCPeerConnection offer with the createOffer()

The RTCOfferOptions dictionary is used to provide optional settings
creating an `web.audio.RTCPeerConnection` offer with the `createOffer()`
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web.rtc.RTCPeerConnectionIceEvent

The RTCPeerConnectionIceEvent interface represents events that in relation to ICE candidates with the target, usually an web.audio.RTCPeerConnection.

The RTCPeerConnectionIceEvent interface represents events that
in relation to ICE candidates with the target, usually an `web.audio.RTCPeerConnection`.
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web.rtc.RTCRtpCodecParameters

The web.rtc.RTCRtpCodecParameters dictionary, part of the WebRTC is used to describe the configuration parameters for a single codec.

The `web.rtc.RTCRtpCodecParameters` dictionary, part of the WebRTC
is used to describe the configuration parameters for a single
codec.
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web.rtc.RTCRtpContributingSource

The RTCRtpContributingSource dictionary of the the WebRTC API used by getContributingSources() to provide information about given contributing source (CSRC), including the most recent time packet that the source contributed was played out.

The RTCRtpContributingSource dictionary of the the WebRTC API
used by `getContributingSources()` to provide information about
given contributing source (CSRC), including the most recent time
packet that the source contributed was played out.
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web.rtc.RTCRtpEncodingParameters

An instance of the WebRTC API's RTCRtpEncodingParameters dictionary a single configuration of a codec for an web.audio.RTCRtpSender.

An instance of the WebRTC API's RTCRtpEncodingParameters dictionary
a single configuration of a codec for an `web.audio.RTCRtpSender`.
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web.rtc.RTCRtpReceiver

The RTCRtpReceiver interface of the WebRTC API manages the reception decoding of data for a web.audio.MediaStreamTrack on an web.audio.RTCPeerConnection.

The RTCRtpReceiver interface of the WebRTC API manages the reception
decoding of data for a `web.audio.MediaStreamTrack` on an `web.audio.RTCPeerConnection`.
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web.rtc.RTCRtpStreamStats

The web.rtc.RTCRtpStreamStats dictionary is returned by the RTCRtpSender.getStats(), and RTCRtpReceiver.getStats() methods provide detailed statistics about WebRTC connectivity.

The `web.rtc.RTCRtpStreamStats` dictionary is returned by the
`RTCRtpSender.getStats()`, and `RTCRtpReceiver.getStats()` methods
provide detailed statistics about WebRTC connectivity.
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web.rtc.RTCRtpSynchronizationSource

The RTCRtpSynchronizationSource dictionary of the the WebRTC is used by getSynchronizationSources() to describe a particular source (SSRC).

The RTCRtpSynchronizationSource dictionary of the the WebRTC
is used by `getSynchronizationSources()` to describe a particular
source (SSRC).
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web.rtc.RTCRtpTransceiver

The WebRTC interface RTCRtpTransceiver describes a permanent of an web.audio.RTCRtpSender and an web.rtc.RTCRtpReceiver, with some shared state.

The WebRTC interface RTCRtpTransceiver describes a permanent
of an `web.audio.RTCRtpSender` and an `web.rtc.RTCRtpReceiver`,
with some shared state.
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web.rtc.RTCRtpTransceiverInit

The RTCRtpTransceiverInit dictionary is used when calling the function RTCPeerConnection.addTransceiver() to provide configuration for the new transceiver.

The RTCRtpTransceiverInit dictionary is used when calling the
function `RTCPeerConnection.addTransceiver()` to provide configuration
for the new transceiver.
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web.rtc.RTCSctpTransport

The RTCSctpTransport interface provides information which describes Stream Control Transmission Protocol (SCTP) transport. This provides about limitations of the transport, but also provides a way to the underlying Datagram Transport Layer Security (DTLS) transport which SCTP packets for all of an web.audio.RTCPeerConnection's channels are sent and received.

The RTCSctpTransport interface provides information which describes
Stream Control Transmission Protocol (SCTP) transport. This provides
about limitations of the transport, but also provides a way to
the underlying Datagram Transport Layer Security (DTLS) transport
which SCTP packets for all of an `web.audio.RTCPeerConnection`'s
channels are sent and received.
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web.rtc.RTCStats

The RTCStats dictionary is the basic statistics object used by statistics monitoring model, providing the properties required all statistics data objects.

The RTCStats dictionary is the basic statistics object used by
statistics monitoring model, providing the properties required
all statistics data objects.
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web.rtc.RTCTrackEvent

The WebRTC API interface RTCTrackEvent represents the track event, is sent when a new web.audio.MediaStreamTrack is added to an which is part of the web.audio.RTCPeerConnection.

The WebRTC API interface RTCTrackEvent represents the track event,
is sent when a new `web.audio.MediaStreamTrack` is added to an
which is part of the `web.audio.RTCPeerConnection`.
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web.rtc.RTCTrackEventInit

The WebRTC API's RTCTrackEventInit dictionary is used to provide describing an web.rtc.RTCTrackEvent when instantiating a new event using new RTCTrackEvent().

The WebRTC API's RTCTrackEventInit dictionary is used to provide
describing an `web.rtc.RTCTrackEvent` when instantiating a new
event using `new RTCTrackEvent()`.
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