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.

web.rtc interfaces.

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

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

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

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

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

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

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

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 API's RTCIceCandidateStats dictionary provides statistics
to an `web.rtc.RTCIceCandidate`.

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

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 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 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 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 WebRTC API's RTCInboundRtpStreamStats dictionary, based upon
and `web.rtc.RTCStats`, contains statistics related to the receiving
of an RTP stream on the local end of the `web.audio.RTCPeerConnection`.

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

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

The RTCOutboundRtpStreamStats dictionary is the `web.rtc.RTCStats`-based
which provides metrics and statistics related to an outbound
stream being sent by an `web.audio.RTCRtpSender`.

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

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

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.

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

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

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

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

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

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

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

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