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 `rtc.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 `rtc.RTCDataChannelEvent`
which represents a `web.datachannel` event. These events sent
an `rtc.RTCPeerConnection` when its remote peer is asking to
an `rtc.RTCDataChannel` between the two peers.

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

Listen to these events using `web.addEventListener()` or by assigning
event listener to the oneventname property of this interface.

RTCDTMFSender Events.

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 `rtc.RTCPeerConnection`.

The WebRTC API's `rtc.RTCIceCandidateInit` dictionary, which
the information needed to fundamentally describe an `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 `rtc.RTCPeerConnection`
connected and configured as described by the specified pair of
candidates.

The WebRTC API's RTCIceCandidateStats dictionary provides statistics
to an `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 RTCIceTransport interface provides access to information
the ICE transport layer over which the data is being sent and

RTCIceTransport Events.

The RTCIdentityErrorEvent interface represents an error associated
the identity provider (idP). This is usually for an `rtc.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 `rtc.RTCPeerConnection`.
only event sent with this type is identityresult..

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

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

The RTCPeerConnection interface represents a WebRTC connection
the local computer and a remote peer. It provides methods to
to a remote peer, maintain and monitor the connection, and close
connection once it's no longer needed.

RTCPeerConnection Events.

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

The RTCRtpContributingSource dictionary of the the WebRTC API
used by `web.getContributingSources()` to provide information
a given contributing source (CSRC), including the most recent
a 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 `rtc.RTCRtpSender`.

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

The RTCRtpSender interface provides the ability to control and
details about how a particular `media.MediaStreamTrack` is encoded
sent to a remote peer.

The WebRTC API's RTCRtpSendParameters dictionary is used to specify
parameters for an `rtc.RTCRtpSender` when calling its `web.setParameters()`

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

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

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

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 `rtc.RTCPeerConnection`'s data
are sent and received.

The RTCSessionDescription interface describes one end of a connection—or
connection—and how it's configured. Each RTCSessionDescription
of a description `web.type` indicating which part of the offer/answer
process it describes and of the SDP descriptor of the session.

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 `media.MediaStreamTrack` is added to an `rtc.RTCRtpReceiver`
is part of the `rtc.RTCPeerConnection`.

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