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.
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.
(add-ice-candidate this & args)
Method.
When a web site or app using web.audio.RTCPeerConnection
receives
new ICE candidate from the remote peer over its signaling channel,
delivers the newly-received candidate to the browser's ICE agent
calling RTCPeerConnection.addIceCandidate().
`aPromise = pc.addIceCandidate(candidate);
addIceCandidate(candidate, successCallback, failureCallback);`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addIceCandidate
Method. When a web site or app using `web.audio.RTCPeerConnection` receives new ICE candidate from the remote peer over its signaling channel, delivers the newly-received candidate to the browser's ICE agent calling RTCPeerConnection.addIceCandidate(). `aPromise = pc.addIceCandidate(candidate); addIceCandidate(candidate, successCallback, failureCallback);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addIceCandidate`
(add-stream this media-stream)
Method.
[Deprecated]
The obsolete web.audio.RTCPeerConnection
method addStream()
a web.streams.MediaStream
as a local source of audio or video.
of using this obsolete method, you should instead use addTrack()
for each track you wish to send to the remote peer.
rtcPeerConnection.addStream(mediaStream);
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream
Method. [Deprecated] The obsolete `web.audio.RTCPeerConnection` method addStream() a `web.streams.MediaStream` as a local source of audio or video. of using this obsolete method, you should instead use `addTrack()` for each track you wish to send to the remote peer. `rtcPeerConnection.addStream(mediaStream);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream`
(add-track this & args)
Method.
The web.audio.RTCPeerConnection
method addTrack() adds a new
track to the set of tracks which will be transmitted to the other
rtpSender = rtcPeerConnection.addTrack(track, stream...);
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addTrack
Method. The `web.audio.RTCPeerConnection` method addTrack() adds a new track to the set of tracks which will be transmitted to the other `rtpSender = rtcPeerConnection.addTrack(track, stream...);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addTrack`
(can-trickle-ice-candidates this)
Property.
[Read Only]
The read-only web.audio.RTCPeerConnection
property canTrickleIceCandidates
a js.Boolean
which indicates whether or not the remote peer
accept trickled ICE candidates.
var canTrickle = RTCPeerConnection.canTrickleIceCandidates;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/canTrickleIceCandidates
Property. [Read Only] The read-only `web.audio.RTCPeerConnection` property canTrickleIceCandidates a `js.Boolean` which indicates whether or not the remote peer accept trickled ICE candidates. `var canTrickle = RTCPeerConnection.canTrickleIceCandidates;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/canTrickleIceCandidates`
(close this & args)
Method.
[Experimental]
The RTCPeerConnection.close() method closes the current peer
`peerConnection.close();
This method has no parameters, and returns nothing.
Calling this method terminates the RTCPeerConnection's ICE agent, ending any ongoing ICE processing and any active streams. This also releases any resources in use by the ICE agent, including TURN permissions. All web.audio.RTCRtpSender
objects are considered to be stopped once this returns (they may still be in the process of stopping, but for all intents and purposes, they're stopped).
Once this method returns, the signaling state as returned by RTCPeerConnection.signalingState
is closed.
Make sure that you delete all references to the previous web.audio.RTCPeerConnection
before attempting to create a new one that connects to the same remote peer, as not doing so might result in some errors depending on the browser.`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/close
Method. [Experimental] The RTCPeerConnection.close() method closes the current peer `peerConnection.close(); This method has no parameters, and returns nothing. Calling this method terminates the RTCPeerConnection's ICE agent, ending any ongoing ICE processing and any active streams. This also releases any resources in use by the ICE agent, including TURN permissions. All `web.audio.RTCRtpSender` objects are considered to be stopped once this returns (they may still be in the process of stopping, but for all intents and purposes, they're stopped). Once this method returns, the signaling state as returned by `RTCPeerConnection.signalingState` is closed. Make sure that you delete all references to the previous `web.audio.RTCPeerConnection` before attempting to create a new one that connects to the same remote peer, as not doing so might result in some errors depending on the browser.` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/close`
(connection-state this)
Property.
[Read Only]
The read-only connectionState property of the web.audio.RTCPeerConnection
indicates the current state of the peer connection by returning
of the string values specified by the enum RTCPeerConnectionState.
var connectionState = RTCPeerConnection.connectionState;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/connectionState
Property. [Read Only] The read-only connectionState property of the `web.audio.RTCPeerConnection` indicates the current state of the peer connection by returning of the string values specified by the enum RTCPeerConnectionState. `var connectionState = RTCPeerConnection.connectionState;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/connectionState`
(create-answer this & args)
Method.
The createAnswer() method on the web.audio.RTCPeerConnection
creates an SDP answer to an offer received from a remote peer
the offer/answer negotiation of a WebRTC connection. The answer
information about any media already attached to the session,
and options supported by the browser, and any ICE candidates
gathered. The answer is delivered to the returned js.Promise
,
should then be sent to the source of the offer to continue the
process.
`aPromise = RTCPeerConnection.createAnswer([options]);
RTCPeerConnection.createAnswer(successCallback, failureCallback[, options]);`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createAnswer
Method. The createAnswer() method on the `web.audio.RTCPeerConnection` creates an SDP answer to an offer received from a remote peer the offer/answer negotiation of a WebRTC connection. The answer information about any media already attached to the session, and options supported by the browser, and any ICE candidates gathered. The answer is delivered to the returned `js.Promise`, should then be sent to the source of the offer to continue the process. `aPromise = RTCPeerConnection.createAnswer([options]); RTCPeerConnection.createAnswer(successCallback, failureCallback[, options]);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createAnswer`
(create-data-channel this & args)
Method.
The createDataChannel() method on the web.audio.RTCPeerConnection
creates a new channel linked with the remote peer, over which
kind of data may be transmitted.
dataChannel = RTCPeerConnection.createDataChannel(label[, options]);
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createDataChannel
Method. The createDataChannel() method on the `web.audio.RTCPeerConnection` creates a new channel linked with the remote peer, over which kind of data may be transmitted. `dataChannel = RTCPeerConnection.createDataChannel(label[, options]);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createDataChannel`
(create-offer this & args)
Method.
The createOffer() method of the web.audio.RTCPeerConnection
initiates the creation of an SDP offer for the purpose of starting
new WebRTC connection to a remote peer.
`aPromise = myPeerConnection.createOffer([options]);
myPeerConnection.createOffer(successCallback, failureCallback, [options])`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createOffer
Method. The createOffer() method of the `web.audio.RTCPeerConnection` initiates the creation of an SDP offer for the purpose of starting new WebRTC connection to a remote peer. `aPromise = myPeerConnection.createOffer([options]); myPeerConnection.createOffer(successCallback, failureCallback, [options])` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createOffer`
(current-local-description this)
Property.
[Read Only]
The read-only property RTCPeerConnection.currentLocalDescription
an web.audio.RTCSessionDescription
object describing the local
of the connection as it was most recently successfully negotiated
the last time the web.audio.RTCPeerConnection
finished negotiating
connecting to a remote peer. Also included is a list of any ICE
that may already have been generated by the ICE agent since the
or answer represented by the description was first instantiated.
sessionDescription = RTCPeerConnection.currentLocalDescription;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/currentLocalDescription
Property. [Read Only] The read-only property RTCPeerConnection.currentLocalDescription an `web.audio.RTCSessionDescription` object describing the local of the connection as it was most recently successfully negotiated the last time the `web.audio.RTCPeerConnection` finished negotiating connecting to a remote peer. Also included is a list of any ICE that may already have been generated by the ICE agent since the or answer represented by the description was first instantiated. `sessionDescription = RTCPeerConnection.currentLocalDescription;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/currentLocalDescription`
(current-remote-description this)
Property.
[Read Only]
The read-only property RTCPeerConnection.currentRemoteDescription
an web.audio.RTCSessionDescription
object describing the remote
of the connection as it was most recently successfully negotiated
the last time the web.audio.RTCPeerConnection
finished negotiating
connecting to a remote peer. Also included is a list of any ICE
that may already have been generated by the ICE agent since the
or answer represented by the description was first instantiated.
sessionDescription = RTCPeerConnection.currentRemoteDescription;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/currentRemoteDescription
Property. [Read Only] The read-only property RTCPeerConnection.currentRemoteDescription an `web.audio.RTCSessionDescription` object describing the remote of the connection as it was most recently successfully negotiated the last time the `web.audio.RTCPeerConnection` finished negotiating connecting to a remote peer. Also included is a list of any ICE that may already have been generated by the ICE agent since the or answer represented by the description was first instantiated. `sessionDescription = RTCPeerConnection.currentRemoteDescription;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/currentRemoteDescription`
(default-ice-servers this)
Property.
[Read Only] [Experimental]
The read-only property RTCPeerConnection.defaultIceServers returns
array of objects based on the web.rtc.RTCIceServer
dictionary,
indicates what, if any, ICE servers the browser will use by default
none are provided to the web.audio.RTCPeerConnection
in its
However, browsers are not required to provide any default ICE
at all.
var defaultIceServers = RTCPeerConnection.defaultIceServers;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/defaultIceServers
Property. [Read Only] [Experimental] The read-only property RTCPeerConnection.defaultIceServers returns array of objects based on the `web.rtc.RTCIceServer` dictionary, indicates what, if any, ICE servers the browser will use by default none are provided to the `web.audio.RTCPeerConnection` in its However, browsers are not required to provide any default ICE at all. `var defaultIceServers = RTCPeerConnection.defaultIceServers;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/defaultIceServers`
(generate-certificate this keygen-algorithm)
Method.
The generateCertificate() method of the web.audio.RTCPeerConnection
creates and stores an X.509 certificate and corresponding private
then returns an web.rtc.RTCCertificate
, providing access to
var cert = RTCPeerConnection.generateCertificate(keygenAlgorithm)
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/generateCertificate
Method. The generateCertificate() method of the `web.audio.RTCPeerConnection` creates and stores an X.509 certificate and corresponding private then returns an `web.rtc.RTCCertificate`, providing access to `var cert = RTCPeerConnection.generateCertificate(keygenAlgorithm)` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/generateCertificate`
(get-configuration this)
Method.
[Experimental]
The RTCPeerConnection.getConfiguration() method returns an web.rtc.RTCConfiguration
which indicates the current configuration of the web.audio.RTCPeerConnection
which the method is called.
var configuration = RTCPeerConnection.getConfiguration();
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getConfiguration
Method. [Experimental] The RTCPeerConnection.getConfiguration() method returns an `web.rtc.RTCConfiguration` which indicates the current configuration of the `web.audio.RTCPeerConnection` which the method is called. `var configuration = RTCPeerConnection.getConfiguration();` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getConfiguration`
(get-identity-assertion this & args)
Method.
[Experimental]
The RTCPeerConnection.getIdentityAssertion() method initiates
gathering of an identity assertion. This has an effect only if
signalingState
is not "closed".
`pc.getIdentityAssertion();
There is neither parameter nor return value for this method.`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getIdentityAssertion
Method. [Experimental] The RTCPeerConnection.getIdentityAssertion() method initiates gathering of an identity assertion. This has an effect only if `signalingState` is not \"closed\". `pc.getIdentityAssertion(); There is neither parameter nor return value for this method.` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getIdentityAssertion`
(get-local-streams this & args)
Method.
[Deprecated] [Experimental]
The RTCPeerConnection.getLocalStreams() method returns an array MediaStream associated with the local end of the connection. array may be empty.
`var mediaStreams[] = pc.getLocalStreams();
The return value is an js.Array
of web.streams.MediaStream
objects.`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getLocalStreams
Method. [Deprecated] [Experimental] The RTCPeerConnection.getLocalStreams() method returns an array MediaStream associated with the local end of the connection. array may be empty. `var mediaStreams[] = pc.getLocalStreams(); The return value is an `js.Array` of `web.streams.MediaStream` objects.` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getLocalStreams`
(get-receivers this)
Method.
[Experimental]
The RTCPeerConnection.getReceivers() method returns an array
web.rtc.RTCRtpReceiver
objects, each of which represents one
receiver. Each RTP receiver manages the reception and decoding
data for a web.audio.MediaStreamTrack
on an web.audio.RTCPeerConnection
var receivers[] = RTCPeerConnection.getReceivers();
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getReceivers
Method. [Experimental] The RTCPeerConnection.getReceivers() method returns an array `web.rtc.RTCRtpReceiver` objects, each of which represents one receiver. Each RTP receiver manages the reception and decoding data for a `web.audio.MediaStreamTrack` on an `web.audio.RTCPeerConnection` `var receivers[] = RTCPeerConnection.getReceivers();` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getReceivers`
(get-remote-streams this & args)
Method.
[Deprecated] [Experimental]
The RTCPeerConnection.getRemoteStreams() method returns an array MediaStream associated with the remote end of the connection. array may be empty.
`var mediaStreams[] = pc.getRemoteStreams();
The return value is an js.Array
of web.streams.MediaStream
objects.`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getRemoteStreams
Method. [Deprecated] [Experimental] The RTCPeerConnection.getRemoteStreams() method returns an array MediaStream associated with the remote end of the connection. array may be empty. `var mediaStreams[] = pc.getRemoteStreams(); The return value is an `js.Array` of `web.streams.MediaStream` objects.` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getRemoteStreams`
(get-senders this)
Method.
The web.audio.RTCPeerConnection
method getSenders() returns
array of web.audio.RTCRtpSender
objects, each of which represents
RTP sender responsible for transmitting one track's data.
var senders = RTCPeerConnection.getSenders();
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getSenders
Method. The `web.audio.RTCPeerConnection` method getSenders() returns array of `web.audio.RTCRtpSender` objects, each of which represents RTP sender responsible for transmitting one track's data. `var senders = RTCPeerConnection.getSenders();` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getSenders`
(get-stats this selector)
Method.
The web.audio.RTCPeerConnection
method getStats() returns a
which resolves with data providing statistics about either the
connection or about the specified web.audio.MediaStreamTrack
.
promise = rtcPeerConnection.getStats(selector)
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getStats
Method. The `web.audio.RTCPeerConnection` method getStats() returns a which resolves with data providing statistics about either the connection or about the specified `web.audio.MediaStreamTrack`. `promise = rtcPeerConnection.getStats(selector)` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getStats`
(get-stream-by-id this id)
Method.
[Experimental] [Obsolute]
The RTCPeerConnection.getStreamById() method returns the MediaStream the given id that is associated with local or remote end of the If no stream matches, it returns null.
var mediaStream = pc.getStream(id);
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getStreamById
Method. [Experimental] [Obsolute] The RTCPeerConnection.getStreamById() method returns the MediaStream the given id that is associated with local or remote end of the If no stream matches, it returns null. `var mediaStream = pc.getStream(id);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getStreamById`
(get-transceivers this)
Method.
The web.audio.RTCPeerConnection
interface's getTransceivers()
returns a list of the web.rtc.RTCRtpTransceiver
objects being
to send and receive data on the connection.
transceiverList = rtcPeerConnection.getTransceivers();
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getTransceivers
Method. The `web.audio.RTCPeerConnection` interface's getTransceivers() returns a list of the `web.rtc.RTCRtpTransceiver` objects being to send and receive data on the connection. `transceiverList = rtcPeerConnection.getTransceivers();` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getTransceivers`
(ice-connection-state this)
Property.
[Read Only] [Experimental]
The read-only property RTCPeerConnection.iceConnectionState returns
enum of type RTCIceConnectionState which state of the ICE agent
with the web.audio.RTCPeerConnection
.
var state = RTCPeerConnection.iceConnectionState;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceConnectionState
Property. [Read Only] [Experimental] The read-only property RTCPeerConnection.iceConnectionState returns enum of type RTCIceConnectionState which state of the ICE agent with the `web.audio.RTCPeerConnection`. `var state = RTCPeerConnection.iceConnectionState;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceConnectionState`
(ice-gathering-state this)
Property.
[Read Only] [Experimental]
The read-only property RTCPeerConnection.iceGatheringState returns enum of type RTCIceGatheringState that describes connection's gathering state. This lets you detect, for example, when collection ICE candidates has finished.
var state = RTCPeerConnection.iceGatheringState;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceGatheringState
Property. [Read Only] [Experimental] The read-only property RTCPeerConnection.iceGatheringState returns enum of type RTCIceGatheringState that describes connection's gathering state. This lets you detect, for example, when collection ICE candidates has finished. `var state = RTCPeerConnection.iceGatheringState;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceGatheringState`
(local-description this)
Property.
[Read Only] [Experimental]
The read-only property RTCPeerConnection.localDescription returns
web.audio.RTCSessionDescription
describing the session for
local end of the connection. If it has not yet been set, this
null.
`var sessionDescription = peerConnection.localDescription;
On a more fundamental level, the returned value is the value of RTCPeerConnection.pendingLocalDescription
if that property isn't null; otherwise, the value of RTCPeerConnection.currentLocalDescription
is returned. See Pending and current descriptions in WebRTC connectivity for details on this algorithm and why it's used.`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/localDescription
Property. [Read Only] [Experimental] The read-only property RTCPeerConnection.localDescription returns `web.audio.RTCSessionDescription` describing the session for local end of the connection. If it has not yet been set, this null. `var sessionDescription = peerConnection.localDescription; On a more fundamental level, the returned value is the value of `RTCPeerConnection.pendingLocalDescription` if that property isn't null; otherwise, the value of `RTCPeerConnection.currentLocalDescription` is returned. See Pending and current descriptions in WebRTC connectivity for details on this algorithm and why it's used.` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/localDescription`
(onaddstream this)
Property.
[Deprecated]
The RTCPeerConnection.onaddstream event handler is a property
the code to execute when the addstream event, of type web.rtc.MediaStreamEvent
,
received by this web.audio.RTCPeerConnection
. Such an event
sent when a web.streams.MediaStream
is added to this connection
the remote peer. The event is sent immediately after the call
and doesn't wait for the result of the SDP negotiation.
rtcPeerConnection.onaddstream = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
Property. [Deprecated] The RTCPeerConnection.onaddstream event handler is a property the code to execute when the addstream event, of type `web.rtc.MediaStreamEvent`, received by this `web.audio.RTCPeerConnection`. Such an event sent when a `web.streams.MediaStream` is added to this connection the remote peer. The event is sent immediately after the call and doesn't wait for the result of the SDP negotiation. `rtcPeerConnection.onaddstream = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream`
(onconnectionstatechange this)
Property.
The RTCPeerConnection.onconnectionstatechange property specifies
EventHandler
which is called to handle the connectionstatechange
when it occurs on an instance of web.audio.RTCPeerConnection
.
happens whenever the aggregate state of the connection changes.
RTCPeerConnection.onconnectionstatechange = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onconnectionstatechange
Property. The RTCPeerConnection.onconnectionstatechange property specifies `EventHandler` which is called to handle the connectionstatechange when it occurs on an instance of `web.audio.RTCPeerConnection`. happens whenever the aggregate state of the connection changes. `RTCPeerConnection.onconnectionstatechange = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onconnectionstatechange`
(ondatachannel this)
Property.
[Experimental]
The RTCPeerConnection.ondatachannel property is an EventHandler
specifies a function which is called when the datachannel event
on an web.audio.RTCPeerConnection
. This event, of type web.rtc.RTCDataChannelEvent
,
sent when an web.rtc.RTCDataChannel
is added to the connection
the remote peer calling createDataChannel()
.
RTCPeerConnection.ondatachannel = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ondatachannel
Property. [Experimental] The RTCPeerConnection.ondatachannel property is an `EventHandler` specifies a function which is called when the datachannel event on an `web.audio.RTCPeerConnection`. This event, of type `web.rtc.RTCDataChannelEvent`, sent when an `web.rtc.RTCDataChannel` is added to the connection the remote peer calling `createDataChannel()`. `RTCPeerConnection.ondatachannel = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ondatachannel`
(onicecandidate this)
Property.
The RTCPeerConnection property onicecandidate
property is an
which specifies a function to be called when the icecandidate
occurs on an web.audio.RTCPeerConnection
instance. This happens
the local ICE agent needs to deliver a message to the other peer
the signaling server.
rtcPeerConnection.onicecandidate = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onicecandidate
Property. The RTCPeerConnection property `onicecandidate` property is an which specifies a function to be called when the icecandidate occurs on an `web.audio.RTCPeerConnection` instance. This happens the local ICE agent needs to deliver a message to the other peer the signaling server. `rtcPeerConnection.onicecandidate = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onicecandidate`
(oniceconnectionstatechange this)
Property.
[Experimental]
The RTCPeerConnection.oniceconnectionstatechange property is
event handler which specifies a function to be called when the
event is fired on an web.audio.RTCPeerConnection
instance.
happens when the state of the connection's ICE agent, as represented
the iceConnectionState
property, changes.
RTCPeerConnection.oniceconnectionstatechange = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/oniceconnectionstatechange
Property. [Experimental] The RTCPeerConnection.oniceconnectionstatechange property is event handler which specifies a function to be called when the event is fired on an `web.audio.RTCPeerConnection` instance. happens when the state of the connection's ICE agent, as represented the `iceConnectionState` property, changes. `RTCPeerConnection.oniceconnectionstatechange = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/oniceconnectionstatechange`
(onicegatheringstatechange this)
Property.
[Experimental]
The RTCPeerConnection.onicegatheringstatechange property is an
which specifies a function to be called when the icegatheringstatechange
is sent to an web.audio.RTCPeerConnection
instance. This happens
the ICE gathering state—that is, whether or not the ICE agent
actively gathering candidates—changes.
RTCPeerConnection.onicegatheringstatechange = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onicegatheringstatechange
Property. [Experimental] The RTCPeerConnection.onicegatheringstatechange property is an which specifies a function to be called when the icegatheringstatechange is sent to an `web.audio.RTCPeerConnection` instance. This happens the ICE gathering state—that is, whether or not the ICE agent actively gathering candidates—changes. `RTCPeerConnection.onicegatheringstatechange = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onicegatheringstatechange`
(onidentityresult this)
Property.
[Experimental]
The RTCPeerConnection.onidentityresult event handler is a property
the code to execute when the identityresult event, of type web.rtc.RTCIdentityEvent
,
received by this web.audio.RTCPeerConnection
. Such an event
sent when an identity assertion is generated, via getIdentityAssertion()
during the creation of an offer or an answer.
peerconnection.onidentityresult = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidentityresult
Property. [Experimental] The RTCPeerConnection.onidentityresult event handler is a property the code to execute when the identityresult event, of type `web.rtc.RTCIdentityEvent`, received by this `web.audio.RTCPeerConnection`. Such an event sent when an identity assertion is generated, via `getIdentityAssertion()` during the creation of an offer or an answer. `peerconnection.onidentityresult = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidentityresult`
(onidpassertionerror this)
Property.
[Obsolute]
The RTCPeerConnection.onidpassertionerror event handler is a
containing the code to execute whent the idpassertionerror event,
type web.rtc.RTCIdentityErrorEvent
, is received by this web.audio.RTCPeerConnection
.
an event is sent when the associated identity provider (IdP)
an error while generating an identity assertion.
peerconnection.onidpassertionerror = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidpassertionerror
Property. [Obsolute] The RTCPeerConnection.onidpassertionerror event handler is a containing the code to execute whent the idpassertionerror event, type `web.rtc.RTCIdentityErrorEvent`, is received by this `web.audio.RTCPeerConnection`. an event is sent when the associated identity provider (IdP) an error while generating an identity assertion. `peerconnection.onidpassertionerror = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidpassertionerror`
(onidpvalidationerror this)
Property.
[Obsolute]
The RTCPeerConnection.onidpvalidationerror event handler is a
containing the code to execute whent the idpvalidationerror event,
type web.rtc.RTCIdentityErrorEvent
, is received by this web.audio.RTCPeerConnection
.
an event is sent when the associated identity provider (IdP)
an error while validating an identity assertion.
peerconnection.onidpvalidationerror = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidpvalidationerror
Property. [Obsolute] The RTCPeerConnection.onidpvalidationerror event handler is a containing the code to execute whent the idpvalidationerror event, type `web.rtc.RTCIdentityErrorEvent`, is received by this `web.audio.RTCPeerConnection`. an event is sent when the associated identity provider (IdP) an error while validating an identity assertion. `peerconnection.onidpvalidationerror = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidpvalidationerror`
(onnegotiationneeded this)
Property.
The web.audio.RTCPeerConnection
interface's onnegotiationneeded
is an web.EventListener
which specifies a function which is
to handle the negotiationneeded event when it occurs on an web.audio.RTCPeerConnection
This event is fired when a change has occurred which requires
negotiation. This negotiation should be carried out as the offerer,
some session changes cannot be negotiated as the answerer.
RTCPeerConnection.onnegotiationneeded = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onnegotiationneeded
Property. The `web.audio.RTCPeerConnection` interface's onnegotiationneeded is an `web.EventListener` which specifies a function which is to handle the negotiationneeded event when it occurs on an `web.audio.RTCPeerConnection` This event is fired when a change has occurred which requires negotiation. This negotiation should be carried out as the offerer, some session changes cannot be negotiated as the answerer. `RTCPeerConnection.onnegotiationneeded = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onnegotiationneeded`
(onpeeridentity this)
Property.
[Obsolute]
The RTCPeerConnection.onpeeridentity event handler is a property
the code to execute whent the peeridentity event, of type web.event.Event
,
received by this web.audio.RTCPeerConnection
. Such an event
sent when an identity assertion, received from a peer, has been
validated.
peerconnection.onpeeridentity = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onpeeridentity
Property. [Obsolute] The RTCPeerConnection.onpeeridentity event handler is a property the code to execute whent the peeridentity event, of type `web.event.Event`, received by this `web.audio.RTCPeerConnection`. Such an event sent when an identity assertion, received from a peer, has been validated. `peerconnection.onpeeridentity = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onpeeridentity`
(onremovestream this)
Property.
[Deprecated] [Experimental]
The RTCPeerConnection.onremovestream event handler is a property
the code to execute when the removestream event, of type web.rtc.MediaStreamEvent
,
received by this web.audio.RTCPeerConnection
. Such an event
sent when a web.streams.MediaStream
is removed from this connection.
peerconnection.onremovestream = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onremovestream
Property. [Deprecated] [Experimental] The RTCPeerConnection.onremovestream event handler is a property the code to execute when the removestream event, of type `web.rtc.MediaStreamEvent`, received by this `web.audio.RTCPeerConnection`. Such an event sent when a `web.streams.MediaStream` is removed from this connection. `peerconnection.onremovestream = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onremovestream`
(onsignalingstatechange this)
Property.
The onsignalingstatechange property of the web.audio.RTCPeerConnection
is an EventHandler
which specifies a function to be called
the signalingstatechange event occurs on an web.audio.RTCPeerConnection
RTCPeerConnection.onsignalingstatechange = errorHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onsignalingstatechange
Property. The onsignalingstatechange property of the `web.audio.RTCPeerConnection` is an `EventHandler` which specifies a function to be called the signalingstatechange event occurs on an `web.audio.RTCPeerConnection` `RTCPeerConnection.onsignalingstatechange = errorHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onsignalingstatechange`
(ontrack this)
Property.
The web.audio.RTCPeerConnection
property ontrack is an EventHandler
specifies a function to be called when the track event occurs,
that a track has been added to the web.audio.RTCPeerConnection
.
RTCPeerConnection.ontrack = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
Property. The `web.audio.RTCPeerConnection` property ontrack is an `EventHandler` specifies a function to be called when the track event occurs, that a track has been added to the `web.audio.RTCPeerConnection`. `RTCPeerConnection.ontrack = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack`
(peer-identity this)
Property.
[Read Only]
The read-only web.audio.RTCPeerConnection
property peerIdentity
a JavaScript js.Promise
that resolves to an web.rtc.RTCIdentityAssertion
contains a web.dom.DOMString
identifying the remote peer.
var identity = rtcPeerConnection.peerIdentity;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/peerIdentity
Property. [Read Only] The read-only `web.audio.RTCPeerConnection` property peerIdentity a JavaScript `js.Promise` that resolves to an `web.rtc.RTCIdentityAssertion` contains a `web.dom.DOMString` identifying the remote peer. `var identity = rtcPeerConnection.peerIdentity;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/peerIdentity`
(pending-local-description this)
Property.
[Read Only]
The read-only property RTCPeerConnection.pendingLocalDescription
an web.audio.RTCSessionDescription
object describing a pending
change for the local end of the connection. This does not describe
connection as it currently stands, but as it may exist in the
future. Use RTCPeerConnection.currentLocalDescription
or RTCPeerConnection.localDescription
get the current state of the endpoint. For details on the difference,
Pending and current descriptions in WebRTC connectivity.
sessionDescription = RTCPeerConnection.pendingLocalDescription;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/pendingLocalDescription
Property. [Read Only] The read-only property RTCPeerConnection.pendingLocalDescription an `web.audio.RTCSessionDescription` object describing a pending change for the local end of the connection. This does not describe connection as it currently stands, but as it may exist in the future. Use `RTCPeerConnection.currentLocalDescription` or `RTCPeerConnection.localDescription` get the current state of the endpoint. For details on the difference, Pending and current descriptions in WebRTC connectivity. `sessionDescription = RTCPeerConnection.pendingLocalDescription;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/pendingLocalDescription`
(pending-remote-description this)
Property.
[Read Only]
The read-only property RTCPeerConnection.pendingRemoteDescription
an web.audio.RTCSessionDescription
object describing a pending
change for the remote end of the connection. This does not describe
connection as it currently stands, but as it may exist in the
future. Use RTCPeerConnection.currentRemoteDescription
or RTCPeerConnection.remoteDescription
get the current session description for the remote endpoint.
details on the difference, see Pending and current descriptions
WebRTC connectivity.
sessionDescription = RTCPeerConnection.pendingRemoteDescription;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/pendingRemoteDescription
Property. [Read Only] The read-only property RTCPeerConnection.pendingRemoteDescription an `web.audio.RTCSessionDescription` object describing a pending change for the remote end of the connection. This does not describe connection as it currently stands, but as it may exist in the future. Use `RTCPeerConnection.currentRemoteDescription` or `RTCPeerConnection.remoteDescription` get the current session description for the remote endpoint. details on the difference, see Pending and current descriptions WebRTC connectivity. `sessionDescription = RTCPeerConnection.pendingRemoteDescription;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/pendingRemoteDescription`
(remote-description this)
Property.
[Read Only] [Experimental]
The read-only property RTCPeerConnection.remoteDescription returns
web.audio.RTCSessionDescription
describing the session (which
configuration and media information) for the remote end of the
If this hasn't been set yet, this is null.
`var sessionDescription = peerConnection.remoteDescription;
On a more fundamental level, the returned value is the value of RTCPeerConnection.pendingRemoteDescription
if that property isn't null; otherwise, the value of RTCPeerConnection.currentRemoteDescription
is returned. See Pending and current descriptions in WebRTC connectivity for details on this algorithm and why it's used.`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/remoteDescription
Property. [Read Only] [Experimental] The read-only property RTCPeerConnection.remoteDescription returns `web.audio.RTCSessionDescription` describing the session (which configuration and media information) for the remote end of the If this hasn't been set yet, this is null. `var sessionDescription = peerConnection.remoteDescription; On a more fundamental level, the returned value is the value of `RTCPeerConnection.pendingRemoteDescription` if that property isn't null; otherwise, the value of `RTCPeerConnection.currentRemoteDescription` is returned. See Pending and current descriptions in WebRTC connectivity for details on this algorithm and why it's used.` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/remoteDescription`
(remove-stream this media-stream)
Method.
[Deprecated] [Experimental]
The RTCPeerConnection.removeStream() method removes a web.streams.MediaStream
a local source of audio or video. If the negotiation already
a new one will be needed for the remote peer to be able to use
Because this method has been deprecated, you should instead use
if your target browser versions have implemented it.
RTCPeerConnection.removeStream(mediaStream);
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/removeStream
Method. [Deprecated] [Experimental] The RTCPeerConnection.removeStream() method removes a `web.streams.MediaStream` a local source of audio or video. If the negotiation already a new one will be needed for the remote peer to be able to use Because this method has been deprecated, you should instead use if your target browser versions have implemented it. `RTCPeerConnection.removeStream(mediaStream);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/removeStream`
(remove-track this sender)
Method.
The RTCPeerConnection.removeTrack() method tells the local end
the connection to stop sending media from the specified track,
actually removing the corresponding web.audio.RTCRtpSender
the list of senders as reported by RTCPeerConnection.getSenders()
.
pc.removeTrack(sender);
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/removeTrack
Method. The RTCPeerConnection.removeTrack() method tells the local end the connection to stop sending media from the specified track, actually removing the corresponding `web.audio.RTCRtpSender` the list of senders as reported by `RTCPeerConnection.getSenders()`. `pc.removeTrack(sender);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/removeTrack`
(restart-ice this)
Method.
The WebRTC API's web.audio.RTCPeerConnection
interface offers
restartIce() method to allow a web application to easily request
ICE candidate gathering be redone on both ends of the connection.
rtcPeerConnection.restartIce();
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/restartIce
Method. The WebRTC API's `web.audio.RTCPeerConnection` interface offers restartIce() method to allow a web application to easily request ICE candidate gathering be redone on both ends of the connection. `rtcPeerConnection.restartIce();` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/restartIce`
(sctp this)
Property.
[Read Only] [Experimental]
The read-only sctp property on the web.audio.RTCPeerConnection
returns an web.rtc.RTCSctpTransport
describing the SCTP transport
which SCTP data is being sent and received. If SCTP hasn't been
this value is null.
var sctp = RTCPeerConnection.sctp;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/sctp
Property. [Read Only] [Experimental] The read-only sctp property on the `web.audio.RTCPeerConnection` returns an `web.rtc.RTCSctpTransport` describing the SCTP transport which SCTP data is being sent and received. If SCTP hasn't been this value is null. `var sctp = RTCPeerConnection.sctp;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/sctp`
(set-configuration this configuration)
Method.
[Experimental]
The RTCPeerConnection.setConfiguration() method sets the current
of the web.audio.RTCPeerConnection
based on the values included
the specified web.rtc.RTCConfiguration
object. This lets you
the ICE servers used by the connection and which transport policies
use.
RTCPeerConnection.setConfiguration(configuration);
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setConfiguration
Method. [Experimental] The RTCPeerConnection.setConfiguration() method sets the current of the `web.audio.RTCPeerConnection` based on the values included the specified `web.rtc.RTCConfiguration` object. This lets you the ICE servers used by the connection and which transport policies use. `RTCPeerConnection.setConfiguration(configuration);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setConfiguration`
(set-identity-provider this & args)
Method.
[Experimental]
The RTCPeerConnection.setIdentityProvider() method sets the Identity (IdP) to the triplet given in parameter: its name, the protocol to communicate with it (optional) and an optional username. The will be used only when an assertion is needed.
`pc.setIdentityProvider(domainname [, protocol] [, username]);
There is no return value for this method.`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setIdentityProvider
Method. [Experimental] The RTCPeerConnection.setIdentityProvider() method sets the Identity (IdP) to the triplet given in parameter: its name, the protocol to communicate with it (optional) and an optional username. The will be used only when an assertion is needed. `pc.setIdentityProvider(domainname [, protocol] [, username]); There is no return value for this method.` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setIdentityProvider`
(set-local-description this & args)
Method.
[Experimental]
The RTCPeerConnection.setLocalDescription() method changes the description associated with the connection. This description the properties of the local end of the connection, including media format.
`aPromise = RTCPeerConnection.setLocalDescription(sessionDescription);
pc.setLocalDescription(sessionDescription, successCallback, errorCallback);`
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setLocalDescription
Method. [Experimental] The RTCPeerConnection.setLocalDescription() method changes the description associated with the connection. This description the properties of the local end of the connection, including media format. `aPromise = RTCPeerConnection.setLocalDescription(sessionDescription); pc.setLocalDescription(sessionDescription, successCallback, errorCallback);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setLocalDescription`
(set-onaddstream! this val)
Property.
[Deprecated]
The RTCPeerConnection.onaddstream event handler is a property
the code to execute when the addstream event, of type web.rtc.MediaStreamEvent
,
received by this web.audio.RTCPeerConnection
. Such an event
sent when a web.streams.MediaStream
is added to this connection
the remote peer. The event is sent immediately after the call
and doesn't wait for the result of the SDP negotiation.
rtcPeerConnection.onaddstream = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
Property. [Deprecated] The RTCPeerConnection.onaddstream event handler is a property the code to execute when the addstream event, of type `web.rtc.MediaStreamEvent`, received by this `web.audio.RTCPeerConnection`. Such an event sent when a `web.streams.MediaStream` is added to this connection the remote peer. The event is sent immediately after the call and doesn't wait for the result of the SDP negotiation. `rtcPeerConnection.onaddstream = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream`
(set-onconnectionstatechange! this val)
Property.
The RTCPeerConnection.onconnectionstatechange property specifies
EventHandler
which is called to handle the connectionstatechange
when it occurs on an instance of web.audio.RTCPeerConnection
.
happens whenever the aggregate state of the connection changes.
RTCPeerConnection.onconnectionstatechange = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onconnectionstatechange
Property. The RTCPeerConnection.onconnectionstatechange property specifies `EventHandler` which is called to handle the connectionstatechange when it occurs on an instance of `web.audio.RTCPeerConnection`. happens whenever the aggregate state of the connection changes. `RTCPeerConnection.onconnectionstatechange = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onconnectionstatechange`
(set-ondatachannel! this val)
Property.
[Experimental]
The RTCPeerConnection.ondatachannel property is an EventHandler
specifies a function which is called when the datachannel event
on an web.audio.RTCPeerConnection
. This event, of type web.rtc.RTCDataChannelEvent
,
sent when an web.rtc.RTCDataChannel
is added to the connection
the remote peer calling createDataChannel()
.
RTCPeerConnection.ondatachannel = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ondatachannel
Property. [Experimental] The RTCPeerConnection.ondatachannel property is an `EventHandler` specifies a function which is called when the datachannel event on an `web.audio.RTCPeerConnection`. This event, of type `web.rtc.RTCDataChannelEvent`, sent when an `web.rtc.RTCDataChannel` is added to the connection the remote peer calling `createDataChannel()`. `RTCPeerConnection.ondatachannel = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ondatachannel`
(set-onicecandidate! this val)
Property.
The RTCPeerConnection property onicecandidate
property is an
which specifies a function to be called when the icecandidate
occurs on an web.audio.RTCPeerConnection
instance. This happens
the local ICE agent needs to deliver a message to the other peer
the signaling server.
rtcPeerConnection.onicecandidate = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onicecandidate
Property. The RTCPeerConnection property `onicecandidate` property is an which specifies a function to be called when the icecandidate occurs on an `web.audio.RTCPeerConnection` instance. This happens the local ICE agent needs to deliver a message to the other peer the signaling server. `rtcPeerConnection.onicecandidate = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onicecandidate`
(set-oniceconnectionstatechange! this val)
Property.
[Experimental]
The RTCPeerConnection.oniceconnectionstatechange property is
event handler which specifies a function to be called when the
event is fired on an web.audio.RTCPeerConnection
instance.
happens when the state of the connection's ICE agent, as represented
the iceConnectionState
property, changes.
RTCPeerConnection.oniceconnectionstatechange = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/oniceconnectionstatechange
Property. [Experimental] The RTCPeerConnection.oniceconnectionstatechange property is event handler which specifies a function to be called when the event is fired on an `web.audio.RTCPeerConnection` instance. happens when the state of the connection's ICE agent, as represented the `iceConnectionState` property, changes. `RTCPeerConnection.oniceconnectionstatechange = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/oniceconnectionstatechange`
(set-onicegatheringstatechange! this val)
Property.
[Experimental]
The RTCPeerConnection.onicegatheringstatechange property is an
which specifies a function to be called when the icegatheringstatechange
is sent to an web.audio.RTCPeerConnection
instance. This happens
the ICE gathering state—that is, whether or not the ICE agent
actively gathering candidates—changes.
RTCPeerConnection.onicegatheringstatechange = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onicegatheringstatechange
Property. [Experimental] The RTCPeerConnection.onicegatheringstatechange property is an which specifies a function to be called when the icegatheringstatechange is sent to an `web.audio.RTCPeerConnection` instance. This happens the ICE gathering state—that is, whether or not the ICE agent actively gathering candidates—changes. `RTCPeerConnection.onicegatheringstatechange = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onicegatheringstatechange`
(set-onidentityresult! this val)
Property.
[Experimental]
The RTCPeerConnection.onidentityresult event handler is a property
the code to execute when the identityresult event, of type web.rtc.RTCIdentityEvent
,
received by this web.audio.RTCPeerConnection
. Such an event
sent when an identity assertion is generated, via getIdentityAssertion()
during the creation of an offer or an answer.
peerconnection.onidentityresult = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidentityresult
Property. [Experimental] The RTCPeerConnection.onidentityresult event handler is a property the code to execute when the identityresult event, of type `web.rtc.RTCIdentityEvent`, received by this `web.audio.RTCPeerConnection`. Such an event sent when an identity assertion is generated, via `getIdentityAssertion()` during the creation of an offer or an answer. `peerconnection.onidentityresult = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidentityresult`
(set-onidpassertionerror! this val)
Property.
[Obsolute]
The RTCPeerConnection.onidpassertionerror event handler is a
containing the code to execute whent the idpassertionerror event,
type web.rtc.RTCIdentityErrorEvent
, is received by this web.audio.RTCPeerConnection
.
an event is sent when the associated identity provider (IdP)
an error while generating an identity assertion.
peerconnection.onidpassertionerror = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidpassertionerror
Property. [Obsolute] The RTCPeerConnection.onidpassertionerror event handler is a containing the code to execute whent the idpassertionerror event, type `web.rtc.RTCIdentityErrorEvent`, is received by this `web.audio.RTCPeerConnection`. an event is sent when the associated identity provider (IdP) an error while generating an identity assertion. `peerconnection.onidpassertionerror = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidpassertionerror`
(set-onidpvalidationerror! this val)
Property.
[Obsolute]
The RTCPeerConnection.onidpvalidationerror event handler is a
containing the code to execute whent the idpvalidationerror event,
type web.rtc.RTCIdentityErrorEvent
, is received by this web.audio.RTCPeerConnection
.
an event is sent when the associated identity provider (IdP)
an error while validating an identity assertion.
peerconnection.onidpvalidationerror = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidpvalidationerror
Property. [Obsolute] The RTCPeerConnection.onidpvalidationerror event handler is a containing the code to execute whent the idpvalidationerror event, type `web.rtc.RTCIdentityErrorEvent`, is received by this `web.audio.RTCPeerConnection`. an event is sent when the associated identity provider (IdP) an error while validating an identity assertion. `peerconnection.onidpvalidationerror = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onidpvalidationerror`
(set-onnegotiationneeded! this val)
Property.
The web.audio.RTCPeerConnection
interface's onnegotiationneeded
is an web.EventListener
which specifies a function which is
to handle the negotiationneeded event when it occurs on an web.audio.RTCPeerConnection
This event is fired when a change has occurred which requires
negotiation. This negotiation should be carried out as the offerer,
some session changes cannot be negotiated as the answerer.
RTCPeerConnection.onnegotiationneeded = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onnegotiationneeded
Property. The `web.audio.RTCPeerConnection` interface's onnegotiationneeded is an `web.EventListener` which specifies a function which is to handle the negotiationneeded event when it occurs on an `web.audio.RTCPeerConnection` This event is fired when a change has occurred which requires negotiation. This negotiation should be carried out as the offerer, some session changes cannot be negotiated as the answerer. `RTCPeerConnection.onnegotiationneeded = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onnegotiationneeded`
(set-onpeeridentity! this val)
Property.
[Obsolute]
The RTCPeerConnection.onpeeridentity event handler is a property
the code to execute whent the peeridentity event, of type web.event.Event
,
received by this web.audio.RTCPeerConnection
. Such an event
sent when an identity assertion, received from a peer, has been
validated.
peerconnection.onpeeridentity = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onpeeridentity
Property. [Obsolute] The RTCPeerConnection.onpeeridentity event handler is a property the code to execute whent the peeridentity event, of type `web.event.Event`, received by this `web.audio.RTCPeerConnection`. Such an event sent when an identity assertion, received from a peer, has been validated. `peerconnection.onpeeridentity = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onpeeridentity`
(set-onremovestream! this val)
Property.
[Deprecated] [Experimental]
The RTCPeerConnection.onremovestream event handler is a property
the code to execute when the removestream event, of type web.rtc.MediaStreamEvent
,
received by this web.audio.RTCPeerConnection
. Such an event
sent when a web.streams.MediaStream
is removed from this connection.
peerconnection.onremovestream = function;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onremovestream
Property. [Deprecated] [Experimental] The RTCPeerConnection.onremovestream event handler is a property the code to execute when the removestream event, of type `web.rtc.MediaStreamEvent`, received by this `web.audio.RTCPeerConnection`. Such an event sent when a `web.streams.MediaStream` is removed from this connection. `peerconnection.onremovestream = function;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onremovestream`
(set-onsignalingstatechange! this val)
Property.
The onsignalingstatechange property of the web.audio.RTCPeerConnection
is an EventHandler
which specifies a function to be called
the signalingstatechange event occurs on an web.audio.RTCPeerConnection
RTCPeerConnection.onsignalingstatechange = errorHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onsignalingstatechange
Property. The onsignalingstatechange property of the `web.audio.RTCPeerConnection` is an `EventHandler` which specifies a function to be called the signalingstatechange event occurs on an `web.audio.RTCPeerConnection` `RTCPeerConnection.onsignalingstatechange = errorHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onsignalingstatechange`
(set-ontrack! this val)
Property.
The web.audio.RTCPeerConnection
property ontrack is an EventHandler
specifies a function to be called when the track event occurs,
that a track has been added to the web.audio.RTCPeerConnection
.
RTCPeerConnection.ontrack = eventHandler;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
Property. The `web.audio.RTCPeerConnection` property ontrack is an `EventHandler` specifies a function to be called when the track event occurs, that a track has been added to the `web.audio.RTCPeerConnection`. `RTCPeerConnection.ontrack = eventHandler;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack`
(set-remote-description this session-description)
Method.
The RTCPeerConnection.setRemoteDescription() method changes the description associated with the connection. This description the properties of the remote end of the connection, including media format.
aPromise = pc.setRemoteDescription(sessionDescription);
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setRemoteDescription
Method. The RTCPeerConnection.setRemoteDescription() method changes the description associated with the connection. This description the properties of the remote end of the connection, including media format. `aPromise = pc.setRemoteDescription(sessionDescription);` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setRemoteDescription`
(signaling-state this)
Property.
[Read Only]
The read-only signalingState property on the web.audio.RTCPeerConnection
returns one of the string values specified by the RTCSignalingState
these values describe the state of the signaling process on the
end of the connection while connecting or reconnecting to another
See Signaling in Lifetime of a WebRTC session for more details
the signaling process.
var state = RTCPeerConnection.signalingState;
See also: https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/signalingState
Property. [Read Only] The read-only signalingState property on the `web.audio.RTCPeerConnection` returns one of the string values specified by the RTCSignalingState these values describe the state of the signaling process on the end of the connection while connecting or reconnecting to another See Signaling in Lifetime of a WebRTC session for more details the signaling process. `var state = RTCPeerConnection.signalingState;` See also: `https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/signalingState`
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