RTCPeerConnection - Web APIs 编辑

Event handlers

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

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Constructor

RTCPeerConnection()

The RTCPeerConnection() constructor returns a newly-created RTCPeerConnection, which represents a connection between the local device and a remote peer.

Properties

Also inherits properties from: EventTarget

canTrickleIceCandidates

The read-only RTCPeerConnection property canTrickleIceCandidates returns a Boolean which indicates whether or not the remote peer can accept trickled ICE candidates.

connectionState

The read-only connectionState property of the RTCPeerConnection interface indicates the current state of the peer connection by returning one of the string values specified by the enum RTCPeerConnectionState.

currentLocalDescription undefined

The read-only property RTCPeerConnection.currentLocalDescription returns an RTCSessionDescription object describing the local end of the connection as it was most recently successfully negotiated since the last time the  RTCPeerConnection finished negotiating and connecting to a remote peer. Also included is a list of any ICE candidates that may already have been generated by the ICE agent since the offer or answer represented by the description was first instantiated.

currentRemoteDescription undefined

The read-only property RTCPeerConnection.currentRemoteDescription returns an RTCSessionDescription object describing the remote end of the connection as it was most recently successfully negotiated since the last time the RTCPeerConnection finished negotiating and connecting to a remote peer. Also included is a list of any ICE candidates that may already have been generated by the ICE agent since the offer or answer represented by the description was first instantiated.

getDefaultIceServers()

The getDefaultIceServers() method of the RTCPeerConnection interface returns an array of objects based on the RTCIceServer dictionary, which indicates what, if any, ICE servers the browser will use by default if none are provided to the RTCPeerConnection in its RTCConfiguration. However, browsers are not required to provide any default ICE servers at all.

iceConnectionState undefined

The read-only property RTCPeerConnection.iceConnectionState returns an enum of type RTCIceConnectionState which state of the ICE agent associated with the RTCPeerConnection.

iceGatheringState undefined

The read-only property RTCPeerConnection.iceGatheringState returns an enum of type RTCIceGatheringState that describes connection's ICE gathering state. This lets you detect, for example, when collection of ICE candidates has finished.

localDescription undefined

The read-only property RTCPeerConnection.localDescription returns an RTCSessionDescription describing the session for the local end of the connection. If it has not yet been set, this is null.

peerIdentity undefined

The read-only RTCPeerConnection property peerIdentity returns a JavaScript Promise that resolves to an RTCIdentityAssertion which contains a DOMString identifying the remote peer.

pendingLocalDescription undefined

The read-only property RTCPeerConnection.pendingLocalDescription returns an RTCSessionDescription object describing a pending configuration change for the local end of the connection. This does not describe the connection as it currently stands, but as it may exist in the near future. Use RTCPeerConnection.currentLocalDescription or RTCPeerConnection.localDescription to get the current state of the endpoint. For details on the difference, see /wiki/en-US/docs/Web/API/WebRTC_API/Connectivity.

pendingRemoteDescription undefined

The read-only property RTCPeerConnection.pendingRemoteDescription returns an RTCSessionDescription object describing a pending configuration change for the remote end of the connection. This does not describe the connection as it currently stands, but as it may exist in the near future. Use RTCPeerConnection.currentRemoteDescription or RTCPeerConnection.remoteDescription to get the current session description for the remote endpoint. For details on the difference, see /wiki/en-US/docs/Web/API/WebRTC_API/Connectivity.

remoteDescription undefined

The read-only property RTCPeerConnection.remoteDescription returns a RTCSessionDescription describing the session (which includes configuration and media information) for the remote end of the connection. If this hasn't been set yet, this is null.

sctp

The read-only sctp property on the RTCPeerConnection interface returns an RTCSctpTransport describing the SCTP transport over which SCTP data is being sent and received. If SCTP hasn't been negotiated, this value is null.

signalingState undefined

The read-only signalingState property on the RTCPeerConnection interface returns one of the string values specified by the RTCSignalingState enum; these values describe the state of the signaling process on the local end of the connection while connecting or reconnecting to another peer. See /wiki/en-US/docs/Web/API/WebRTC_API/Session_lifetime for more details about the signaling process.

Also inherits event handlers from: EventTarget

onaddstream

The RTCPeerConnection.onaddstream event handler is a property containing the code to execute when the addstream event, of type MediaStreamEvent, is received by this RTCPeerConnection. Such an event is sent when a MediaStream is added to this connection by the remote peer. The event is sent immediately after the call RTCPeerConnection.setRemoteDescription and doesn't wait for the result of the SDP negotiation.

onconnectionstatechange

The RTCPeerConnection.onconnectionstatechange property specifies an EventHandler which is called to handle the connectionstatechange event when it occurs on an instance of RTCPeerConnection. This happens whenever the aggregate state of the connection changes.

ondatachannel

The RTCPeerConnection.ondatachannel property is an EventHandler which specifies a function which is called when the datachannel event occurs on an RTCPeerConnection. This event, of type RTCDataChannelEvent, is sent when an RTCDataChannel is added to the connection by the remote peer calling RTCPeerConnection.createDataChannel.

onicecandidate

The RTCPeerConnection property RTCPeerConnection.onicecandidate property is an EventHandler which specifies a function to be called when the icecandidate event occurs on an RTCPeerConnection instance. This happens whenever the local ICE agent needs to deliver a message to the other peer through the signaling server.

oniceconnectionstatechange

The RTCPeerConnection.oniceconnectionstatechange property is an event handler which specifies a function to be called when the iceconnectionstatechange event is fired on an RTCPeerConnection instance. This happens when the state of the connection's ICE agent, as represented by the RTCPeerConnection.iceConnectionState property, changes.

onicegatheringstatechange

The RTCPeerConnection.onicegatheringstatechange property is an EventHandler which specifies a function to be called when the icegatheringstatechange event is sent to an RTCPeerConnection instance. This happens when the ICE gathering state—that is, whether or not the ICE agent is actively gathering candidates—changes.

onidentityresult

The RTCPeerConnection.onidentityresult event handler is a property containing the code to execute when the identityresult event, of type RTCIdentityEvent, is received by this RTCPeerConnection. Such an event is sent when an identity assertion is generated, via RTCPeerConnection.getIdentityAssertion() or during the creation of an offer or an answer.

onidpassertionerror

The RTCPeerConnection.onidpassertionerror event handler is a property containing the code to execute whent the idpassertionerror event, of type RTCIdentityErrorEvent, is received by this RTCPeerConnection. Such an event is sent when the associated identity provider (IdP) encounters an error while generating an identity assertion.

onidpvalidationerror

The RTCPeerConnection.onidpvalidationerror event handler is a property containing the code to execute whent the idpvalidationerror event, of type RTCIdentityErrorEvent, is received by this RTCPeerConnection. Such an event is sent when the associated identity provider (IdP) encounters an error while validating an identity assertion.

onnegotiationneeded

The RTCPeerConnection interface's onnegotiationneeded property is an EventListener which specifies a function which is called to handle the negotiationneeded event when it occurs on an RTCPeerConnection instance. This event is fired when a change has occurred which requires session negotiation. This negotiation should be carried out as the offerer, because some session changes cannot be negotiated as the answerer.

onpeeridentity

The RTCPeerConnection.onpeeridentity event handler is a property containing the code to execute whent the peeridentity event, of type Event, is received by this RTCPeerConnection. Such an event is sent when an identity assertion, received from a peer, has been successfully validated.

onremovestream

The removestream event has been removed from the WebRTC specification in favor of the existing removetrack event on the remote MediaStream and the corresponding MediaStream.onremovetrack event handler property of the remote MediaStream. The RTCPeerConnection API is now track-based, so having zero tracks in the remote stream is equivalent to the remote stream being removed and the old removestream event.

onsignalingstatechange

The onsignalingstatechange event handler property of the RTCPeerConnection interface specifies a function to be called when the signalingstatechange event occurs on an RTCPeerConnection interface.

ontrack

The RTCPeerConnection property ontrack is an EventHandler which specifies a function to be called when the track event occurs, indicating that a track has been added to the RTCPeerConnection.

Methods

Also inherits methods from: EventTarget

addIceCandidate()

When a web site or app using RTCPeerConnection receives a new ICE candidate from the remote peer over its signaling channel, it delivers the newly-received candidate to the browser's ICE agent by calling RTCPeerConnection.addIceCandidate().

addStream()

The obsolete RTCPeerConnection method addStream() adds a MediaStream as a local source of audio or video. Instead of using this obsolete method, you should instead use RTCPeerConnection.addTrack once for each track you wish to send to the remote peer.

addTrack()

The RTCPeerConnection method addTrack() adds a new media track to the set of tracks which will be transmitted to the other peer.

close()

The RTCPeerConnection.close() method closes the current peer connection.

createAnswer()

The createAnswer() method on the RTCPeerConnection interface creates an SDP answer to an offer received from a remote peer during the offer/answer negotiation of a WebRTC connection. The answer contains information about any media already attached to the session, codecs and options supported by the browser, and any ICE candidates already gathered. The answer is delivered to the returned Promise, and should then be sent to the source of the offer to continue the negotiation process.

createDataChannel()

The createDataChannel() method on the RTCPeerConnection interface creates a new channel linked with the remote peer, over which any kind of data may be transmitted.

createOffer()

The createOffer() method of the RTCPeerConnection interface initiates the creation of an SDP offer for the purpose of starting a new WebRTC connection to a remote peer.

generateCertificate() static function

The static  RTCPeerConnection.generateCertificate() function creates an X.509 certificate and corresponding private key, returning a promise that resolves with the new RTCCertificate once it's generated.

getConfiguration()

The RTCPeerConnection.getConfiguration() method returns an RTCConfiguration object which indicates the current configuration of the RTCPeerConnection on which the method is called.

getIdentityAssertion()

The RTCPeerConnection.getIdentityAssertion() method initiates the gathering of an identity assertion. This has an effect only if the RTCPeerConnection.signalingState is not "closed".

getReceivers()

The RTCPeerConnection.getReceivers() method returns an array of RTCRtpReceiver objects, each of which represents one RTP receiver. Each RTP receiver manages the reception and decoding of data for a MediaStreamTrack on an RTCPeerConnection

getSenders()

The RTCPeerConnection method getSenders() returns an array of RTCRtpSender objects, each of which represents the RTP sender responsible for transmitting one track's data.

getStats()

The RTCPeerConnection method getStats() returns a promise which resolves with data providing statistics about either the overall connection or about the specified MediaStreamTrack.

getStreamById()

The RTCPeerConnection.getStreamById() method returns the MediaStream with the given id that is associated with local or remote end of the connection. If no stream matches, it returns null.

getTransceivers()

The RTCPeerConnection interface's getTransceivers() method returns a list of the RTCRtpTransceiver objects being used to send and receive data on the connection.

removeStream()

The RTCPeerConnection.removeStream() method removes a MediaStream as a local source of audio or video. If the negotiation already happened, a new one will be needed for the remote peer to be able to use it. Because this method has been deprecated, you should instead use RTCPeerConnection.removeTrack if your target browser versions have implemented it.

removeTrack()

The RTCPeerConnection.removeTrack() method tells the local end of the connection to stop sending media from the specified track, without actually removing the corresponding RTCRtpSender from the list of senders as reported by RTCPeerConnection.getSenders().

restartIce()

The WebRTC API's RTCPeerConnection interface offers the restartIce() method to allow a web application to easily request that ICE candidate gathering be redone on both ends of the connection.

setConfiguration()

The RTCPeerConnection.setConfiguration() method sets the current configuration of the RTCPeerConnection based on the values included in the specified RTCConfiguration object. This lets you change the ICE servers used by the connection and which transport policies to use.

setIdentityProvider()

The RTCPeerConnection.setIdentityProvider() method sets the Identity Provider (IdP) to the triplet given in parameter: its name, the protocol used to communicate with it (optional) and an optional username. The IdP will be used only when an assertion is needed.

setLocalDescription()

The RTCPeerConnection method RTCPeerConnection.setLocalDescription changes the local description associated with the connection. This description specifies the properties of the local end of the connection, including the media format.

setRemoteDescription()

The RTCPeerConnection method setRemoteDescription() sets the specified session description as the remote peer's current offer or answer. The description specifies the properties of the remote end of the connection, including the media format.

Obsolete method

The following method was obsoleted long ago and was never implemented in all major browsers.

RTCPeerConnection.createDTMFSender()

Creates a new RTCDTMFSender, associated to a specific MediaStreamTrack, that will be able to send DTMF phone signaling over the connection.

Events

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

connectionstatechange

Sent to the RTCPeerConnection object when the overall connectivity status of the RTCPeerConnection changes.
Also available through the onconnectionstatechange event handler property.

datachannel

Sent to the RTCPeerConnection object when the remote peer adds an RTCDataChannel to the connection.
Also available through the ondatachannel event handler property.

icecandidate

Sent to the peer connection to request that the specified candidate be transmitted to the remote peer.
Also available through the onicecandidate event handler property.

icecandidateerror

An error of type RTCPeerConnectionIceErrorEvent which is sent to the connection if an error occurred during ICE candidate gathering. The event's properties describe the error.
Also available through the onicecandidateerror event handler property.

iceconnectionstatechange

Sent to the RTCPeerConnection when the state of the ICE connection changes, such as when it disconnects.
Also available using the oniceconnectionstatechange event handler property.

icegatheringstatechange

Sent to the RTCPeerConnection when the ICE layer's gathering state, reflected by iceGatheringState, changes. This indicates whether ICE negotiation has not yet begun (new), has begun gathering candidates (gathering), or has completed (complete).
Also available using the onicegatheringstatechange event handler property.

isolationchange

Sent to the RTCPeerConnection when the isolated property on one of the MediaStreamTrack objects associated with the connection changes value. A track is isolated if its content cannot be accessed by the owning document due to lack of authentication or if the track comes from a cross-origin source.
Also available through the onisolationchange event handler property.

negotiationneeded

Sent to the RTCPeerConnection when negotiation or renegotiation of the ICE connection needs to be performed; this can happen both when first opening a connection as well as when it's neccessary to adapt to changing network conditions. The receiver should respond by creating an offer and sending it to the other peer.
Also available as the onnegotiationneeded event handler property.

signalingstatechange

The signalingstatechange event is sent to the RTCPeerConnection when the connection's ICE signaling state changes.
Also available through the onsignalingstatechange event handler property.

track

The track event is sent after a new track has been added to one of the RTCRtpReceiver instances which comprise the connection.
Available as the ontrack event handler property.

Obsolete events

addstream

Sent when a new MediaStream has been added to the connection. Instead of watching for this obsolete event, you should watch each for track events; one is sent for each MediaStreamTrack added to the connection.
Available as the onaddstream event handler property.

identityresult

In old versions of the WebRTC specification, this event was used to indicate that an identity assertion is available. Now, you should instead wait for a the promise returned by peerIdentity to resolve with an identity.
Also available using the onidentityresult event handler property.

idpassertionerror

In old versions of the WebRTC specification, this event was used to indicate that an error occurred while attempting to generate an identity assertion. Now, you should instead wait for a the promise returned by peerIdentity to be rejected with an error.
Also available as the onidpinsertionerror event handler property.

idpvalidationerror

In old versions of the WebRTC specification, this event was used to indicate that an error occurred while attempting to validate an identity assertion. Now, you should instead wait for a the promise returned by peerIdentity to be rejected with an error.
Also available using the onpeerdentity event handler property.

peeridentity

In old versions of the WebRTC specification, this event was used to deliver a received identity. Now, you should instead wait for a the promise returned by peerIdentity to resolve with an identity.

removestream

Sent to the RTCPeerConnection when a MediaStream is removed from the connection. Instead of watching for this obsolete event, you should watch each stream for removetrack events on each stream within the RTCPeerConnection.
Also available as the onaddstream event handler property.

Constants

RTCBundlePolicy enum

The RTCBundlePolicy enum defines string constants which are used to request a specific policy for gathering ICE candidates if the remote peer isn't "BUNDLE-aware" (compatible with the SDP BUNDLE standard for bundling multiple media streams on a single transport link). All browser implementations are BUNDLE-aware.

If the remote endpoint is BUNDLE-aware, all media tracks and data channels are bundled onto a single transport at the completion of negotiation, regardless of policy used, and any superfluous transports that were created initially are closed at that point.

Note: In technical terms, a BUNDLE lets all media flow between two peers flow across a single 5-tuple; that is, from a single IP and port on one peer to a single IP and port on the other peer, using the same transport protocol.

RTCIceConnectionState enum

The RTCIceConnectionState enum defines the string constants used to describe the current state of the ICE agent and its connection to the ICE server (that is, the STUN or TURN server).

RTCIceGatheringState enum

The RTCIceGatheringState enum defines string constants which reflect the current status of ICE gathering, as returned using the RTCPeerConnection.iceGatheringState property. You can detect when this value changes by watching for an event of type icegatheringstatechange.

RTCIceTransportPolicy enum

The RTCIceTransportPolicy enum defines string constants which can be used to limit the transport policies of the ICE candidates to be considered during the connection process.

RTCPeerConnectionState enum

The RTCPeerConnectionState enum defines string constants which describe states in which the RTCPeerConnection may be. These values are returned by the connectionState property. This state essentially represents the aggregate state of all ICE transports (which are of type RTCIceTransport or RTCDtlsTransport) being used by the connection.

RTCRtcpMuxPolicy enum

The RTCRtcpMuxPolicy enum defines string constants which specify what ICE candidates are gathered to support non-multiplexed RTCP. <<<add a link to info about multiplexed RTCP.

RTCSignalingState enum

The RTCSignalingState enum specifies the possible values of RTCPeerConnection.signalingState, which indicates where in the process of signaling the exchange of offer and answer the connection currently is.

Specifications

Browser compatibility

BCD tables only load in the browser

See also

• https://github.com/jesup/nightly-gupshup/blob/master/static/js/chat.js
• http://www.html5rocks.com/en/tutorials/webrtc/basics/
• TutorRoom: Node.js HTML5 video capture, peer-to-peer video and filesharing application (source on GitHub)