RTCPeerConnection

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.

var RTCPeerConnection = window.RTCPeerConnection || window.mozRTCPeerConnection || 
                        window.webkitRTCPeerConnection;
var RTCSessionDescription = window.RTCSessionDescription || window.mozRTCSessionDescription;

navigator.getUserMedia = navigator.getUserMedia || navigator.mozGetUserMedia ||
                         navigator.webkitGetUserMedia;

RTCPeerConnection can receive events, so it implements the EventTarget interface.

Constructor

RTCPeerConnection.RTCPeerConnection()

Constructor; returns a new RTCPeerConnection object.

Properties

This interface inherits properties from its parent interface, EventTarget.

RTCPeerConnection.canTrickleIceCandidates Read only

true if the remote peer is able to accept trickled ICE candidates, otherwise false. This determination is made by inspecting the remote description as set when the local RTCPeerConnection.setRemoteDescription() function has been called. If that has not yet been done, this value is null (not false, but null).

RTCPeerConnection.connectionState Read only

blah

RTCPeerConnection.currentLocalDescription Read only

blah

RTCPeerConnection.currentRemoteDescription Read only

blah

RTCPeerConnection.defaultIceServers Read only

blah

RTCPeerConnection.iceConnectionState Read only

Returns an enum of type RTCIceConnectionState that describes the ICE connection state for the connection. When this value changes, a iceconnectionstatechange event is fired on the object.

RTCPeerConnection.iceGatheringState Read only

Returns an enum of type RTCIceGatheringState that describes the ICE gathering state for the connection.

RTCPeerConnection.idpLoginUrl Read only

blah

RTCPeerConnection.localDescription Read only

Returns a RTCSessionDescription describing the session for the local end of the connection. If it has not yet been set, it can be null.

RTCPeerConnection.peerIdentity Read only

Returns a RTCIdentityAssertion, that is a couple of a domain name (idp) and a name (name) representing the identity of the remote peer of this connection, once set and verified. If no peer has yet been set and verified, this property will return null. Once set, via the appropriate method, it can't be changed.

RTCPeerConnection.pendingLocalDescription Read only

blah

RTCPeerConnection.pendingRemoteDescription Read only

blah

RTCPeerConnection.remoteDescription Read only

blah

RTCPeerConnection.sctp Read only

blah

RTCPeerConnection.signalingState Read only

Returns an enum of type RTCSignalingState that describes the signaling state of the local connection. This state describes the SDP offer that defines the configuration of the connection; this includes information including the description of the locally associated objects of type MediaStream, the codec/RTP/RTCP options, and the candidates gathered by the ICE agent. When this value changes, a signalingstatechange event is fired on the object.

Event handlers

RTCPeerConnection.onconnectionstatechange

blah

RTCPeerConnection.ondatachannel

Is the event handler called when the datachannel event is received. Such an event is sent when a RTCDataChannel is added to this connection.

RTCPeerConnection.onnegotiationneeded

Is the event handler called when the negotiationneeded event, sent by the browser to inform that negotiation will be required at some point in the future, is received.

RTCPeerConnection.onicecandidate

Is the event handler called when the icecandidate event is received. Such an event is sent when a RTCICECandidate object is added to the script.

RTCPeerConnection.onicecandidateerror

blah

RTCPeerConnection.oniceconnectionstatechange

Is the event handler called when the iceconnectionstatechange event is received. Such an event is sent when the value of iceConnectionState changes.

RTCPeerConnection.onicegatheringstatechange

blah

RTCPeerConnection.onsignalingstatechange

Is the event handler called when the signalingstatechange event, sent when the value of signalingState changes, is received.

RTCPeerConnection.ontrack

blah

old:

RTCPeerConnection.onaddstream

Is the event handler called when the addstream event is received. 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.

RTCPeerConnection.onidentityresult

Is the event handler called when the identityresult event is received. Such an event is sent when an identity assertion is generated, via getIdentityAssertion(), or during the creation of an offer or an answer.

RTCPeerConnection.onidpassertionerror

Is the event handler called when the idpassertionerror event is received. Such an event is sent when the associated identity provider (IdP) encounters an error while generating an identity assertion.

RTCPeerConnection.onidpvalidationerror

The event handler called when the idpvalidationerror event is received. Such an event is sent when the associated identity provider (IdP) encounters an error while validating an identity assertion.

RTCPeerConnection.onpeeridentity

Is the event handler called when the peeridentity event, sent when a peer identity has been set and verified on this connection, is received.

RTCPeerConnection.onremovestream

Is the event handler called when the removestream event, sent when a MediaStream is removed from this connection, is received.

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 compatible with the SDP BUNDLE standard for bundling multiple media streams on a single transport link.

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.

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.

Methods

RTCPeerConnection.addIceCandidate()

When a new ICE candidate arrives over the signaling channel, call this method to deliver it to the local WebRTC ICE agent.

RTCPeerConnection.addTrack()

blah

RTCPeerConnection.addTransceiver()

blah

RTCPeerConnection.close()

Abruptly closes a connection.

RTCPeerConnection.createAnswer()

Creates an answer to the offer received by the remote peer, in a two-part offer/answer negotiation of a connection. The two first parameters are respectively success and error callbacks, the optional third one represent options for the answer to be created.

RTCPeerConnection.createDataChannel()

Creates a new RTCDataChannel associated with this connection. The method takes a dictionary as parameter, with the configuration required for the underlying data channel, like its reliability.

RTCPeerConnection.createOffer()

Creates a request to find a remote peer with a specific configuration.

RTCPeerConnection.generateCertificate()

Creates and stores an X.509 certificate and corresponding private key then returns an RTCCertificate, providing access to it.

RTCPeerConnection.getConfiguration()

blah

RTCPeerConnection.peerIdentity

blah

RTCPeerConnection.getReceivers()

blah

RTCPeerConnection.getSenders()

blah

RTCPeerConnection.getStats()

Creates a new RTCStatsReport that contains and allows access to statistics regarding the connection.

RTCPeerConnection.getTransceivers()

blah

RTCPeerConnection.removeTrack()

blah

RTCPeerConnection.setConfiguration()

blah

RTCPeerConnection.setIdentityProvider()

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 will be needed.

RTCPeerConnection.setLocalDescription()

Changes the local description associated with the connection. The description defines the properties of the connection like its codec. The connection is affected by this change and must be able to support both old and new descriptions. The method takes three parameters, a RTCSessionDescription object to set, and two callbacks, one called if the change of description succeeds, another called if it failed.

RTCPeerConnection.setRemoteDescription()

Changes the remote description associated with the connection. The description defines the properties of the connection like its codec. The connection is affected by this change and must be able to support both old and new descriptions. The method takes three parameters, a RTCSessionDescription object to set, and two callbacks, one called if the change of description succeeds, another called if it failed.

Obsolete methods

These methods were included in older versions of the WebRTC specification, but have since been removed.

RTCPeerConnection.addStream()

Adds 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.

RTCPeerConnection.createDTMFSender()

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

RTCPeerConnection.getIdentityAssertion()

Initiates the gathering of an identity assertion. This has an effect only if the signalingState is not "closed". It is not expected for the application dealing with the RTCPeerConnection: this is automatically done; an explicit call only allows to anticipate the need.

RTCPeerConnection.getLocalStreams()

Returns an array of MediaStream associated with the local end of the connection. The array may be empty.

RTCPeerConnection.getRemoteStreams()

Returns an array of MediaStream associated with the remote end of the connection. The array may be empty.

RTCPeerConnection.getStreamById()

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.

RTCPeerConnection.removeStream()

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 stop using it.

RTCPeerConnection.setIdentityProvider()

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 will be needed.

Constructor

new RTCPeerConnection(
      optional RTCConfiguration configuration
);

Creates and returns a new RTCPeerConnection object.

Specifications

Browser compatibility

See also

• https://github.com/jesup/nightly-gupshup/blob/master/static/js/chat.js
• http://www.html5rocks.com/en/tutorials/webrtc/basics/#toc-simple