RTCPeerConnection

Baseline Widely available *

This feature is well established and works across many devices and browser versions. It’s been available across browsers since ⁨September 2017⁩.

* Some parts of this feature may have varying levels of support.

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.

Constructor

RTCPeerConnection(): Returns a new RTCPeerConnection, representing a connection between the local device and a remote peer.

Instance properties

Also inherits properties from EventTarget.

canTrickleIceCandidates Read only: Returns a boolean value which indicates whether or not the remote peer can accept trickled ICE candidates.
connectionState Read only: Indicates the current state of the peer connection by returning one of the strings: new, connecting, connected, disconnected, failed, or closed.
currentLocalDescription Read only: Returns an RTCSessionDescription object describing the local end of the connection as it was most recently successfully negotiated since the last time this 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 Read only: Returns an RTCSessionDescription object describing the remote end of the connection as it was most recently successfully negotiated since the last time this 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.
iceConnectionState Read only: Returns a string which state of the ICE agent associated with this RTCPeerConnection. It can be one of the following values: new, checking, connected, completed, failed, disconnected, or closed.
iceGatheringState Read only: Returns a string that describes connection's ICE gathering state. This lets you detect, for example, when collection of ICE candidates has finished. Possible values are: new, gathering, or complete.
localDescription Read only: Returns an RTCSessionDescription describing the session for the local end of the connection. If it has not yet been set, returns null.
peerIdentity Read only: Returns a Promise that resolves to an RTCIdentityAssertion which contains a string identifying the remote peer. Once this promise resolves successfully, the resulting identity is the target peer identity and will not change for the duration of the connection.
pendingLocalDescription Read only: 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.
pendingRemoteDescription Read only: 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.
remoteDescription Read only: Returns an RTCSessionDescription object describing the session, including configuration and media information, for the remote end of the connection. If this hasn't been set yet, this returns null.
sctp Read only: Returns an RTCSctpTransport object describing the SCTP transport layer over which SCTP data is being sent and received. If SCTP hasn't been negotiated, this value is null.
signalingState Read only: Returns a string describing the state of the signaling process on the local end of the connection while connecting or reconnecting to another peer. It is one of the following values: stable, have-local-offer, have-remote-offer, have-local-pranswer, have-remote-pranswer, or closed.

Static methods

RTCPeerConnection.generateCertificate(): Creates an X.509 certificate and its corresponding private key, returning a Promise that resolves with the new RTCCertificate once it is generated.

Instance methods

Also inherits methods from EventTarget.

addIceCandidate(): Adds a new remote candidate to the RTCPeerConnection's remote description, which describes the state of the remote end of the connection.
addTrack(): Adds a new MediaStreamTrack to the set of tracks which will be transmitted to the other peer.
addTransceiver(): Creates a new RTCRtpTransceiver and adds it to the set of transceivers associated with the connection. Each transceiver represents a bidirectional stream, with both an RTCRtpSender and an RTCRtpReceiver associated with it.
close(): Closes the current peer connection.
createAnswer(): Initiates the creation of 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.
createDataChannel(): Initiates the creation a new channel linked with the remote peer, over which any kind of data may be transmitted. This can be useful for back-channel content, such as images, file transfer, text chat, game update packets, and so forth.
createOffer(): Initiates the creation of an SDP offer for the purpose of starting a new WebRTC connection to a remote peer. The SDP offer includes information about any MediaStreamTrack objects already attached to the WebRTC session, codec, and options supported by the browser, as well as any candidates already gathered by the ICE agent, for the purpose of being sent over the signaling channel to a potential peer to request a connection or to update the configuration of an existing connection.
getConfiguration(): Returns an object which indicates the current configuration of the connection.
getIdentityAssertion(): Initiates the gathering of an identity assertion and returns a Promise which resolves to an identity assertion encoded as a string. This has an effect only if signalingState is not closed.
getReceivers(): Returns an array of RTCRtpReceiver objects, each of which represents one RTP receiver.
getSenders(): Returns an array of RTCRtpSender objects, each of which represents the RTP sender responsible for transmitting one track's data.
getStats(): Returns a Promise which resolves with data providing statistics about either the overall connection or about the specified MediaStreamTrack.
getTransceivers(): Returns a list of all the RTCRtpTransceiver objects being used to send and receive data on the connection.
removeTrack(): 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 getSenders(). If the track is already stopped, or is not in the connection's senders list, this method has no effect.
restartIce(): Allows to easily request that ICE candidate gathering be redone on both ends of the connection. This simplifies the process by allowing the same method to be used by either the caller or the receiver to trigger an ICE restart.
setConfiguration(): Sets the current configuration of the connection based on the values included in the specified object. This lets you change the ICE servers used by the connection and which transport policies to use.
setIdentityProvider(): Sets the Identity Provider (IdP) to the triplet given in parameter: its name, the protocol used to communicate with it and an username. The protocol and the username are optional.
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. It returns a Promise which is fulfilled once the description has been changed, asynchronously.
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. It returns a Promise which is fulfilled once the description has been changed, asynchronously.

Obsolete methods

addStream() Deprecated Non-standard: Adds a MediaStream as a local source of audio or video. Instead of using this obsolete method, you should instead use addTrack() once for each track you wish to send to the remote peer.
createDTMFSender() Deprecated Non-standard: Creates a new RTCDTMFSender, associated to a specific MediaStreamTrack, that will be able to send DTMF phone signaling over the connection.
removeStream() Deprecated Non-standard: Removes a MediaStream as a local source of audio or video. Because this method is obsolete, you should instead use removeTrack().

Events

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

connectionstatechange: Sent when the overall connectivity status of the RTCPeerConnection changes.
datachannel: Sent when the remote peer adds an RTCDataChannel to the connection.
icecandidate: Sent to request that the specified candidate be transmitted to the remote peer.
icecandidateerror: Sent to the connection if an error occurred during ICE candidate gathering. The event describes the error.
iceconnectionstatechange: Sent when the state of the ICE connection changes, such as when it disconnects.
icegatheringstatechange: Sent 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).
negotiationneeded: Sent 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 is necessary to adapt to changing network conditions. The receiver should respond by creating an offer and sending it to the other peer.
signalingstatechange: Sent when the connection's ICE signaling state changes.
track: Sent after a new track has been added to one of the RTCRtpReceiver instances which comprise the connection.

Obsolete events

addstream Deprecated Non-standard: Sent when a new MediaStream has been added to the connection. Instead of listening for this obsolete event, you should listen for track events; one is sent for each MediaStreamTrack added to the connection.
removestream Deprecated Non-standard: Sent when a MediaStream is removed from the connection. Instead of listening for this obsolete event, you should listen for removetrack events on each stream.

Specifications

Specification
WebRTC: Real-Time Communication in Browsers # interface-definition

Browser compatibility

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This page was last modified on ⁨Sep 25, 2025⁩ by MDN contributors.

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