webrtc.interfaces.rtc_peer_connection#

class webrtc.interfaces.rtc_peer_connection.RTCPeerConnection(native_obj=None)#

Bases: webrtc.base.WebRTCObject

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.

addTrack(track: webrtc.MediaStreamTrack, stream: Optional[Union[webrtc.MediaStream, List[webrtc.MediaStream]]] = None) → webrtc.RTCRtpSender#: Alias for add_track

addTransceiver(track_or_kind: Union[webrtc.MediaStreamTrack, webrtc.MediaType], init: Optional[webrtc.RtpTransceiverInit] = None) → webrtc.RTCRtpTransceiver#: Alias for add_transceiver

add_track(track: webrtc.MediaStreamTrack, stream: Optional[Union[webrtc.MediaStream, List[webrtc.MediaStream]]] = None) → webrtc.RTCRtpSender#

Adds a new webrtc.MediaStreamTrack to the set of tracks which will be transmitted to the other peer.

Parameters

track (webrtc.MediaStreamTrack) – A webrtc.MediaStreamTrack object representing the media track to add to the peer connection.
stream (webrtc.MediaStream or list of webrtc.MediaStream, optional) – One or more local webrtc.MediaStream objects to which the track should be added.

Returns

The webrtc.RTCRtpSender object which will be used to transmit the media data.

Return type

webrtc.RTCRtpSender

add_transceiver(track_or_kind: Union[webrtc.MediaStreamTrack, webrtc.MediaType], init: Optional[webrtc.RtpTransceiverInit] = None) → webrtc.RTCRtpTransceiver#

Creates a new webrtc.RTCRtpTransceiver and adds it to the set of transceivers associated with the connection. Each transceiver represents a bidirectional stream, with both an webrtc.RTCRtpSender and an webrtc.RTCRtpReceiver associated with it.

Parameters

track_or_kind (webrtc.MediaStreamTrack) – A webrtc.MediaStreamTrack to associate with the transceiver, or a member of webrtc.MediaType which is used as the kind of the receiver’s track, and by extension of the webrtc.RTCRtpReceiver itself.
init (webrtc.RtpTransceiverInit, optional) – An object for specifying any options when creating the new transceiver.

Returns

The webrtc.RTCRtpSender object which will be used to transmit the media data.

Return type

webrtc.RTCRtpSender

close()#: Closes the current peer connection.

async createAnswer() → webrtc.RTCSessionDescription#: Alias for create_answer

async createOffer() → webrtc.RTCSessionDescription#: Alias for create_offer

async create_answer() → webrtc.RTCSessionDescription#: Initiates the creation 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 machine, and any ICE candidates already gathered.

async create_offer() → webrtc.RTCSessionDescription#: 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 machine, 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.

getReceivers() → List[webrtc.RTCRtpReceiver]#: Alias for get_receivers

getSenders() → List[webrtc.RTCRtpSender]#: Alias for get_senders

getTransceivers() → List[webrtc.RTCRtpTransceiver]#: Alias for get_transceivers

get_receivers() → List[webrtc.RTCRtpReceiver]#

Returns an array of webrtc.RTCRtpReceiver objects, each of which represents one RTP receiver. Each RTP receiver manages the reception and decoding of data for a webrtc.MediaStreamTrack on an webrtc.RTCPeerConnection.

Note

The order of the returned webrtc.RTCRtpReceiver is not defined by the specification, and may change from one call to get_receivers to the next.

Returns

An array of webrtc.RTCRtpReceiver objects, one for each: track on the connection. The array is empty if there are no RTP receivers on the connection.

Return type

list of webrtc.RTCRtpReceiver

get_senders() → List[webrtc.RTCRtpSender]#

Returns an array of webrtc.RTCRtpSender objects, each of which represents the RTP sender responsible for transmitting one track’s data. A sender object provides methods and properties for examining and controlling the encoding and transmission of the track’s data.

Note

The order of the returned webrtc.RTCRtpSenders is not defined by the specification, and may change from one call to get_senders to the next.

Returns: An array of webrtc.RTCRtpSender objects, one for each track on the connection. The array is empty if there are no RTP senders on the connection.
Return type: list of webrtc.RTCRtpSender

get_transceivers() → List[webrtc.RTCRtpTransceiver]#

Returns a list of the webrtc.RTCRtpTransceiver objects being used to send and receive data on the connection.

Returns: An array of the webrtc.RTCRtpTransceiver objects representing the transceivers handling sending and receiving all media on the webrtc.RTCPeerConnection. The list is in the order in which the transceivers were added to the connection.
Return type: list of webrtc.RTCRtpTransceiver

removeTrack(sender: webrtc.RTCRtpSender) → None#: Alias for remove_track

remove_track(sender: webrtc.RTCRtpSender) → None#: Tells the local end of the connection to stop sending media from the specified track, without actually removing the corresponding webrtc.RTCRtpSender from the list of senders as reported by get_senders`. If the track is already stopped, or is not in the connection’s senders list, this method has no effect.

restartIce() → None#: Alias for restart_ice

restart_ice() → None#: 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.

property sctp: Optional[webrtc.RTCSctpTransport]#

An object describing the SCTP transport layer over which SCTP data is being sent and received. If SCTP hasn’t been negotiated, this value is None

Type: webrtc.RTCSctpTransport, optional

async setLocalDescription(sdp: webrtc.RTCSessionDescription)#: Alias for set_local_description

async setRemoteDescription(sdp: webrtc.RTCSessionDescription)#: Alias for set_remote_description

async set_local_description(sdp: webrtc.RTCSessionDescription)#

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 Coroutine. A result will be returned once the description has been changed, asynchronously.

Parameters: sdp (webrtc.RTCSessionDescription) – A webrtc.RTCSessionDescription object that describes one end of a connection or potential connection.
Return type: None

async set_remote_description(sdp: webrtc.RTCSessionDescription)#

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 Coroutine. A result will be returned once the description has been changed, asynchronously.

Parameters: sdp (webrtc.RTCSessionDescription) – A webrtc.RTCSessionDescription object that describes one end of a connection or potential connection.
Return type: None