webrtc package#

class webrtc.CricketIceGatheringState(value: int)#

Bases: object

Members:

new

gathering

complete

complete: wrtc.CricketIceGatheringState#
gathering: wrtc.CricketIceGatheringState#
property name: str#

str

Type

type

new: wrtc.CricketIceGatheringState#
property value: int#

int

Type

type

class webrtc.DtlsTransportState(value: int)#

Bases: object

Members:

new

connecting

connected

closed

failed

closed: wrtc.DtlsTransportState#
connected: wrtc.DtlsTransportState#
connecting: wrtc.DtlsTransportState#
failed: wrtc.DtlsTransportState#
property name: str#

str

Type

type

new: wrtc.DtlsTransportState#
property value: int#

int

Type

type

class webrtc.MediaStream(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The MediaStream interface represents a stream of media content. A stream consists of several tracks, such as video or audio tracks. Each track is specified as an instance of webrtc.MediaStreamTrack.

property active: bool#

A value that returns true if the webrtc.MediaStream is active, or false otherwise.

Type

bool

addTrack(track: webrtc.MediaStreamTrack)#

Alias for add_track

add_track(track: webrtc.MediaStreamTrack)#

Stores a copy of the webrtc.MediaStreamTrack given as argument. If the track has already been added to the webrtc.MediaStream object, nothing happens.

clone() → webrtc.MediaStream#

Returns a clone of the webrtc.MediaStream object. The clone will, however, have a unique value for id.

getAudioTracks() → List[webrtc.MediaStreamTrack]#

Alias for get_audio_tracks

getTrackById(track_id: str) → Optional[webrtc.MediaStreamTrack]#

Alias for get_track_by_id

getTracks() → List[webrtc.MediaStreamTrack]#

Alias for get_tracks

getVideoTracks() → List[webrtc.MediaStreamTrack]#

Alias for get_video_tracks

get_audio_tracks() → List[webrtc.MediaStreamTrack]#

Returns a list of the webrtc.MediaStreamTrack objects stored in the webrtc.MediaStream object that have their kind attribute set to “audio”. The order is not defined, and may not only vary from one machine to another, but also from one call to another.

get_track_by_id(track_id: str) → Optional[webrtc.MediaStreamTrack]#

Returns the track whose ID corresponds to the one given in parameters, track_id. If no parameter is given, or if no track with that ID does exist, it returns None. If several tracks have the same ID, it returns the first one.

get_tracks() → List[webrtc.MediaStreamTrack]#

Returns a list of all webrtc.MediaStreamTrack objects stored in the webrtc.MediaStream object, regardless of the value of the kind attribute. The order is not defined, and may not only vary from one machine to another, but also from one call to another.

get_video_tracks() → List[webrtc.MediaStreamTrack]#

Returns a list of the webrtc.MediaStreamTrack objects stored in the webrtc.MediaStream object that have their kind attribute set to “video”. The order is not defined, and may not only vary from one machine to another, but also from one call to another.

property id: str#

A String containing 36 characters denoting a universally unique identifier (UUID) for the object.

Type

str

removeTrack(track: webrtc.MediaStreamTrack)#

Alias for remove_track

remove_track(track: webrtc.MediaStreamTrack)#

Removes the webrtc.MediaStreamTrack given as argument. If the track is not part of the webrtc.MediaStream object, nothing happens.

class webrtc.MediaStreamSourceState(value: int)#

Bases: object

Members:

initializing

live

ended

muted

ended: wrtc.MediaStreamSourceState#
initializing: wrtc.MediaStreamSourceState#
live: wrtc.MediaStreamSourceState#
muted: wrtc.MediaStreamSourceState#
property name: str#

str

Type

type

property value: int#

int

Type

type

class webrtc.MediaStreamTrack(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The MediaStreamTrack interface represents a single media track within a stream; typically, these are audio or video tracks, but other track types may exist as well.

clone() → webrtc.MediaStreamTrack#

Returns a duplicate of the webrtc.MediaStreamTrack.

property enabled: bool#

A Boolean whose value of true if the track is enabled, that is allowed to render the media source stream; or false if it is disabled, that is not rendering the media source stream but silence and blackness. If the track has been disconnected, this value can be changed but has no more effect.

Type

bool

property id: str#

A unique identifier (GUID) for the track.

Type

bool

property kind: webrtc.MediaType#

Indicating type of media. Audio or video. It doesn’t change if the track is deassociated from its source.

Type

webrtc.MediaType

property muted: bool#

A value indicating whether the track is unable to provide media data due to a technical issue.

Type

bool

property readyState: webrtc.MediaStreamTrackState#

Alias for ready_state

property ready_state: webrtc.MediaStreamTrackState#

Returns an enumerated value giving the status of the track.

Type

webrtc.MediaStreamTrackState

stop()#

Stops playing the source associated to the track, both the source and the track are deassociated. The track state is set to ended.

class webrtc.MediaStreamTrackState(value: int)#

Bases: object

Members:

live

ended

ended: wrtc.MediaStreamTrackState#
live: wrtc.MediaStreamTrackState#
property name: str#

str

Type

type

property value: int#

int

Type

type

class webrtc.MediaType(value: int)#

Bases: object

Members:

audio

video

data

unsupported

audio: wrtc.MediaType#
data: wrtc.MediaType#
property name: str#

str

Type

type

unsupported: wrtc.MediaType#
property value: int#

int

Type

type

video: wrtc.MediaType#
exception webrtc.PythonWebRTCException#

Bases: wrtc.PythonWebRTCExceptionBase, Exception, BaseException

exception webrtc.PythonWebRTCExceptionBase#

Bases: Exception, BaseException

class webrtc.RTCAudioSource(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The webrtc.MediaStreamTrack interface represents a single media track within a stream; typically, these are audio or video tracks, but other track types may exist as well.

createTrack() → webrtc.MediaStreamTrack#

Alias for create_track

create_track() → webrtc.MediaStreamTrack#

Create webrtc.MediaStreamTrack whose source is the webrtc.RTCAudioSource.

Returns

A webrtc.MediaStreamTrack object representing the media track.

Return type

webrtc.MediaStreamTrack

onData(data: webrtc.RTCOnDataEvent)#

Alias for on_data

on_data(data: webrtc.RTCOnDataEvent)#

Push a new audio samples to every non-stopped local audio webrtc.MediaStreamTrack created with createTrack.

Parameters

data (webrtc.RTCOnDataEvent) – A webrtc.RTCOnDataEvent object representing new audio samples.

Return type

None

class webrtc.RTCDtlsTransport(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The webrtc.RTCDtlsTransport interface provides access to information about the Datagram Transport Layer Security (DTLS) transport over which a webrtc.RTCPeerConnection’s RTP and RTCP packets are sent and received by its webrtc.RTCRtpSender and webrtc.RTCRtpReceiver objects.

property iceTransport: webrtc.RTCIceTransport#

Alias for ice_transport

property ice_transport: webrtc.RTCIceTransport#

Returns a reference to the underlying webrtc.RTCIceTransport object.

Type

webrtc.RTCIceTransport

property state: webrtc.DtlsTransportState#

Returns a member of webrtc.DtlsTransportState which describes the underlying Datagram Transport Layer Security (DTLS) transport state.

Type

webrtc.DtlsTransportState

exception webrtc.RTCException#

Bases: wrtc.PythonWebRTCExceptionBase, Exception, BaseException

class webrtc.RTCIceComponent(value: int)#

Bases: object

Members:

RTP

RTCP

RTCP: wrtc.RTCIceComponent#
RTP: wrtc.RTCIceComponent#
property name: str#

str

Type

type

property value: int#

int

Type

type

class webrtc.RTCIceConnectionState(value: int)#

Bases: object

Members:

new

checking

connected

completed

failed

disconnected

closed

max

checking: wrtc.RTCIceConnectionState#
closed: wrtc.RTCIceConnectionState#
completed: wrtc.RTCIceConnectionState#
connected: wrtc.RTCIceConnectionState#
disconnected: wrtc.RTCIceConnectionState#
failed: wrtc.RTCIceConnectionState#
max: wrtc.RTCIceConnectionState#
property name: str#

str

Type

type

new: wrtc.RTCIceConnectionState#
property value: int#

int

Type

type

class webrtc.RTCIceGatheringState(value: int)#

Bases: object

Members:

new

gathering

complete

complete: wrtc.RTCIceGatheringState#
gathering: wrtc.RTCIceGatheringState#
property name: str#

str

Type

type

new: wrtc.RTCIceGatheringState#
property value: int#

int

Type

type

class webrtc.RTCIceRole(value: int)#

Bases: object

Members:

controlling

controlled

unknown

controlled: wrtc.RTCIceRole#
controlling: wrtc.RTCIceRole#
property name: str#

str

Type

type

unknown: wrtc.RTCIceRole#
property value: int#

int

Type

type

class webrtc.RTCIceTransport(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The webrtc.RTCIceTransport interface provides access to information about the ICE transport layer over which the data is being sent and received. This is particularly useful if you need to access state information about the connection.

property component: webrtc.RTCIceComponent#

The ICE component being used by the transport. The value is one of the strings from the webrtc.RTCIceComponent enumerated type: “RTP” or “RTSP”.

property gatheringState: webrtc.CricketIceGatheringState#

Alias for gathering_state

property gathering_state: webrtc.CricketIceGatheringState#

A member of webrtc.CricketIceGatheringState enum, indicating which current gathering state of the ICE agent

property role: webrtc.RTCIceRole#

A member of webrtc.RTCIceRole; this indicates whether the ICE agent is the one that makes the final decision as to the candidate pair to use or not.

property state: webrtc.RTCIceTransportState#

A member of webrtc.RTCIceTransportState indicating what the current state of the ICE agent is.

Note

For more details of values: https://developer.mozilla.org/en-US/docs/Web/API/RTCIceTransportState

class webrtc.RTCIceTransportState(value: int)#

Bases: object

Members:

new

checking

connected

completed

disconnected

failed

closed

checking: wrtc.RTCIceTransportState#
closed: wrtc.RTCIceTransportState#
completed: wrtc.RTCIceTransportState#
connected: wrtc.RTCIceTransportState#
disconnected: wrtc.RTCIceTransportState#
failed: wrtc.RTCIceTransportState#
property name: str#

str

Type

type

new: wrtc.RTCIceTransportState#
property value: int#

int

Type

type

class webrtc.RTCOnDataEvent(audio_data: AnyStr, number_of_frames: int)#

Bases: webrtc.base.WebRTCObject

Audio data samples representation.

Note

webrtc.RTCOnDataEvent should represent 10 ms of audio samples.

property audioData: AnyStr#

Alias for audio_data

property audio_data: AnyStr#

An audio data.

Type

AnyStr

property bitsPerSample: int#

Alias for bits_per_sample

property bits_per_sample: int#

Bits per sample.

Type

int

property channelCount: int#

Alias for channel_count

property channel_count: int#

A channel count.

Type

int

property numberOfFrames: int#

Alias for number_of_frames

property number_of_frames: int#

A number of frames.

Type

int

property sampleRate: int#

Alias for sample_rate

property sample_rate: int#

A sample rate.

Type

int

class webrtc.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
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
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

class webrtc.RTCPeerConnectionState(value: int)#

Bases: object

Members:

new

connecting

connected

disconnected

failed

closed

closed: wrtc.RTCPeerConnectionState#
connected: wrtc.RTCPeerConnectionState#
connecting: wrtc.RTCPeerConnectionState#
disconnected: wrtc.RTCPeerConnectionState#
failed: wrtc.RTCPeerConnectionState#
property name: str#

str

Type

type

new: wrtc.RTCPeerConnectionState#
property value: int#

int

Type

type

class webrtc.RTCRtpReceiver(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The webrtc.RTCRtpReceiver interface of the WebRTC API manages the reception and decoding of data for a webrtc.MediaStreamTrack on an webrtc.RTCPeerConnection.

property track: webrtc.MediaStreamTrack#

The webrtc.MediaStreamTrack associated with the current webrtc.RTCRtpReceiver instance.

Type

webrtc.MediaStreamTrack

property transport: Optional[webrtc.RTCDtlsTransport]#

An object representing the underlying transport being used by the receiver to exchange packets with the remote peer, or null if the receiver isn’t yet connected to transport.

Type

webrtc.RTCDtlsTransport

class webrtc.RTCRtpSender(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The webrtc.MediaStreamTrack interface represents a single media track within a stream; typically, these are audio or video tracks, but other track types may exist as well.

property track: Optional[webrtc.MediaStreamTrack]#

The webrtc.MediaStreamTrack which is being handled by the webrtc.RTCRtpSender. If track is None, the webrtc.RTCRtpSender doesn’t transmit anything.

Type

webrtc.MediaStreamTrack, optional

property transport: Optional[webrtc.RTCDtlsTransport]#

An object representing the underlying transport being used by the sender to exchange packets with the remote peer, or null if the sender isn’t yet connected to transport.

Type

webrtc.RTCDtlsTransport

class webrtc.RTCRtpTransceiver(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The WebRTC interface webrtc.RTCRtpTransceiver describes a permanent pairing of an webrtc.RTCRtpSender and an webrtc.RTCRtpReceiver, along with some shared state.

property currentDirection: Optional[webrtc.TransceiverDirection]#

Alias for current_direction

property current_direction: Optional[webrtc.TransceiverDirection]#

A member of webrtc.TransceiverDirection enum, indicating the current directionality of the transceiver.

property direction: webrtc.TransceiverDirection#

A member of webrtc.TransceiverDirection enum, indicating the transceiver’s preferred direction.

Note

The transceiver’s current direction is indicated by the currentDirection property.

property mid: Optional[str]#

A str which uniquely identifies the pairing of source and destination of the transceiver’s stream. Its value is taken from the media ID of the SDP m-line. This value is None if negotiation has not completed.

property receiver: webrtc.RTCRtpReceiver#

A webrtc.RTCRtpReceiver object which is responsible for receiving and decoding incoming media data whose media ID is the same as the current value of mid.

property sender: webrtc.RTCRtpSender#

A webrtc.RTCRtpSender object used to encode and send media whose media ID matches the current value of mid.

stop() → None#

Permanently stops the transceiver by stopping both the associated webrtc.RTCRtpSender and webrtc.RTCRtpReceiver.

Note

The stopped property was provided to return True if the connection is stopped. That property has been deprecated and will be removed at some point. Instead, check the value of currentDirection. If it’s webrtc.TransceiverDirection.stopped, the transceiver has been stopped.

property stopped: bool#

A bool value which is True if the transceiver’s sender will no longer send data, and its receiver will no longer receive data. If either or both are still at work, the result is False.

Warning

Deprecated: This feature is no longer recommended.

class webrtc.RTCSctpTransport(native_obj=None)#

Bases: webrtc.base.WebRTCObject

The webrtc.RTCSctpTransport interface provides information which describes a Stream Control Transmission Protocol (SCTP) transport. This provides information about limitations of the transport, but also provides a way to access the underlying Datagram Transport Layer Security (DTLS) transport over which SCTP packets for all of an webrtc.RTCPeerConnection’s data channels are sent and received.

property maxChannels: Optional[int]#

Alias for max_channels

property maxMessageSize: Optional[float]#

Alias for max_message_size

property max_channels: Optional[int]#

An integer value indicating the maximum number of webrtc.RTCDataChannel that can be open simultaneously.

Type

int, optional

property max_message_size: Optional[float]#

An integer value indicating the maximum size, in bytes, of a message which can be sent using the webrtc.RTCDataChannel.send method.

Type

float, optional

property state: webrtc.SctpTransportState#

A enumerated value indicating the state of the SCTP transport.

Type

webrtc.SctpTransportState

property transport: webrtc.RTCDtlsTransport#

An object representing the DTLS transport used for the transmission and receipt of data packets.

Type

webrtc.RTCDtlsTransport

class webrtc.RTCSdpType(value: int)#

Bases: object

Members:

offer

pranswer

answer

rollback

answer: wrtc.RTCSdpType#
property name: str#

str

Type

type

offer: wrtc.RTCSdpType#
pranswer: wrtc.RTCSdpType#
rollback: wrtc.RTCSdpType#
property value: int#

int

Type

type

class webrtc.RTCSessionDescription(rtc_session_description_init: webrtc.RTCSessionDescriptionInit)#

Bases: webrtc.base.WebRTCObject

The webrtc.RTCSessionDescription interface describes one end of a connection or potential connection and how it’s configured. Each webrtcRTCSessionDescription consists of a description type indicating which part of the offer/answer negotiation process it describes and of the SDP descriptor of the session.

The process of negotiating a connection between two peers involves exchanging webrtc.RTCSessionDescription objects back and forth, with each description suggesting one combination of connection configuration options that the sender of the description supports. Once the two peers agree upon a configuration for the connection, negotiation is complete.

Note

Note: Constructor with webrtc.RTCSessionDescriptionInit is no longer necessary, however; RTCPeerConnection.setLocalDescription() and other methods which take SDP as input now directly accept an object conforming to the :obj:webrtc.RTCSessionDescriptionInit` object, so you don’t have to instantiate an webrtc.RTCSessionDescription yourself.

Warning

Deprecated: Constructor with webrtc.RTCSessionDescriptionInit is no longer recommended. Though some browsers might still support it, it may have already been removed from the relevant web standards, may be in the process of being dropped, or may only be kept for compatibility purposes.

property sdp#

A string containing a SDP message describing the session.

Type

str

property type: webrtc.RTCSdpType#

A member of the webrtc.RTCSdpType enum.

Type

webrtc.RTCSdpType

class webrtc.RTCSessionDescriptionInit(type: webrtc.RTCSdpType, sdp='')#

Bases: webrtc.base.WebRTCObject

An object providing the default values for the session description.

property sdp: str#

A string containing a SDP message describing the session. This value is an empty string (‘’) by default and may not be None.

Type

str

property type: webrtc.RTCSdpType#

A member of the webrtc.RTCSdpType enum.

Type

webrtc.RTCSdpType

class webrtc.RtpEncodingParameters(active: Optional[bool] = True, max_bitrate: Optional[int] = None, max_framerate: Optional[float] = None, rid: Optional[str] = None, scale_resolution_down_by: Optional[float] = None)#

Bases: webrtc.base.WebRTCObject

Model describes a single configuration of a codec for an webrtc.RTCRtpSender.

property active: bool#

If true, the described encoding is currently actively being used. That is, for RTP senders, the encoding is currently being used to send data, while for receivers, the encoding is being used to decode received data. The default value is true.

Type

bool

property maxBitrate: Optional[int]#

Alias for max_bitrate

property maxFramerate: Optional[int]#

Alias for max_framerate

property max_bitrate: Optional[int]#

An unsigned long integer indicating the maximum number of bits per second to allow for this encoding. Other parameters may further constrain the bit rate, such as the value of max_framerate or transport or physical network limitations.

Type

int, optional

property max_framerate: Optional[int]#

A double-precision floating-point value specifying the maximum number of frames per second to allow for this encoding.

Type

int, optional

property rid: Optional[str]#

A string which, if set, specifies an RTP stream ID (RID) to be sent using the RID header extension. This parameter cannot be modified using webrtc.RTCRtpSender.set_parameters. Its value can only be set when the transceiver is first created.

Type

str, optional

property scaleResolutionDownBy: Optional[float]#

Alias for scale_resolution_down_by

property scale_resolution_down_by: Optional[float]#

Only used for senders whose track’s kind is video, this is a double-precision floating-point value specifying a factor by which to scale down the video during encoding. The default value, 1.0, means that the sent video’s size will be the same as the original. A value of 2.0 scales the video frames down by a factor of 2 in each dimension, resulting in a video 1/4 the size of the original. The value must not be less than 1.0 (you can’t use this to scale the video up).

Type

float, optional

class webrtc.RtpTransceiverInit(direction: Optional[webrtc.RtpTransceiverDirection] = None, send_encodings: Optional[List[webrtc.RtpEncodingParameters]] = None, streams: Optional[List[webrtc.MediaStream]] = None)#

Bases: webrtc.base.WebRTCObject

A model for specifying any options when creating the new transceiver.

property direction: webrtc.TransceiverDirection#

The new transceiver’s preferred directionality. This value is used to initialize the new webrtc.RTCRtpTransceiver object’s webrtc.RTCRtpTransceiver.direction property.

Type

webrtc.TransceiverDirection

property sendEncodings: List[webrtc.RtpEncodingParameters]#

Alias for send_encodings

property send_encodings: List[webrtc.RtpEncodingParameters]#

A list of encodings to allow when sending RTP media from the webrtc.RTCRtpSender. Each entry is of type webrtc.RtpEncodingParameters.

Type

list of webrtc.RtpEncodingParameters

property streams: List[webrtc.MediaStream]#

A list of webrtc.MediaStream objects to add to the transceiver’s webrtc.RTCRtpReceiver; when the remote peer’s webrtc.RTCPeerConnection’s track event occurs, these are the streams that will be specified by that event.

Type

list of webrtc.MediaStream

class webrtc.SctpTransportState(value: int)#

Bases: object

Members:

new

connecting

connected

closed

closed: wrtc.SctpTransportState#
connected: wrtc.SctpTransportState#
connecting: wrtc.SctpTransportState#
property name: str#

str

Type

type

new: wrtc.SctpTransportState#
property value: int#

int

Type

type

exception webrtc.SdpParseException#

Bases: wrtc.PythonWebRTCExceptionBase, Exception, BaseException

class webrtc.TransceiverDirection(value: int)#

Bases: object

Members:

sendrecv

sendonly

recvonly

inactive

stopped

inactive: wrtc.TransceiverDirection#
property name: str#

str

Type

type

recvonly: wrtc.TransceiverDirection#
sendonly: wrtc.TransceiverDirection#
sendrecv: wrtc.TransceiverDirection#
stopped: wrtc.TransceiverDirection#
property value: int#

int

Type

type

class webrtc.WebRTCObject(native_obj=None)#

Bases: object

webrtc.getUserMedia(constraints=None) → webrtc.MediaStream#

Method prompts the user for permission to use a media input which produces a webrtc.MediaStream with tracks containing the requested types of media.

Returns

A webrtc.MediaStream object representing the media stream.

Return type

webrtc.MediaStream

webrtc.get_user_media(constraints=None) → webrtc.MediaStream#

Method prompts the user for permission to use a media input which produces a webrtc.MediaStream with tracks containing the requested types of media.

Returns

A webrtc.MediaStream object representing the media stream.

Return type

webrtc.MediaStream

Subpackages#

Submodules#