# GstWebRTC Enumerations

## GstWebRTCDTLSTransport

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstWebRTCDTLSTransport

### Class structure

#### GstWebRTCDTLSTransportClass

### GstWebRTC.WebRTCDTLSTransportClass

## GstWebRTC.WebRTCDTLSTransport

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCDTLSTransport

## GstWebRTC.WebRTCDTLSTransport

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCDTLSTransport

### Properties

#### certificate

    “certificate” gchar *

Flags : Read / Write

#### certificate

    “certificate” String

Flags : Read / Write

#### certificate

    “self.props.certificate” str

Flags : Read / Write

#### client

    “client” gboolean

Flags : Read / Write

#### client

    “client” Number

Flags : Read / Write

#### client

    “self.props.client” bool

Flags : Read / Write

#### remote-certificate

    “remote-certificate” gchar *

Flags : Read

#### remote-certificate

    “remote-certificate” String

Flags : Read

#### remote_certificate

    “self.props.remote_certificate” str

Flags : Read

#### session-id

    “session-id” guint

Flags : Read / Write / Construct Only

#### session-id

    “session-id” Number

Flags : Read / Write / Construct Only

#### session_id

    “self.props.session_id” int

Flags : Read / Write / Construct Only

#### state

    “state” GstWebRTCDTLSTransportState *

Flags : Read

#### state

    “state” GstWebRTC.WebRTCDTLSTransportState

Flags : Read

#### state

    “self.props.state” GstWebRTC.WebRTCDTLSTransportState

Flags : Read

#### transport

    “transport” GstWebRTCICETransport *

Flags : Read

#### transport

    “transport” GstWebRTC.WebRTCICETransport

Flags : Read

#### transport

    “self.props.transport” GstWebRTC.WebRTCICETransport

Flags : Read

## GstWebRTCDataChannel

    GObject
        ╰──GstWebRTCDataChannel

### Class structure

#### GstWebRTCDataChannelClass

### GstWebRTC.WebRTCDataChannelClass

## GstWebRTC.WebRTCDataChannel

    GObject.Object
        ╰──GstWebRTC.WebRTCDataChannel

## GstWebRTC.WebRTCDataChannel

    GObject.Object
        ╰──GstWebRTC.WebRTCDataChannel

### Methods

#### gst_webrtc_data_channel_close

    gst_webrtc_data_channel_close (GstWebRTCDataChannel * channel)

Close the channel.

Parameters:

`channel` –

a GstWebRTCDataChannel

#### GstWebRTC.WebRTCDataChannel.prototype.close

    function GstWebRTC.WebRTCDataChannel.prototype.close(): {
        // javascript wrapper for 'gst_webrtc_data_channel_close'
    }

Close the channel.

Parameters:

`channel` (GstWebRTC.WebRTCDataChannel) –

a GstWebRTC.WebRTCDataChannel

#### GstWebRTC.WebRTCDataChannel.close

    def GstWebRTC.WebRTCDataChannel.close (self):
        #python wrapper for 'gst_webrtc_data_channel_close'

Close the channel.

Parameters:

`channel` (GstWebRTC.WebRTCDataChannel) –

a GstWebRTC.WebRTCDataChannel

#### gst_webrtc_data_channel_send_data

    gst_webrtc_data_channel_send_data (GstWebRTCDataChannel * channel,
                                       GBytes * data)

Send data as a data message over channel.

Parameters:

`channel` –

a GstWebRTCDataChannel

`data` ( [nullable]) –

a GBytes or NULL

#### GstWebRTC.WebRTCDataChannel.prototype.send_data

    function GstWebRTC.WebRTCDataChannel.prototype.send_data(data: GLib.Bytes): {
        // javascript wrapper for 'gst_webrtc_data_channel_send_data'
    }

Send data as a data message over channel.

Parameters:

`channel` (GstWebRTC.WebRTCDataChannel) –

a GstWebRTC.WebRTCDataChannel

`data` (GLib.Bytes) –

a GLib.Bytes or null

#### GstWebRTC.WebRTCDataChannel.send_data

    def GstWebRTC.WebRTCDataChannel.send_data (self, data):
        #python wrapper for 'gst_webrtc_data_channel_send_data'

Send data as a data message over channel.

Parameters:

`channel` (GstWebRTC.WebRTCDataChannel) –

a GstWebRTC.WebRTCDataChannel

`data` (GLib.Bytes) –

a GLib.Bytes or None

#### gst_webrtc_data_channel_send_string

    gst_webrtc_data_channel_send_string (GstWebRTCDataChannel * channel,
                                         const gchar * str)

Send str as a string message over channel.

Parameters:

`channel` –

a GstWebRTCDataChannel

`str` ( [nullable]) –

a string or NULL

#### GstWebRTC.WebRTCDataChannel.prototype.send_string

    function GstWebRTC.WebRTCDataChannel.prototype.send_string(str: String): {
        // javascript wrapper for 'gst_webrtc_data_channel_send_string'
    }

Send str as a string message over channel.

Parameters:

`channel` (GstWebRTC.WebRTCDataChannel) –

a GstWebRTC.WebRTCDataChannel

`str` (String) –

a string or null

#### GstWebRTC.WebRTCDataChannel.send_string

    def GstWebRTC.WebRTCDataChannel.send_string (self, str):
        #python wrapper for 'gst_webrtc_data_channel_send_string'

Send str as a string message over channel.

Parameters:

`channel` (GstWebRTC.WebRTCDataChannel) –

a GstWebRTC.WebRTCDataChannel

`str` (str) –

a string or None

### Signals

#### on-buffered-amount-low

    on_buffered_amount_low_callback (GstWebRTCDataChannel * self,
                                     gpointer user_data)

Parameters:

`self` –
No description available
`user_data` –
No description available

Flags: Run Last

#### on-buffered-amount-low

    function on_buffered_amount_low_callback(self: GstWebRTC.WebRTCDataChannel, user_data: Object): {
        // javascript callback for the 'on-buffered-amount-low' signal
    }

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`user_data` (Object) –
No description available

Flags: Run Last

#### on-buffered-amount-low

    def on_buffered_amount_low_callback (self, *user_data):
        #python callback for the 'on-buffered-amount-low' signal

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`user_data` (variadic) –
No description available

Flags: Run Last

#### on-close

    on_close_callback (GstWebRTCDataChannel * self,
                       gpointer user_data)

Parameters:

`self` –
No description available
`user_data` –
No description available

Flags: Run Last

#### on-close

    function on_close_callback(self: GstWebRTC.WebRTCDataChannel, user_data: Object): {
        // javascript callback for the 'on-close' signal
    }

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`user_data` (Object) –
No description available

Flags: Run Last

#### on-close

    def on_close_callback (self, *user_data):
        #python callback for the 'on-close' signal

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`user_data` (variadic) –
No description available

Flags: Run Last

#### on-error

    on_error_callback (GstWebRTCDataChannel * self,
                       GError * error,
                       gpointer user_data)

Parameters:

`self` –
No description available
`error` –

the GError thrown

`user_data` –
No description available

Flags: Run Last

#### on-error

    function on_error_callback(self: GstWebRTC.WebRTCDataChannel, error: GLib.Error, user_data: Object): {
        // javascript callback for the 'on-error' signal
    }

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`error` (GLib.Error) –

the GLib.Error thrown

`user_data` (Object) –
No description available

Flags: Run Last

#### on-error

    def on_error_callback (self, error, *user_data):
        #python callback for the 'on-error' signal

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`error` (GLib.Error) –

the GLib.Error thrown

`user_data` (variadic) –
No description available

Flags: Run Last

#### on-message-data

    on_message_data_callback (GstWebRTCDataChannel * self,
                              GBytes * data,
                              gpointer user_data)

Parameters:

`self` –
No description available
`data` ( [nullable]) –

a GBytes of the data received

`user_data` –
No description available

Flags: Run Last

#### on-message-data

    function on_message_data_callback(self: GstWebRTC.WebRTCDataChannel, data: GLib.Bytes, user_data: Object): {
        // javascript callback for the 'on-message-data' signal
    }

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`data` (GLib.Bytes) –

a GLib.Bytes of the data received

`user_data` (Object) –
No description available

Flags: Run Last

#### on-message-data

    def on_message_data_callback (self, data, *user_data):
        #python callback for the 'on-message-data' signal

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`data` (GLib.Bytes) –

a GLib.Bytes of the data received

`user_data` (variadic) –
No description available

Flags: Run Last

#### on-message-string

    on_message_string_callback (GstWebRTCDataChannel * self,
                                gchar * data,
                                gpointer user_data)

Parameters:

`self` –
No description available
`data` ( [nullable]) –

the data received as a string

`user_data` –
No description available

Flags: Run Last

#### on-message-string

    function on_message_string_callback(self: GstWebRTC.WebRTCDataChannel, data: String, user_data: Object): {
        // javascript callback for the 'on-message-string' signal
    }

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`data` (String) –

the data received as a string

`user_data` (Object) –
No description available

Flags: Run Last

#### on-message-string

    def on_message_string_callback (self, data, *user_data):
        #python callback for the 'on-message-string' signal

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`data` (str) –

the data received as a string

`user_data` (variadic) –
No description available

Flags: Run Last

#### on-open

    on_open_callback (GstWebRTCDataChannel * self,
                      gpointer user_data)

Parameters:

`self` –
No description available
`user_data` –
No description available

Flags: Run Last

#### on-open

    function on_open_callback(self: GstWebRTC.WebRTCDataChannel, user_data: Object): {
        // javascript callback for the 'on-open' signal
    }

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`user_data` (Object) –
No description available

Flags: Run Last

#### on-open

    def on_open_callback (self, *user_data):
        #python callback for the 'on-open' signal

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`user_data` (variadic) –
No description available

Flags: Run Last

### Action Signals

#### close

    g_signal_emit_by_name (self, "close", user_data);

Close the data channel

Parameters:

`self` (GstWebRTCDataChannel *) –
No description available
`user_data` (gpointer) –
No description available

Flags: Run Last / Action

#### close

    let ret = self.emit ("close", user_data);

Close the data channel

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`user_data` (Object) –
No description available

Flags: Run Last / Action

#### close

    ret = self.emit ("close", user_data)

Close the data channel

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`user_data` (variadic) –
No description available

Flags: Run Last / Action

#### send-data

    g_signal_emit_by_name (self, "send-data", data, user_data);

Parameters:

`self` (GstWebRTCDataChannel *) –
No description available
`data` (GBytes *, [nullable]) –

a GBytes with the data

`user_data` (gpointer) –
No description available

Flags: Run Last / Action

#### send-data

    let ret = self.emit ("send-data", data, user_data);

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`data` (GLib.Bytes) –

a GLib.Bytes with the data

`user_data` (Object) –
No description available

Flags: Run Last / Action

#### send-data

    ret = self.emit ("send-data", data, user_data)

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`data` (GLib.Bytes) –

a GLib.Bytes with the data

`user_data` (variadic) –
No description available

Flags: Run Last / Action

#### send-string

    g_signal_emit_by_name (self, "send-string", data, user_data);

Parameters:

`self` (GstWebRTCDataChannel *) –
No description available
`data` (gchar *, [nullable]) –

the data to send as a string

`user_data` (gpointer) –
No description available

Flags: Run Last / Action

#### send-string

    let ret = self.emit ("send-string", data, user_data);

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`data` (String) –

the data to send as a string

`user_data` (Object) –
No description available

Flags: Run Last / Action

#### send-string

    ret = self.emit ("send-string", data, user_data)

Parameters:

`self` (GstWebRTC.WebRTCDataChannel) –
No description available
`data` (str) –

the data to send as a string

`user_data` (variadic) –
No description available

Flags: Run Last / Action

### Properties

#### buffered-amount

    “buffered-amount” guint64

Flags : Read

#### buffered-amount

    “buffered-amount” Number

Flags : Read

#### buffered_amount

    “self.props.buffered_amount” int

Flags : Read

#### buffered-amount-low-threshold

    “buffered-amount-low-threshold” guint64

Flags : Read / Write

#### buffered-amount-low-threshold

    “buffered-amount-low-threshold” Number

Flags : Read / Write

#### buffered_amount_low_threshold

    “self.props.buffered_amount_low_threshold” int

Flags : Read / Write

#### id

    “id” gint

Flags : Read / Write / Construct Only

#### id

    “id” Number

Flags : Read / Write / Construct Only

#### id

    “self.props.id” int

Flags : Read / Write / Construct Only

#### label

    “label” gchar *

Flags : Read / Write / Construct Only

#### label

    “label” String

Flags : Read / Write / Construct Only

#### label

    “self.props.label” str

Flags : Read / Write / Construct Only

#### max-packet-lifetime

    “max-packet-lifetime” gint

Flags : Read / Write / Construct Only

#### max-packet-lifetime

    “max-packet-lifetime” Number

Flags : Read / Write / Construct Only

#### max_packet_lifetime

    “self.props.max_packet_lifetime” int

Flags : Read / Write / Construct Only

#### max-retransmits

    “max-retransmits” gint

Flags : Read / Write / Construct Only

#### max-retransmits

    “max-retransmits” Number

Flags : Read / Write / Construct Only

#### max_retransmits

    “self.props.max_retransmits” int

Flags : Read / Write / Construct Only

#### negotiated

    “negotiated” gboolean

Flags : Read / Write / Construct Only

#### negotiated

    “negotiated” Number

Flags : Read / Write / Construct Only

#### negotiated

    “self.props.negotiated” bool

Flags : Read / Write / Construct Only

#### ordered

    “ordered” gboolean

Flags : Read / Write / Construct Only

#### ordered

    “ordered” Number

Flags : Read / Write / Construct Only

#### ordered

    “self.props.ordered” bool

Flags : Read / Write / Construct Only

#### priority

    “priority” GstWebRTCPriorityType *

Flags : Read / Write / Construct Only

#### priority

    “priority” GstWebRTC.WebRTCPriorityType

Flags : Read / Write / Construct Only

#### priority

    “self.props.priority” GstWebRTC.WebRTCPriorityType

Flags : Read / Write / Construct Only

#### protocol

    “protocol” gchar *

Flags : Read / Write / Construct Only

#### protocol

    “protocol” String

Flags : Read / Write / Construct Only

#### protocol

    “self.props.protocol” str

Flags : Read / Write / Construct Only

#### ready-state

    “ready-state” GstWebRTCDataChannelState *

Flags : Read

#### ready-state

    “ready-state” GstWebRTC.WebRTCDataChannelState

Flags : Read

#### ready_state

    “self.props.ready_state” GstWebRTC.WebRTCDataChannelState

Flags : Read

## GstWebRTCICETransport

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstWebRTCICETransport

### Class structure

#### GstWebRTCICETransportClass

### GstWebRTC.WebRTCICETransportClass

## GstWebRTC.WebRTCICETransport

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCICETransport

## GstWebRTC.WebRTCICETransport

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCICETransport

### Signals

#### on-new-candidate

    on_new_candidate_callback (GstWebRTCICETransport * self,
                               gchar * object,
                               gpointer user_data)

Parameters:

`self` –
No description available
`object` –
No description available
`user_data` –
No description available

Flags: Run Last

#### on-new-candidate

    function on_new_candidate_callback(self: GstWebRTC.WebRTCICETransport, object: String, user_data: Object): {
        // javascript callback for the 'on-new-candidate' signal
    }

Parameters:

`self` (GstWebRTC.WebRTCICETransport) –
No description available
`object` (String) –
No description available
`user_data` (Object) –
No description available

Flags: Run Last

#### on-new-candidate

    def on_new_candidate_callback (self, object, *user_data):
        #python callback for the 'on-new-candidate' signal

Parameters:

`self` (GstWebRTC.WebRTCICETransport) –
No description available
`object` (str) –
No description available
`user_data` (variadic) –
No description available

Flags: Run Last

#### on-selected-candidate-pair-change

    on_selected_candidate_pair_change_callback (GstWebRTCICETransport * self,
                                                gpointer user_data)

Parameters:

`self` –
No description available
`user_data` –
No description available

Flags: Run Last

#### on-selected-candidate-pair-change

    function on_selected_candidate_pair_change_callback(self: GstWebRTC.WebRTCICETransport, user_data: Object): {
        // javascript callback for the 'on-selected-candidate-pair-change' signal
    }

Parameters:

`self` (GstWebRTC.WebRTCICETransport) –
No description available
`user_data` (Object) –
No description available

Flags: Run Last

#### on-selected-candidate-pair-change

    def on_selected_candidate_pair_change_callback (self, *user_data):
        #python callback for the 'on-selected-candidate-pair-change' signal

Parameters:

`self` (GstWebRTC.WebRTCICETransport) –
No description available
`user_data` (variadic) –
No description available

Flags: Run Last

### Properties

#### component

    “component” GstWebRTCICEComponent *

Flags : Read / Write / Construct Only

#### component

    “component” GstWebRTC.WebRTCICEComponent

Flags : Read / Write / Construct Only

#### component

    “self.props.component” GstWebRTC.WebRTCICEComponent

Flags : Read / Write / Construct Only

#### gathering-state

    “gathering-state” GstWebRTCICEGatheringState *

Flags : Read

#### gathering-state

    “gathering-state” GstWebRTC.WebRTCICEGatheringState

Flags : Read

#### gathering_state

    “self.props.gathering_state” GstWebRTC.WebRTCICEGatheringState

Flags : Read

#### state

    “state” GstWebRTCICEConnectionState *

Flags : Read

#### state

    “state” GstWebRTC.WebRTCICEConnectionState

Flags : Read

#### state

    “self.props.state” GstWebRTC.WebRTCICEConnectionState

Flags : Read

## GstWebRTCRTPReceiver

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstWebRTCRTPReceiver

An object to track the receiving aspect of the stream

Mostly matches the WebRTC RTCRtpReceiver interface.

### Class structure

#### GstWebRTCRTPReceiverClass

### GstWebRTC.WebRTCRTPReceiverClass

## GstWebRTC.WebRTCRTPReceiver

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCRTPReceiver

An object to track the receiving aspect of the stream

Mostly matches the WebRTC RTCRtpReceiver interface.

## GstWebRTC.WebRTCRTPReceiver

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCRTPReceiver

An object to track the receiving aspect of the stream

Mostly matches the WebRTC RTCRtpReceiver interface.

### Properties

#### transport

    “transport” GstWebRTCDTLSTransport *

The DTLS transport for this receiver

Flags : Read

#### transport

    “transport” GstWebRTC.WebRTCDTLSTransport

The DTLS transport for this receiver

Flags : Read

#### transport

    “self.props.transport” GstWebRTC.WebRTCDTLSTransport

The DTLS transport for this receiver

Flags : Read

## GstWebRTCRTPSender

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstWebRTCRTPSender

An object to track the sending aspect of the stream

Mostly matches the WebRTC RTCRtpSender interface.

### Class structure

#### GstWebRTCRTPSenderClass

### GstWebRTC.WebRTCRTPSenderClass

## GstWebRTC.WebRTCRTPSender

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCRTPSender

An object to track the sending aspect of the stream

Mostly matches the WebRTC RTCRtpSender interface.

## GstWebRTC.WebRTCRTPSender

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCRTPSender

An object to track the sending aspect of the stream

Mostly matches the WebRTC RTCRtpSender interface.

### Methods

#### gst_webrtc_rtp_sender_set_priority

    gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender * sender,
                                        GstWebRTCPriorityType priority)

Sets the content of the IPv4 Type of Service (ToS), also known as DSCP (Differentiated Services Code Point). This also sets the Traffic Class field of IPv6.

Parameters:

`sender` –

a GstWebRTCRTPSender

`priority` –

The priority of this sender

Since : 1.20

#### GstWebRTC.WebRTCRTPSender.prototype.set_priority

    function GstWebRTC.WebRTCRTPSender.prototype.set_priority(priority: GstWebRTC.WebRTCPriorityType): {
        // javascript wrapper for 'gst_webrtc_rtp_sender_set_priority'
    }

Sets the content of the IPv4 Type of Service (ToS), also known as DSCP (Differentiated Services Code Point). This also sets the Traffic Class field of IPv6.

Parameters:

`sender` (GstWebRTC.WebRTCRTPSender) –

a GstWebRTC.WebRTCRTPSender

`priority` (GstWebRTC.WebRTCPriorityType) –

The priority of this sender

Since : 1.20

#### GstWebRTC.WebRTCRTPSender.set_priority

    def GstWebRTC.WebRTCRTPSender.set_priority (self, priority):
        #python wrapper for 'gst_webrtc_rtp_sender_set_priority'

Sets the content of the IPv4 Type of Service (ToS), also known as DSCP (Differentiated Services Code Point). This also sets the Traffic Class field of IPv6.

Parameters:

`sender` (GstWebRTC.WebRTCRTPSender) –

a GstWebRTC.WebRTCRTPSender

`priority` (GstWebRTC.WebRTCPriorityType) –

The priority of this sender

Since : 1.20

### Properties

#### priority

    “priority” GstWebRTCPriorityType *

The priority from which to set the DSCP field on packets

Flags : Read / Write

#### priority

    “priority” GstWebRTC.WebRTCPriorityType

The priority from which to set the DSCP field on packets

Flags : Read / Write

#### priority

    “self.props.priority” GstWebRTC.WebRTCPriorityType

The priority from which to set the DSCP field on packets

Flags : Read / Write

#### transport

    “transport” GstWebRTCDTLSTransport *

The DTLS transport for this sender

Flags : Read

#### transport

    “transport” GstWebRTC.WebRTCDTLSTransport

The DTLS transport for this sender

Flags : Read

#### transport

    “self.props.transport” GstWebRTC.WebRTCDTLSTransport

The DTLS transport for this sender

Flags : Read

## GstWebRTCRTPTransceiver

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstWebRTCRTPTransceiver

Mostly matches the WebRTC RTCRtpTransceiver interface.

### Class structure

#### GstWebRTCRTPTransceiverClass

### GstWebRTC.WebRTCRTPTransceiverClass

## GstWebRTC.WebRTCRTPTransceiver

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCRTPTransceiver

Mostly matches the WebRTC RTCRtpTransceiver interface.

## GstWebRTC.WebRTCRTPTransceiver

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstWebRTC.WebRTCRTPTransceiver

Mostly matches the WebRTC RTCRtpTransceiver interface.

### Properties

#### codec-preferences

    “codec-preferences” GstCaps *

Caps representing the codec preferences.

Flags : Read / Write

#### codec-preferences

    “codec-preferences” Gst.Caps

Caps representing the codec preferences.

Flags : Read / Write

#### codec_preferences

    “self.props.codec_preferences” Gst.Caps

Caps representing the codec preferences.

Flags : Read / Write

#### current-direction

    “current-direction” GstWebRTCRTPTransceiverDirection *

The transceiver's current directionality, or none if the transceiver is stopped or has never participated in an exchange of offers and answers. To change the transceiver's directionality, set the value of the direction property.

Flags : Read

#### current-direction

    “current-direction” GstWebRTC.WebRTCRTPTransceiverDirection

The transceiver's current directionality, or none if the transceiver is stopped or has never participated in an exchange of offers and answers. To change the transceiver's directionality, set the value of the direction property.

Flags : Read

#### current_direction

    “self.props.current_direction” GstWebRTC.WebRTCRTPTransceiverDirection

The transceiver's current directionality, or none if the transceiver is stopped or has never participated in an exchange of offers and answers. To change the transceiver's directionality, set the value of the direction property.

Flags : Read

#### direction

    “direction” GstWebRTCRTPTransceiverDirection *

Direction of the transceiver.

Flags : Read / Write

#### direction

    “direction” GstWebRTC.WebRTCRTPTransceiverDirection

Direction of the transceiver.

Flags : Read / Write

#### direction

    “self.props.direction” GstWebRTC.WebRTCRTPTransceiverDirection

Direction of the transceiver.

Flags : Read / Write

#### kind

    “kind” GstWebRTCKind *

The kind of media this transceiver transports

Flags : Read

#### kind

    “kind” GstWebRTC.WebRTCKind

The kind of media this transceiver transports

Flags : Read

#### kind

    “self.props.kind” GstWebRTC.WebRTCKind

The kind of media this transceiver transports

Flags : Read

#### mid

    “mid” gchar *

The media ID of the m-line associated with this transceiver. This association is established, when possible, whenever either a local or remote description is applied. This field is null if neither a local or remote description has been applied, or if its associated m-line is rejected by either a remote offer or any answer.

Flags : Read

#### mid

    “mid” String

The media ID of the m-line associated with this transceiver. This association is established, when possible, whenever either a local or remote description is applied. This field is null if neither a local or remote description has been applied, or if its associated m-line is rejected by either a remote offer or any answer.

Flags : Read

#### mid

    “self.props.mid” str

The media ID of the m-line associated with this transceiver. This association is established, when possible, whenever either a local or remote description is applied. This field is null if neither a local or remote description has been applied, or if its associated m-line is rejected by either a remote offer or any answer.

Flags : Read

#### mlineindex

    “mlineindex” guint

Flags : Read / Write / Construct Only

#### mlineindex

    “mlineindex” Number

Flags : Read / Write / Construct Only

#### mlineindex

    “self.props.mlineindex” int

Flags : Read / Write / Construct Only

#### receiver

    “receiver” GstWebRTCRTPReceiver *

Flags : Read / Write / Construct Only

#### receiver

    “receiver” GstWebRTC.WebRTCRTPReceiver

Flags : Read / Write / Construct Only

#### receiver

    “self.props.receiver” GstWebRTC.WebRTCRTPReceiver

Flags : Read / Write / Construct Only

#### sender

    “sender” GstWebRTCRTPSender *

Flags : Read / Write / Construct Only

#### sender

    “sender” GstWebRTC.WebRTCRTPSender

Flags : Read / Write / Construct Only

#### sender

    “self.props.sender” GstWebRTC.WebRTCRTPSender

Flags : Read / Write / Construct Only

## Enumerations

### GstWebRTCBundlePolicy

See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.

##### Members

`GST_WEBRTC_BUNDLE_POLICY_NONE` (0) –

none

`GST_WEBRTC_BUNDLE_POLICY_BALANCED` (1) –

balanced

`GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT` (2) –

max-compat

`GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE` (3) –

max-bundle

### GstWebRTC.WebRTCBundlePolicy

See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.

##### Members

`GstWebRTC.WebRTCBundlePolicy.NONE` (0) –

none

`GstWebRTC.WebRTCBundlePolicy.BALANCED` (1) –

balanced

`GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT` (2) –

max-compat

`GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE` (3) –

max-bundle

### GstWebRTC.WebRTCBundlePolicy

See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.

##### Members

`GstWebRTC.WebRTCBundlePolicy.NONE` (0) –

none

`GstWebRTC.WebRTCBundlePolicy.BALANCED` (1) –

balanced

`GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT` (2) –

max-compat

`GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE` (3) –

max-bundle

### GstWebRTCDTLSSetup

##### Members

`GST_WEBRTC_DTLS_SETUP_NONE` (0) –

none

`GST_WEBRTC_DTLS_SETUP_ACTPASS` (1) –

actpass

`GST_WEBRTC_DTLS_SETUP_ACTIVE` (2) –

sendonly

`GST_WEBRTC_DTLS_SETUP_PASSIVE` (3) –

recvonly

### GstWebRTC.WebRTCDTLSSetup

##### Members

`GstWebRTC.WebRTCDTLSSetup.NONE` (0) –

none

`GstWebRTC.WebRTCDTLSSetup.ACTPASS` (1) –

actpass

`GstWebRTC.WebRTCDTLSSetup.ACTIVE` (2) –

sendonly

`GstWebRTC.WebRTCDTLSSetup.PASSIVE` (3) –

recvonly

### GstWebRTC.WebRTCDTLSSetup

##### Members

`GstWebRTC.WebRTCDTLSSetup.NONE` (0) –

none

`GstWebRTC.WebRTCDTLSSetup.ACTPASS` (1) –

actpass

`GstWebRTC.WebRTCDTLSSetup.ACTIVE` (2) –

sendonly

`GstWebRTC.WebRTCDTLSSetup.PASSIVE` (3) –

recvonly

### GstWebRTCDTLSTransportState

##### Members

`GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW` (0) –

new

`GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED` (1) –

closed

`GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED` (2) –

failed

`GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING` (3) –

connecting

`GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED` (4) –

connected

### GstWebRTC.WebRTCDTLSTransportState

##### Members

`GstWebRTC.WebRTCDTLSTransportState.NEW` (0) –

new

`GstWebRTC.WebRTCDTLSTransportState.CLOSED` (1) –

closed

`GstWebRTC.WebRTCDTLSTransportState.FAILED` (2) –

failed

`GstWebRTC.WebRTCDTLSTransportState.CONNECTING` (3) –

connecting

`GstWebRTC.WebRTCDTLSTransportState.CONNECTED` (4) –

connected

### GstWebRTC.WebRTCDTLSTransportState

##### Members

`GstWebRTC.WebRTCDTLSTransportState.NEW` (0) –

new

`GstWebRTC.WebRTCDTLSTransportState.CLOSED` (1) –

closed

`GstWebRTC.WebRTCDTLSTransportState.FAILED` (2) –

failed

`GstWebRTC.WebRTCDTLSTransportState.CONNECTING` (3) –

connecting

`GstWebRTC.WebRTCDTLSTransportState.CONNECTED` (4) –

connected

### GstWebRTCDataChannelState

See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate

##### Members

`GST_WEBRTC_DATA_CHANNEL_STATE_NEW` (0) –

new

`GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING` (1) –

connection

`GST_WEBRTC_DATA_CHANNEL_STATE_OPEN` (2) –

open

`GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING` (3) –

closing

`GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED` (4) –

closed

### GstWebRTC.WebRTCDataChannelState

See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate

##### Members

`GstWebRTC.WebRTCDataChannelState.NEW` (0) –

new

`GstWebRTC.WebRTCDataChannelState.CONNECTING` (1) –

connection

`GstWebRTC.WebRTCDataChannelState.OPEN` (2) –

open

`GstWebRTC.WebRTCDataChannelState.CLOSING` (3) –

closing

`GstWebRTC.WebRTCDataChannelState.CLOSED` (4) –

closed

### GstWebRTC.WebRTCDataChannelState

See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate

##### Members

`GstWebRTC.WebRTCDataChannelState.NEW` (0) –

new

`GstWebRTC.WebRTCDataChannelState.CONNECTING` (1) –

connection

`GstWebRTC.WebRTCDataChannelState.OPEN` (2) –

open

`GstWebRTC.WebRTCDataChannelState.CLOSING` (3) –

closing

`GstWebRTC.WebRTCDataChannelState.CLOSED` (4) –

closed

### GstWebRTCError

See https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype for more information.

##### Members

`GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE` (0) –

data-channel-failure

`GST_WEBRTC_ERROR_DTLS_FAILURE` (1) –

dtls-failure

`GST_WEBRTC_ERROR_FINGERPRINT_FAILURE` (2) –

fingerprint-failure

`GST_WEBRTC_ERROR_SCTP_FAILURE` (3) –

sctp-failure

`GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR` (4) –

sdp-syntax-error

`GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE` (5) –

hardware-encoder-not-available

`GST_WEBRTC_ERROR_ENCODER_ERROR` (6) –

encoder-error

`GST_WEBRTC_ERROR_INVALID_STATE` (7) –

invalid-state (part of WebIDL specification)

`GST_WEBRTC_ERROR_INTERNAL_FAILURE` (8) –

GStreamer-specific failure, not matching any other value from the specification

### GstWebRTC.WebRTCError

See https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype for more information.

##### Members

`GstWebRTC.WebRTCError.DATA_CHANNEL_FAILURE` (0) –

data-channel-failure

`GstWebRTC.WebRTCError.DTLS_FAILURE` (1) –

dtls-failure

`GstWebRTC.WebRTCError.FINGERPRINT_FAILURE` (2) –

fingerprint-failure

`GstWebRTC.WebRTCError.SCTP_FAILURE` (3) –

sctp-failure

`GstWebRTC.WebRTCError.SDP_SYNTAX_ERROR` (4) –

sdp-syntax-error

`GstWebRTC.WebRTCError.HARDWARE_ENCODER_NOT_AVAILABLE` (5) –

hardware-encoder-not-available

`GstWebRTC.WebRTCError.ENCODER_ERROR` (6) –

encoder-error

`GstWebRTC.WebRTCError.INVALID_STATE` (7) –

invalid-state (part of WebIDL specification)

`GstWebRTC.WebRTCError.INTERNAL_FAILURE` (8) –

GStreamer-specific failure, not matching any other value from the specification

### GstWebRTC.WebRTCError

See https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype for more information.

##### Members

`GstWebRTC.WebRTCError.DATA_CHANNEL_FAILURE` (0) –

data-channel-failure

`GstWebRTC.WebRTCError.DTLS_FAILURE` (1) –

dtls-failure

`GstWebRTC.WebRTCError.FINGERPRINT_FAILURE` (2) –

fingerprint-failure

`GstWebRTC.WebRTCError.SCTP_FAILURE` (3) –

sctp-failure

`GstWebRTC.WebRTCError.SDP_SYNTAX_ERROR` (4) –

sdp-syntax-error

`GstWebRTC.WebRTCError.HARDWARE_ENCODER_NOT_AVAILABLE` (5) –

hardware-encoder-not-available

`GstWebRTC.WebRTCError.ENCODER_ERROR` (6) –

encoder-error

`GstWebRTC.WebRTCError.INVALID_STATE` (7) –

invalid-state (part of WebIDL specification)

`GstWebRTC.WebRTCError.INTERNAL_FAILURE` (8) –

GStreamer-specific failure, not matching any other value from the specification

### GstWebRTCFECType

##### Members

`GST_WEBRTC_FEC_TYPE_NONE` (0) –

none

`GST_WEBRTC_FEC_TYPE_ULP_RED` (1) –

ulpfec + red

### GstWebRTC.WebRTCFECType

##### Members

`GstWebRTC.WebRTCFECType.NONE` (0) –

none

`GstWebRTC.WebRTCFECType.ULP_RED` (1) –

ulpfec + red

### GstWebRTC.WebRTCFECType

##### Members

`GstWebRTC.WebRTCFECType.NONE` (0) –

none

`GstWebRTC.WebRTCFECType.ULP_RED` (1) –

ulpfec + red

### GstWebRTCICEComponent

##### Members

`GST_WEBRTC_ICE_COMPONENT_RTP` (0) –

RTP component

`GST_WEBRTC_ICE_COMPONENT_RTCP` (1) –

RTCP component

### GstWebRTC.WebRTCICEComponent

##### Members

`GstWebRTC.WebRTCICEComponent.RTP` (0) –

RTP component

`GstWebRTC.WebRTCICEComponent.RTCP` (1) –

RTCP component

### GstWebRTC.WebRTCICEComponent

##### Members

`GstWebRTC.WebRTCICEComponent.RTP` (0) –

RTP component

`GstWebRTC.WebRTCICEComponent.RTCP` (1) –

RTCP component

### GstWebRTCICEConnectionState

See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate

##### Members

`GST_WEBRTC_ICE_CONNECTION_STATE_NEW` (0) –

new

`GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING` (1) –

checking

`GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED` (2) –

connected

`GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED` (3) –

completed

`GST_WEBRTC_ICE_CONNECTION_STATE_FAILED` (4) –

failed

`GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED` (5) –

disconnected

`GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED` (6) –

closed

### GstWebRTC.WebRTCICEConnectionState

See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate

##### Members

`GstWebRTC.WebRTCICEConnectionState.NEW` (0) –

new

`GstWebRTC.WebRTCICEConnectionState.CHECKING` (1) –

checking

`GstWebRTC.WebRTCICEConnectionState.CONNECTED` (2) –

connected

`GstWebRTC.WebRTCICEConnectionState.COMPLETED` (3) –

completed

`GstWebRTC.WebRTCICEConnectionState.FAILED` (4) –

failed

`GstWebRTC.WebRTCICEConnectionState.DISCONNECTED` (5) –

disconnected

`GstWebRTC.WebRTCICEConnectionState.CLOSED` (6) –

closed

### GstWebRTC.WebRTCICEConnectionState

See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate

##### Members

`GstWebRTC.WebRTCICEConnectionState.NEW` (0) –

new

`GstWebRTC.WebRTCICEConnectionState.CHECKING` (1) –

checking

`GstWebRTC.WebRTCICEConnectionState.CONNECTED` (2) –

connected

`GstWebRTC.WebRTCICEConnectionState.COMPLETED` (3) –

completed

`GstWebRTC.WebRTCICEConnectionState.FAILED` (4) –

failed

`GstWebRTC.WebRTCICEConnectionState.DISCONNECTED` (5) –

disconnected

`GstWebRTC.WebRTCICEConnectionState.CLOSED` (6) –

closed

### GstWebRTCICEGatheringState

See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate

##### Members

`GST_WEBRTC_ICE_GATHERING_STATE_NEW` (0) –

new

`GST_WEBRTC_ICE_GATHERING_STATE_GATHERING` (1) –

gathering

`GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE` (2) –

complete

### GstWebRTC.WebRTCICEGatheringState

See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate

##### Members

`GstWebRTC.WebRTCICEGatheringState.NEW` (0) –

new

`GstWebRTC.WebRTCICEGatheringState.GATHERING` (1) –

gathering

`GstWebRTC.WebRTCICEGatheringState.COMPLETE` (2) –

complete

### GstWebRTC.WebRTCICEGatheringState

See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate

##### Members

`GstWebRTC.WebRTCICEGatheringState.NEW` (0) –

new

`GstWebRTC.WebRTCICEGatheringState.GATHERING` (1) –

gathering

`GstWebRTC.WebRTCICEGatheringState.COMPLETE` (2) –

complete

### GstWebRTCICERole

##### Members

`GST_WEBRTC_ICE_ROLE_CONTROLLED` (0) –

controlled

`GST_WEBRTC_ICE_ROLE_CONTROLLING` (1) –

controlling

### GstWebRTC.WebRTCICERole

##### Members

`GstWebRTC.WebRTCICERole.CONTROLLED` (0) –

controlled

`GstWebRTC.WebRTCICERole.CONTROLLING` (1) –

controlling

### GstWebRTC.WebRTCICERole

##### Members

`GstWebRTC.WebRTCICERole.CONTROLLED` (0) –

controlled

`GstWebRTC.WebRTCICERole.CONTROLLING` (1) –

controlling

### GstWebRTCICETransportPolicy

See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.

##### Members

`GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL` (0) –

all

`GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY` (1) –

relay

### GstWebRTC.WebRTCICETransportPolicy

See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.

##### Members

`GstWebRTC.WebRTCICETransportPolicy.ALL` (0) –

all

`GstWebRTC.WebRTCICETransportPolicy.RELAY` (1) –

relay

### GstWebRTC.WebRTCICETransportPolicy

See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.

##### Members

`GstWebRTC.WebRTCICETransportPolicy.ALL` (0) –

all

`GstWebRTC.WebRTCICETransportPolicy.RELAY` (1) –

relay

### GstWebRTCKind

https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind

##### Members

`GST_WEBRTC_KIND_UNKNOWN` (0) –

Kind has not yet been set

`GST_WEBRTC_KIND_AUDIO` (1) –

Kind is audio

`GST_WEBRTC_KIND_VIDEO` (2) –

Kind is audio

### GstWebRTC.WebRTCKind

https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind

##### Members

`GstWebRTC.WebRTCKind.UNKNOWN` (0) –

Kind has not yet been set

`GstWebRTC.WebRTCKind.AUDIO` (1) –

Kind is audio

`GstWebRTC.WebRTCKind.VIDEO` (2) –

Kind is audio

### GstWebRTC.WebRTCKind

https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind

##### Members

`GstWebRTC.WebRTCKind.UNKNOWN` (0) –

Kind has not yet been set

`GstWebRTC.WebRTCKind.AUDIO` (1) –

Kind is audio

`GstWebRTC.WebRTCKind.VIDEO` (2) –

Kind is audio

### GstWebRTCPeerConnectionState

See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate

##### Members

`GST_WEBRTC_PEER_CONNECTION_STATE_NEW` (0) –

new

`GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING` (1) –

connecting

`GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED` (2) –

connected

`GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED` (3) –

disconnected

`GST_WEBRTC_PEER_CONNECTION_STATE_FAILED` (4) –

failed

`GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED` (5) –

closed

### GstWebRTC.WebRTCPeerConnectionState

See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate

##### Members

`GstWebRTC.WebRTCPeerConnectionState.NEW` (0) –

new

`GstWebRTC.WebRTCPeerConnectionState.CONNECTING` (1) –

connecting

`GstWebRTC.WebRTCPeerConnectionState.CONNECTED` (2) –

connected

`GstWebRTC.WebRTCPeerConnectionState.DISCONNECTED` (3) –

disconnected

`GstWebRTC.WebRTCPeerConnectionState.FAILED` (4) –

failed

`GstWebRTC.WebRTCPeerConnectionState.CLOSED` (5) –

closed

### GstWebRTC.WebRTCPeerConnectionState

See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate

##### Members

`GstWebRTC.WebRTCPeerConnectionState.NEW` (0) –

new

`GstWebRTC.WebRTCPeerConnectionState.CONNECTING` (1) –

connecting

`GstWebRTC.WebRTCPeerConnectionState.CONNECTED` (2) –

connected

`GstWebRTC.WebRTCPeerConnectionState.DISCONNECTED` (3) –

disconnected

`GstWebRTC.WebRTCPeerConnectionState.FAILED` (4) –

failed

`GstWebRTC.WebRTCPeerConnectionState.CLOSED` (5) –

closed

### GstWebRTCPriorityType

See http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype

##### Members

`GST_WEBRTC_PRIORITY_TYPE_VERY_LOW` (1) –

very-low

`GST_WEBRTC_PRIORITY_TYPE_LOW` (2) –

low

`GST_WEBRTC_PRIORITY_TYPE_MEDIUM` (3) –

medium

`GST_WEBRTC_PRIORITY_TYPE_HIGH` (4) –

high

### GstWebRTC.WebRTCPriorityType

See http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype

##### Members

`GstWebRTC.WebRTCPriorityType.VERY_LOW` (1) –

very-low

`GstWebRTC.WebRTCPriorityType.LOW` (2) –

low

`GstWebRTC.WebRTCPriorityType.MEDIUM` (3) –

medium

`GstWebRTC.WebRTCPriorityType.HIGH` (4) –

high

### GstWebRTC.WebRTCPriorityType

See http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype

##### Members

`GstWebRTC.WebRTCPriorityType.VERY_LOW` (1) –

very-low

`GstWebRTC.WebRTCPriorityType.LOW` (2) –

low

`GstWebRTC.WebRTCPriorityType.MEDIUM` (3) –

medium

`GstWebRTC.WebRTCPriorityType.HIGH` (4) –

high

### GstWebRTCRTPTransceiverDirection

##### Members

`GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE` (0) –

none

`GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE` (1) –

inactive

`GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY` (2) –

sendonly

`GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY` (3) –

recvonly

`GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV` (4) –

sendrecv

### GstWebRTC.WebRTCRTPTransceiverDirection

##### Members

`GstWebRTC.WebRTCRTPTransceiverDirection.NONE` (0) –

none

`GstWebRTC.WebRTCRTPTransceiverDirection.INACTIVE` (1) –

inactive

`GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY` (2) –

sendonly

`GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY` (3) –

recvonly

`GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV` (4) –

sendrecv

### GstWebRTC.WebRTCRTPTransceiverDirection

##### Members

`GstWebRTC.WebRTCRTPTransceiverDirection.NONE` (0) –

none

`GstWebRTC.WebRTCRTPTransceiverDirection.INACTIVE` (1) –

inactive

`GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY` (2) –

sendonly

`GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY` (3) –

recvonly

`GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV` (4) –

sendrecv

### GstWebRTCSCTPTransportState

See http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate

##### Members

`GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW` (0) –

new

`GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING` (1) –

connecting

`GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED` (2) –

connected

`GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED` (3) –

closed

### GstWebRTC.WebRTCSCTPTransportState

See http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate

##### Members

`GstWebRTC.WebRTCSCTPTransportState.NEW` (0) –

new

`GstWebRTC.WebRTCSCTPTransportState.CONNECTING` (1) –

connecting

`GstWebRTC.WebRTCSCTPTransportState.CONNECTED` (2) –

connected

`GstWebRTC.WebRTCSCTPTransportState.CLOSED` (3) –

closed

### GstWebRTC.WebRTCSCTPTransportState

See http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate

##### Members

`GstWebRTC.WebRTCSCTPTransportState.NEW` (0) –

new

`GstWebRTC.WebRTCSCTPTransportState.CONNECTING` (1) –

connecting

`GstWebRTC.WebRTCSCTPTransportState.CONNECTED` (2) –

connected

`GstWebRTC.WebRTCSCTPTransportState.CLOSED` (3) –

closed

### GstWebRTCSDPType

See http://w3c.github.io/webrtc-pc/#rtcsdptype

##### Members

`GST_WEBRTC_SDP_TYPE_OFFER` (1) –

offer

`GST_WEBRTC_SDP_TYPE_PRANSWER` (2) –

pranswer

`GST_WEBRTC_SDP_TYPE_ANSWER` (3) –

answer

`GST_WEBRTC_SDP_TYPE_ROLLBACK` (4) –

rollback

### GstWebRTC.WebRTCSDPType

See http://w3c.github.io/webrtc-pc/#rtcsdptype

##### Members

`GstWebRTC.WebRTCSDPType.OFFER` (1) –

offer

`GstWebRTC.WebRTCSDPType.PRANSWER` (2) –

pranswer

`GstWebRTC.WebRTCSDPType.ANSWER` (3) –

answer

`GstWebRTC.WebRTCSDPType.ROLLBACK` (4) –

rollback

### GstWebRTC.WebRTCSDPType

See http://w3c.github.io/webrtc-pc/#rtcsdptype

##### Members

`GstWebRTC.WebRTCSDPType.OFFER` (1) –

offer

`GstWebRTC.WebRTCSDPType.PRANSWER` (2) –

pranswer

`GstWebRTC.WebRTCSDPType.ANSWER` (3) –

answer

`GstWebRTC.WebRTCSDPType.ROLLBACK` (4) –

rollback

### GstWebRTCSignalingState

See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate

##### Members

`GST_WEBRTC_SIGNALING_STATE_STABLE` (0) –

stable

`GST_WEBRTC_SIGNALING_STATE_CLOSED` (1) –

closed

`GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER` (2) –

have-local-offer

`GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER` (3) –

have-remote-offer

`GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER` (4) –

have-local-pranswer

`GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER` (5) –

have-remote-pranswer

### GstWebRTC.WebRTCSignalingState

See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate

##### Members

`GstWebRTC.WebRTCSignalingState.STABLE` (0) –

stable

`GstWebRTC.WebRTCSignalingState.CLOSED` (1) –

closed

`GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_OFFER` (2) –

have-local-offer

`GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_OFFER` (3) –

have-remote-offer

`GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_PRANSWER` (4) –

have-local-pranswer

`GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_PRANSWER` (5) –

have-remote-pranswer

### GstWebRTC.WebRTCSignalingState

See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate

##### Members

`GstWebRTC.WebRTCSignalingState.STABLE` (0) –

stable

`GstWebRTC.WebRTCSignalingState.CLOSED` (1) –

closed

`GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_OFFER` (2) –

have-local-offer

`GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_OFFER` (3) –

have-remote-offer

`GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_PRANSWER` (4) –

have-local-pranswer

`GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_PRANSWER` (5) –

have-remote-pranswer

### GstWebRTCStatsType

##### Members

`GST_WEBRTC_STATS_CODEC` (1) –

codec

`GST_WEBRTC_STATS_INBOUND_RTP` (2) –

inbound-rtp

`GST_WEBRTC_STATS_OUTBOUND_RTP` (3) –

outbound-rtp

`GST_WEBRTC_STATS_REMOTE_INBOUND_RTP` (4) –

remote-inbound-rtp

`GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP` (5) –

remote-outbound-rtp

`GST_WEBRTC_STATS_CSRC` (6) –

csrc

`GST_WEBRTC_STATS_PEER_CONNECTION` (7) –

peer-connectiion

`GST_WEBRTC_STATS_DATA_CHANNEL` (8) –

data-channel

`GST_WEBRTC_STATS_STREAM` (9) –

stream

`GST_WEBRTC_STATS_TRANSPORT` (10) –

transport

`GST_WEBRTC_STATS_CANDIDATE_PAIR` (11) –

candidate-pair

`GST_WEBRTC_STATS_LOCAL_CANDIDATE` (12) –

local-candidate

`GST_WEBRTC_STATS_REMOTE_CANDIDATE` (13) –

remote-candidate

`GST_WEBRTC_STATS_CERTIFICATE` (14) –

certificate

### GstWebRTC.WebRTCStatsType

##### Members

`GstWebRTC.WebRTCStatsType.CODEC` (1) –

codec

`GstWebRTC.WebRTCStatsType.INBOUND_RTP` (2) –

inbound-rtp

`GstWebRTC.WebRTCStatsType.OUTBOUND_RTP` (3) –

outbound-rtp

`GstWebRTC.WebRTCStatsType.REMOTE_INBOUND_RTP` (4) –

remote-inbound-rtp

`GstWebRTC.WebRTCStatsType.REMOTE_OUTBOUND_RTP` (5) –

remote-outbound-rtp

`GstWebRTC.WebRTCStatsType.CSRC` (6) –

csrc

`GstWebRTC.WebRTCStatsType.PEER_CONNECTION` (7) –

peer-connectiion

`GstWebRTC.WebRTCStatsType.DATA_CHANNEL` (8) –

data-channel

`GstWebRTC.WebRTCStatsType.STREAM` (9) –

stream

`GstWebRTC.WebRTCStatsType.TRANSPORT` (10) –

transport

`GstWebRTC.WebRTCStatsType.CANDIDATE_PAIR` (11) –

candidate-pair

`GstWebRTC.WebRTCStatsType.LOCAL_CANDIDATE` (12) –

local-candidate

`GstWebRTC.WebRTCStatsType.REMOTE_CANDIDATE` (13) –

remote-candidate

`GstWebRTC.WebRTCStatsType.CERTIFICATE` (14) –

certificate

### GstWebRTC.WebRTCStatsType

##### Members

`GstWebRTC.WebRTCStatsType.CODEC` (1) –

codec

`GstWebRTC.WebRTCStatsType.INBOUND_RTP` (2) –

inbound-rtp

`GstWebRTC.WebRTCStatsType.OUTBOUND_RTP` (3) –

outbound-rtp

`GstWebRTC.WebRTCStatsType.REMOTE_INBOUND_RTP` (4) –

remote-inbound-rtp

`GstWebRTC.WebRTCStatsType.REMOTE_OUTBOUND_RTP` (5) –

remote-outbound-rtp

`GstWebRTC.WebRTCStatsType.CSRC` (6) –

csrc

`GstWebRTC.WebRTCStatsType.PEER_CONNECTION` (7) –

peer-connectiion

`GstWebRTC.WebRTCStatsType.DATA_CHANNEL` (8) –

data-channel

`GstWebRTC.WebRTCStatsType.STREAM` (9) –

stream

`GstWebRTC.WebRTCStatsType.TRANSPORT` (10) –

transport

`GstWebRTC.WebRTCStatsType.CANDIDATE_PAIR` (11) –

candidate-pair

`GstWebRTC.WebRTCStatsType.LOCAL_CANDIDATE` (12) –

local-candidate

`GstWebRTC.WebRTCStatsType.REMOTE_CANDIDATE` (13) –

remote-candidate

`GstWebRTC.WebRTCStatsType.CERTIFICATE` (14) –

certificate

## Constants

### GST_WEBRTC_API

    #define GST_WEBRTC_API GST_API_EXPORT         /* from config.h */

### GST_WEBRTC_ERROR

    #define GST_WEBRTC_ERROR gst_webrtc_error_quark ()

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