GstWebRTC Enumerations

GstWebRTCDTLSTransport

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

Class structure

GstWebRTCDTLSTransportClass


GstWebRTC.WebRTCDTLSTransportClass


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.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.


GstWebRTC.WebRTCDataChannel.close

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

Close the channel.


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.


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.


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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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:

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.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:

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:

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:

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:

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.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.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:

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:

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.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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