GstWebRTC Enumerations
GstWebRTCICECandidateStats
Members
ipaddr
		(gchar *)
		–
	port
		(guint)
		–
	stream_id
		(guint)
		–
	type
		(const gchar *)
		–
	proto
		(const gchar *)
		–
	relay_proto
		(const gchar *)
		–
	prio
		(guint)
		–
	url
		(gchar *)
		–
	_gst_reserved
		(gpointer *)
		–
	Since : 1.22
GstWebRTC.WebRTCICECandidateStats
Members
ipaddr
		(String)
		–
	port
		(Number)
		–
	stream_id
		(Number)
		–
	type
		(String)
		–
	proto
		(String)
		–
	relay_proto
		(String)
		–
	prio
		(Number)
		–
	url
		(String)
		–
	_gst_reserved
		([ Object ])
		–
	Since : 1.22
GstWebRTC.WebRTCICECandidateStats
Members
ipaddr
		(str)
		–
	port
		(int)
		–
	stream_id
		(int)
		–
	type
		(str)
		–
	proto
		(str)
		–
	relay_proto
		(str)
		–
	prio
		(int)
		–
	url
		(str)
		–
	_gst_reserved
		([ object ])
		–
	Since : 1.22
Methods
gst_webrtc_ice_candidate_stats_copy
GstWebRTCICECandidateStats * gst_webrtc_ice_candidate_stats_copy (GstWebRTCICECandidateStats * stats)
Parameters:
stats
–
The GstWebRTCICE
A copy of stats
Since : 1.22
GstWebRTC.WebRTCICECandidateStats.prototype.copy
function GstWebRTC.WebRTCICECandidateStats.prototype.copy(): {
    // javascript wrapper for 'gst_webrtc_ice_candidate_stats_copy'
}
	Parameters:
A copy of stats
Since : 1.22
GstWebRTC.WebRTCICECandidateStats.copy
def GstWebRTC.WebRTCICECandidateStats.copy (self):
    #python wrapper for 'gst_webrtc_ice_candidate_stats_copy'
	Parameters:
A copy of stats
Since : 1.22
gst_webrtc_ice_candidate_stats_free
gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats)
Helper function to free GstWebRTCICECandidateStats
Parameters:
stats
–
The GstWebRTCICECandidateStats to be free'd
Since : 1.22
GstWebRTC.WebRTCICECandidateStats.prototype.free
function GstWebRTC.WebRTCICECandidateStats.prototype.free(): {
    // javascript wrapper for 'gst_webrtc_ice_candidate_stats_free'
}
Helper function to free GstWebRTC.WebRTCICECandidateStats
Parameters:
The GstWebRTC.WebRTCICECandidateStats to be free'd
Since : 1.22
GstWebRTC.WebRTCICECandidateStats.free
def GstWebRTC.WebRTCICECandidateStats.free (self):
    #python wrapper for 'gst_webrtc_ice_candidate_stats_free'
Helper function to free GstWebRTC.WebRTCICECandidateStats
Parameters:
The GstWebRTC.WebRTCICECandidateStats to be free'd
Since : 1.22
GstWebRTCSCTPTransport
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstWebRTCSCTPTransport
Class structure
GstWebRTCSCTPTransportClass
GstWebRTC.WebRTCSCTPTransportClass
GstWebRTC.WebRTCSCTPTransportClass
GstWebRTC.WebRTCSCTPTransport
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCSCTPTransport
GstWebRTC.WebRTCSCTPTransport
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCSCTPTransport
Properties
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
Since : 1.16
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
Since : 1.16
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
Since : 1.16
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_CONNECTING
		(1)
		–
	connecting
GST_WEBRTC_DATA_CHANNEL_STATE_OPEN
		(2)
		–
	open
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING
		(3)
		–
	closing
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED
		(4)
		–
	closed
Since : 1.16
GstWebRTC.WebRTCDataChannelState
See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate
Members
GstWebRTC.WebRTCDataChannelState.CONNECTING
		(1)
		–
	connecting
GstWebRTC.WebRTCDataChannelState.OPEN
		(2)
		–
	open
GstWebRTC.WebRTCDataChannelState.CLOSING
		(3)
		–
	closing
GstWebRTC.WebRTCDataChannelState.CLOSED
		(4)
		–
	closed
Since : 1.16
GstWebRTC.WebRTCDataChannelState
See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate
Members
GstWebRTC.WebRTCDataChannelState.CONNECTING
		(1)
		–
	connecting
GstWebRTC.WebRTCDataChannelState.OPEN
		(2)
		–
	open
GstWebRTC.WebRTCDataChannelState.CLOSING
		(3)
		–
	closing
GstWebRTC.WebRTCDataChannelState.CLOSED
		(4)
		–
	closed
Since : 1.16
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
GST_WEBRTC_ERROR_INVALID_MODIFICATION
		(9)
		–
	invalid-modification (part of WebIDL specification)
(Since: 1.22)GST_WEBRTC_ERROR_TYPE_ERROR
		(10)
		–
	type-error (maps to JavaScript TypeError)
(Since: 1.22)Since : 1.20
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.INVALID_MODIFICATION
		(9)
		–
	invalid-modification (part of WebIDL specification)
(Since: 1.22)GstWebRTC.WebRTCError.TYPE_ERROR
		(10)
		–
	type-error (maps to JavaScript TypeError)
(Since: 1.22)Since : 1.20
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.INVALID_MODIFICATION
		(9)
		–
	invalid-modification (part of WebIDL specification)
(Since: 1.22)GstWebRTC.WebRTCError.TYPE_ERROR
		(10)
		–
	type-error (maps to JavaScript TypeError)
(Since: 1.22)Since : 1.20
GstWebRTCFECType
Members
GST_WEBRTC_FEC_TYPE_NONE
		(0)
		–
	none
GST_WEBRTC_FEC_TYPE_ULP_RED
		(1)
		–
	ulpfec + red
Since : 1.14.1
GstWebRTC.WebRTCFECType
Members
GstWebRTC.WebRTCFECType.NONE
		(0)
		–
	none
GstWebRTC.WebRTCFECType.ULP_RED
		(1)
		–
	ulpfec + red
Since : 1.14.1
GstWebRTC.WebRTCFECType
Members
GstWebRTC.WebRTCFECType.NONE
		(0)
		–
	none
GstWebRTC.WebRTCFECType.ULP_RED
		(1)
		–
	ulpfec + red
Since : 1.14.1
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
Since : 1.16
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
Since : 1.16
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
Since : 1.16
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 video
Since : 1.20
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 video
Since : 1.20
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 video
Since : 1.20
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
Since : 1.16
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
Since : 1.16
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
Since : 1.16
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
Since : 1.16
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
Since : 1.16
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
Since : 1.16
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
See https://w3c.github.io/webrtc-stats/#dom-rtcstatstype
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-connection
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
See https://w3c.github.io/webrtc-stats/#dom-rtcstatstype
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-connection
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
See https://w3c.github.io/webrtc-stats/#dom-rtcstatstype
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-connection
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 ()
Since : 1.20
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