GstWebRTC Enumerations
Enumerations
GstWebRTCBundlePolicy
GST_WEBRTC_BUNDLE_POLICY_NONE: none GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
Members
GST_WEBRTC_BUNDLE_POLICY_NONE
		(0)
		–
	GST_WEBRTC_BUNDLE_POLICY_BALANCED
		(1)
		–
	GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT
		(2)
		–
	GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE
		(3)
		–
	GstWebRTC.WebRTCBundlePolicy
GST_WEBRTC_BUNDLE_POLICY_NONE: none GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
Members
GstWebRTC.WebRTCBundlePolicy.NONE
		(0)
		–
	GstWebRTC.WebRTCBundlePolicy.BALANCED
		(1)
		–
	GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT
		(2)
		–
	GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE
		(3)
		–
	GstWebRTC.WebRTCBundlePolicy
GST_WEBRTC_BUNDLE_POLICY_NONE: none GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
Members
GstWebRTC.WebRTCBundlePolicy.NONE
		(0)
		–
	GstWebRTC.WebRTCBundlePolicy.BALANCED
		(1)
		–
	GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT
		(2)
		–
	GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE
		(3)
		–
	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
GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
Members
GST_WEBRTC_DATA_CHANNEL_STATE_NEW
		(0)
		–
	GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING
		(1)
		–
	GST_WEBRTC_DATA_CHANNEL_STATE_OPEN
		(2)
		–
	GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING
		(3)
		–
	GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED
		(4)
		–
	GstWebRTC.WebRTCDataChannelState
GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
Members
GstWebRTC.WebRTCDataChannelState.NEW
		(0)
		–
	GstWebRTC.WebRTCDataChannelState.CONNECTING
		(1)
		–
	GstWebRTC.WebRTCDataChannelState.OPEN
		(2)
		–
	GstWebRTC.WebRTCDataChannelState.CLOSING
		(3)
		–
	GstWebRTC.WebRTCDataChannelState.CLOSED
		(4)
		–
	GstWebRTC.WebRTCDataChannelState
GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
Members
GstWebRTC.WebRTCDataChannelState.NEW
		(0)
		–
	GstWebRTC.WebRTCDataChannelState.CONNECTING
		(1)
		–
	GstWebRTC.WebRTCDataChannelState.OPEN
		(2)
		–
	GstWebRTC.WebRTCDataChannelState.CLOSING
		(3)
		–
	GstWebRTC.WebRTCDataChannelState.CLOSED
		(4)
		–
	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
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
Members
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL
		(0)
		–
	GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY
		(1)
		–
	GstWebRTC.WebRTCICETransportPolicy
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
Members
GstWebRTC.WebRTCICETransportPolicy.ALL
		(0)
		–
	GstWebRTC.WebRTCICETransportPolicy.RELAY
		(1)
		–
	GstWebRTC.WebRTCICETransportPolicy
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
Members
GstWebRTC.WebRTCICETransportPolicy.ALL
		(0)
		–
	GstWebRTC.WebRTCICETransportPolicy.RELAY
		(1)
		–
	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
GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low GST_WEBRTC_PRIORITY_TYPE_LOW: low GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium GST_WEBRTC_PRIORITY_TYPE_HIGH: high
Members
GST_WEBRTC_PRIORITY_TYPE_VERY_LOW
		(1)
		–
	GST_WEBRTC_PRIORITY_TYPE_LOW
		(2)
		–
	GST_WEBRTC_PRIORITY_TYPE_MEDIUM
		(3)
		–
	GST_WEBRTC_PRIORITY_TYPE_HIGH
		(4)
		–
	GstWebRTC.WebRTCPriorityType
GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low GST_WEBRTC_PRIORITY_TYPE_LOW: low GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium GST_WEBRTC_PRIORITY_TYPE_HIGH: high
Members
GstWebRTC.WebRTCPriorityType.VERY_LOW
		(1)
		–
	GstWebRTC.WebRTCPriorityType.LOW
		(2)
		–
	GstWebRTC.WebRTCPriorityType.MEDIUM
		(3)
		–
	GstWebRTC.WebRTCPriorityType.HIGH
		(4)
		–
	GstWebRTC.WebRTCPriorityType
GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low GST_WEBRTC_PRIORITY_TYPE_LOW: low GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium GST_WEBRTC_PRIORITY_TYPE_HIGH: high
Members
GstWebRTC.WebRTCPriorityType.VERY_LOW
		(1)
		–
	GstWebRTC.WebRTCPriorityType.LOW
		(2)
		–
	GstWebRTC.WebRTCPriorityType.MEDIUM
		(3)
		–
	GstWebRTC.WebRTCPriorityType.HIGH
		(4)
		–
	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
GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
Members
GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW
		(0)
		–
	GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING
		(1)
		–
	GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED
		(2)
		–
	GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED
		(3)
		–
	GstWebRTC.WebRTCSCTPTransportState
GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
Members
GstWebRTC.WebRTCSCTPTransportState.NEW
		(0)
		–
	GstWebRTC.WebRTCSCTPTransportState.CONNECTING
		(1)
		–
	GstWebRTC.WebRTCSCTPTransportState.CONNECTED
		(2)
		–
	GstWebRTC.WebRTCSCTPTransportState.CLOSED
		(3)
		–
	GstWebRTC.WebRTCSCTPTransportState
GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
Members
GstWebRTC.WebRTCSCTPTransportState.NEW
		(0)
		–
	GstWebRTC.WebRTCSCTPTransportState.CONNECTING
		(1)
		–
	GstWebRTC.WebRTCSCTPTransportState.CONNECTED
		(2)
		–
	GstWebRTC.WebRTCSCTPTransportState.CLOSED
		(3)
		–
	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 */
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