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) –
No description available
GST_WEBRTC_BUNDLE_POLICY_BALANCED (1) –
No description available
GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT (2) –
No description available
GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE (3) –
No description available

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) –
No description available
GstWebRTC.WebRTCBundlePolicy.BALANCED (1) –
No description available
GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT (2) –
No description available
GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE (3) –
No description available

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) –
No description available
GstWebRTC.WebRTCBundlePolicy.BALANCED (1) –
No description available
GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT (2) –
No description available
GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE (3) –
No description available

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) –
No description available
GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING (1) –
No description available
GST_WEBRTC_DATA_CHANNEL_STATE_OPEN (2) –
No description available
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING (3) –
No description available
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED (4) –
No description available

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) –
No description available
GstWebRTC.WebRTCDataChannelState.CONNECTING (1) –
No description available
GstWebRTC.WebRTCDataChannelState.OPEN (2) –
No description available
GstWebRTC.WebRTCDataChannelState.CLOSING (3) –
No description available
GstWebRTC.WebRTCDataChannelState.CLOSED (4) –
No description available

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) –
No description available
GstWebRTC.WebRTCDataChannelState.CONNECTING (1) –
No description available
GstWebRTC.WebRTCDataChannelState.OPEN (2) –
No description available
GstWebRTC.WebRTCDataChannelState.CLOSING (3) –
No description available
GstWebRTC.WebRTCDataChannelState.CLOSED (4) –
No description available

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) –
No description available
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY (1) –
No description available

GstWebRTC.WebRTCICETransportPolicy

GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay

Members
GstWebRTC.WebRTCICETransportPolicy.ALL (0) –
No description available
GstWebRTC.WebRTCICETransportPolicy.RELAY (1) –
No description available

GstWebRTC.WebRTCICETransportPolicy

GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay

Members
GstWebRTC.WebRTCICETransportPolicy.ALL (0) –
No description available
GstWebRTC.WebRTCICETransportPolicy.RELAY (1) –
No description available

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) –
No description available
GST_WEBRTC_PRIORITY_TYPE_LOW (2) –
No description available
GST_WEBRTC_PRIORITY_TYPE_MEDIUM (3) –
No description available
GST_WEBRTC_PRIORITY_TYPE_HIGH (4) –
No description available

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) –
No description available
GstWebRTC.WebRTCPriorityType.LOW (2) –
No description available
GstWebRTC.WebRTCPriorityType.MEDIUM (3) –
No description available
GstWebRTC.WebRTCPriorityType.HIGH (4) –
No description available

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) –
No description available
GstWebRTC.WebRTCPriorityType.LOW (2) –
No description available
GstWebRTC.WebRTCPriorityType.MEDIUM (3) –
No description available
GstWebRTC.WebRTCPriorityType.HIGH (4) –
No description available

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) –
No description available
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING (1) –
No description available
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED (2) –
No description available
GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED (3) –
No description available

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) –
No description available
GstWebRTC.WebRTCSCTPTransportState.CONNECTING (1) –
No description available
GstWebRTC.WebRTCSCTPTransportState.CONNECTED (2) –
No description available
GstWebRTC.WebRTCSCTPTransportState.CLOSED (3) –
No description available

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) –
No description available
GstWebRTC.WebRTCSCTPTransportState.CONNECTING (1) –
No description available
GstWebRTC.WebRTCSCTPTransportState.CONNECTED (2) –
No description available
GstWebRTC.WebRTCSCTPTransportState.CLOSED (3) –
No description available

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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