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class GstWebRTC.WebRTCBundlePolicy(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCBundlePolicy.NONE: none GstWebRTC.WebRTCBundlePolicy.BALANCED: balanced GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT: max-compat GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE: max-bundle See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.

New in version 1.16.

NONE = 0
BALANCED = 1
MAX_COMPAT = 2
MAX_BUNDLE = 3
class GstWebRTC.WebRTCDTLSSetup(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCDTLSSetup.NONE: none GstWebRTC.WebRTCDTLSSetup.ACTPASS: actpass GstWebRTC.WebRTCDTLSSetup.ACTIVE: sendonly GstWebRTC.WebRTCDTLSSetup.PASSIVE: recvonly

NONE = 0
ACTPASS = 1
ACTIVE = 2
PASSIVE = 3
class GstWebRTC.WebRTCDTLSTransportState(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCDTLSTransportState.NEW: new GstWebRTC.WebRTCDTLSTransportState.CLOSED: closed GstWebRTC.WebRTCDTLSTransportState.FAILED: failed GstWebRTC.WebRTCDTLSTransportState.CONNECTING: connecting GstWebRTC.WebRTCDTLSTransportState.CONNECTED: connected

NEW = 0
CLOSED = 1
FAILED = 2
CONNECTING = 3
CONNECTED = 4
class GstWebRTC.WebRTCDataChannelState(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCDataChannelState.NEW: new GstWebRTC.WebRTCDataChannelState.CONNECTING: connection GstWebRTC.WebRTCDataChannelState.OPEN: open GstWebRTC.WebRTCDataChannelState.CLOSING: closing GstWebRTC.WebRTCDataChannelState.CLOSED: closed See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate

New in version 1.16.

NEW = 0
CONNECTING = 1
OPEN = 2
CLOSING = 3
CLOSED = 4
class GstWebRTC.WebRTCFECType(value)

Bases: GObject.GEnum

New in version 1.14.1.

NONE = 0

none

ULP_RED = 1

ulpfec + red

class GstWebRTC.WebRTCICEComponent(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCICEComponent.RTP, GstWebRTC.WebRTCICEComponent.RTCP,

RTP = 0
RTCP = 1
class GstWebRTC.WebRTCICEConnectionState(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCICEConnectionState.NEW: new GstWebRTC.WebRTCICEConnectionState.CHECKING: checking GstWebRTC.WebRTCICEConnectionState.CONNECTED: connected GstWebRTC.WebRTCICEConnectionState.COMPLETED: completed GstWebRTC.WebRTCICEConnectionState.FAILED: failed GstWebRTC.WebRTCICEConnectionState.DISCONNECTED: disconnected GstWebRTC.WebRTCICEConnectionState.CLOSED: closed See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate

NEW = 0
CHECKING = 1
CONNECTED = 2
COMPLETED = 3
FAILED = 4
DISCONNECTED = 5
CLOSED = 6
class GstWebRTC.WebRTCICEGatheringState(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCICEGatheringState.NEW: new GstWebRTC.WebRTCICEGatheringState.GATHERING: gathering GstWebRTC.WebRTCICEGatheringState.COMPLETE: complete See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate

NEW = 0
GATHERING = 1
COMPLETE = 2
class GstWebRTC.WebRTCICERole(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCICERole.CONTROLLED: controlled GstWebRTC.WebRTCICERole.CONTROLLING: controlling

CONTROLLED = 0
CONTROLLING = 1
class GstWebRTC.WebRTCICETransportPolicy(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCICETransportPolicy.ALL: all GstWebRTC.WebRTCICETransportPolicy.RELAY: relay See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.

New in version 1.16.

ALL = 0
RELAY = 1
class GstWebRTC.WebRTCPeerConnectionState(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCPeerConnectionState.NEW: new GstWebRTC.WebRTCPeerConnectionState.CONNECTING: connecting GstWebRTC.WebRTCPeerConnectionState.CONNECTED: connected GstWebRTC.WebRTCPeerConnectionState.DISCONNECTED: disconnected GstWebRTC.WebRTCPeerConnectionState.FAILED: failed GstWebRTC.WebRTCPeerConnectionState.CLOSED: closed See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate

NEW = 0
CONNECTING = 1
CONNECTED = 2
DISCONNECTED = 3
FAILED = 4
CLOSED = 5
class GstWebRTC.WebRTCPriorityType(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCPriorityType.VERY_LOW: very-low GstWebRTC.WebRTCPriorityType.LOW: low GstWebRTC.WebRTCPriorityType.MEDIUM: medium GstWebRTC.WebRTCPriorityType.HIGH: high See http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype

New in version 1.16.

VERY_LOW = 1
LOW = 2
MEDIUM = 3
HIGH = 4
class GstWebRTC.WebRTCRTPTransceiverDirection(value)

Bases: GObject.GEnum

NONE = 0
INACTIVE = 1
SENDONLY = 2
RECVONLY = 3
SENDRECV = 4
class GstWebRTC.WebRTCSCTPTransportState(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCSCTPTransportState.NEW: new GstWebRTC.WebRTCSCTPTransportState.CONNECTING: connecting GstWebRTC.WebRTCSCTPTransportState.CONNECTED: connected GstWebRTC.WebRTCSCTPTransportState.CLOSED: closed See http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate

New in version 1.16.

NEW = 0
CONNECTING = 1
CONNECTED = 2
CLOSED = 3
class GstWebRTC.WebRTCSDPType(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCSDPType.OFFER: offer GstWebRTC.WebRTCSDPType.PRANSWER: pranswer GstWebRTC.WebRTCSDPType.ANSWER: answer GstWebRTC.WebRTCSDPType.ROLLBACK: rollback See http://w3c.github.io/webrtc-pc/#rtcsdptype

classmethod to_string(type)
Parameters:type (GstWebRTC.WebRTCSDPType) – a GstWebRTC.WebRTCSDPType
Returns:the string representation of type or “unknown” when type is not recognized.
Return type:str
OFFER = 1
PRANSWER = 2
ANSWER = 3
ROLLBACK = 4
class GstWebRTC.WebRTCSignalingState(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCSignalingState.STABLE: stable GstWebRTC.WebRTCSignalingState.CLOSED: closed GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_OFFER: have-local-offer GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_OFFER: have-remote-offer GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_PRANSWER: have-local-pranswer GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_PRANSWER: have-remote-pranswer See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate

STABLE = 0
CLOSED = 1
HAVE_LOCAL_OFFER = 2
HAVE_REMOTE_OFFER = 3
HAVE_LOCAL_PRANSWER = 4
HAVE_REMOTE_PRANSWER = 5
class GstWebRTC.WebRTCStatsType(value)

Bases: GObject.GEnum

GstWebRTC.WebRTCStatsType.CODEC: codec GstWebRTC.WebRTCStatsType.INBOUND_RTP: inbound-rtp GstWebRTC.WebRTCStatsType.OUTBOUND_RTP: outbound-rtp GstWebRTC.WebRTCStatsType.REMOTE_INBOUND_RTP: remote-inbound-rtp GstWebRTC.WebRTCStatsType.REMOTE_OUTBOUND_RTP: remote-outbound-rtp GstWebRTC.WebRTCStatsType.CSRC: csrc GstWebRTC.WebRTCStatsType.PEER_CONNECTION: peer-connectiion GstWebRTC.WebRTCStatsType.DATA_CHANNEL: data-channel GstWebRTC.WebRTCStatsType.STREAM: stream GstWebRTC.WebRTCStatsType.TRANSPORT: transport GstWebRTC.WebRTCStatsType.CANDIDATE_PAIR: candidate-pair GstWebRTC.WebRTCStatsType.LOCAL_CANDIDATE: local-candidate GstWebRTC.WebRTCStatsType.REMOTE_CANDIDATE: remote-candidate GstWebRTC.WebRTCStatsType.CERTIFICATE: certificate

CODEC = 1
TRANSPORT = 10
CANDIDATE_PAIR = 11
LOCAL_CANDIDATE = 12
REMOTE_CANDIDATE = 13
CERTIFICATE = 14
INBOUND_RTP = 2
OUTBOUND_RTP = 3
REMOTE_INBOUND_RTP = 4
REMOTE_OUTBOUND_RTP = 5
CSRC = 6
PEER_CONNECTION = 7
DATA_CHANNEL = 8
STREAM = 9