Enums

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class GstWebRTC.WebRTCBundlePolicy(value)

Bases: GObject.GEnum

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

none

BALANCED = 1

balanced

MAX_COMPAT = 2

max-compat

MAX_BUNDLE = 3

max-bundle

class GstWebRTC.WebRTCDTLSRole(value)

Bases: GObject.GEnum

New in version 1.28.

CLIENT = 0

client

SERVER = 1

server

UNKNOWN = 2

unknown

class GstWebRTC.WebRTCDTLSSetup(value)

Bases: GObject.GEnum

NONE = 0

none

ACTPASS = 1

actpass

ACTIVE = 2

sendonly

PASSIVE = 3

recvonly

class GstWebRTC.WebRTCDTLSTransportState(value)

Bases: GObject.GEnum

NEW = 0

new

CLOSED = 1

closed

FAILED = 2

failed

CONNECTING = 3

connecting

CONNECTED = 4

connected

class GstWebRTC.WebRTCDataChannelState(value)

Bases: GObject.GEnum

See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate>

New in version 1.16.

CONNECTING = 1

connecting

OPEN = 2

open

CLOSING = 3

closing

CLOSED = 4

closed

class GstWebRTC.WebRTCError(value)

Bases: GObject.GEnum

See <https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype> for more information.

New in version 1.20.

classmethod quark()
Return type:

int

New in version 1.20.

DATA_CHANNEL_FAILURE = 0

data-channel-failure

DTLS_FAILURE = 1

dtls-failure

TYPE_ERROR = 10

type-error (maps to JavaScript TypeError)

New in version 1.22.

FINGERPRINT_FAILURE = 2

fingerprint-failure

SCTP_FAILURE = 3

sctp-failure

SDP_SYNTAX_ERROR = 4

sdp-syntax-error

HARDWARE_ENCODER_NOT_AVAILABLE = 5

hardware-encoder-not-available

ENCODER_ERROR = 6

encoder-error

INVALID_STATE = 7

invalid-state (part of WebIDL specification)

INTERNAL_FAILURE = 8

GStreamer-specific failure, not matching any other value from the specification

INVALID_MODIFICATION = 9

invalid-modification (part of WebIDL specification)

New in version 1.22.

class GstWebRTC.WebRTCFECType(value)

Bases: GObject.GEnum

New in version 1.14.1.

NONE = 0

none

ULP_RED = 1

ulpfec + red

class GstWebRTC.WebRTCICECandidateProtocolType(value)

Bases: GObject.GEnum

New in version 1.28.

TCP = 0
UDP = 1
class GstWebRTC.WebRTCICECandidateType(value)

Bases: GObject.GEnum

New in version 1.28.

HOST = 0

The candidate is a host candidate, whose IP address as specified in the RTCIceCandidate.address property is in fact the true address of the remote peer.

SERVER_REFLEXIVE = 1

The candidate is a server reflexive candidate; the ip and port are a binding allocated by a NAT for an agent when it sent a packet through the NAT to a server. They can be learned by the STUN server and TURN server to represent the candidate’s peer anonymously.

PEER_REFLEXIVE = 2

The candidate is a peer reflexive candidate; the ip and port are a binding allocated by a NAT when it sent a STUN request to represent the candidate’s peer anonymously.

RELAYED = 3

The candidate is a relay candidate, obtained from a TURN server. The relay candidate’s IP address is an address the TURN server uses to forward the media between the two peers.

class GstWebRTC.WebRTCICEComponent(value)

Bases: GObject.GEnum

RTP = 0

RTP component

RTCP = 1

RTCP component

class GstWebRTC.WebRTCICEConnectionState(value)

Bases: GObject.GEnum

See <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate>

NEW = 0

new

CHECKING = 1

checking

CONNECTED = 2

connected

COMPLETED = 3

completed

FAILED = 4

failed

DISCONNECTED = 5

disconnected

CLOSED = 6

closed

class GstWebRTC.WebRTCICEGatheringState(value)

Bases: GObject.GEnum

See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate>

NEW = 0

new

GATHERING = 1

gathering

COMPLETE = 2

complete

class GstWebRTC.WebRTCICERole(value)

Bases: GObject.GEnum

CONTROLLED = 0

controlled

CONTROLLING = 1

controlling

class GstWebRTC.WebRTCICETcpCandidateType(value)

Bases: GObject.GEnum

New in version 1.28.

ACTIVE = 0

An “active” TCP candidate is one for which the transport will attempt to open an outbound connection but will not receive incoming connection requests.

PASSIVE = 1

A “passive” TCP candidate is one for which the transport will receive incoming connection attempts but not attempt a connection.

SO = 2

An “so” candidate is one for which the transport will attempt to open a connection simultaneously with its peer.

NONE = 3

Value used for non-TCP candidate type.

class GstWebRTC.WebRTCICETransportPolicy(value)

Bases: GObject.GEnum

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

all

RELAY = 1

relay

class GstWebRTC.WebRTCKind(value)

Bases: GObject.GEnum

https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind

New in version 1.20.

UNKNOWN = 0

Kind has not yet been set

AUDIO = 1

Kind is audio

VIDEO = 2

Kind is video

class GstWebRTC.WebRTCPeerConnectionState(value)

Bases: GObject.GEnum

See <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate>

NEW = 0

new

CONNECTING = 1

connecting

CONNECTED = 2

connected

DISCONNECTED = 3

disconnected

FAILED = 4

failed

CLOSED = 5

closed

class GstWebRTC.WebRTCPriorityType(value)

Bases: GObject.GEnum

See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>

New in version 1.16.

VERY_LOW = 1

very-low

LOW = 2

low

MEDIUM = 3

medium

HIGH = 4

high

class GstWebRTC.WebRTCRTPTransceiverDirection(value)

Bases: GObject.GEnum

NONE = 0

none

INACTIVE = 1

inactive

SENDONLY = 2

sendonly

RECVONLY = 3

recvonly

SENDRECV = 4

sendrecv

class GstWebRTC.WebRTCSCTPTransportState(value)

Bases: GObject.GEnum

See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate>

New in version 1.16.

NEW = 0

new

CONNECTING = 1

connecting

CONNECTED = 2

connected

CLOSED = 3

closed

class GstWebRTC.WebRTCSDPType(value)

Bases: GObject.GEnum

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

offer

PRANSWER = 2

pranswer

ANSWER = 3

answer

ROLLBACK = 4

rollback

class GstWebRTC.WebRTCSignalingState(value)

Bases: GObject.GEnum

See <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate>

STABLE = 0

stable

CLOSED = 1

closed

HAVE_LOCAL_OFFER = 2

have-local-offer

HAVE_REMOTE_OFFER = 3

have-remote-offer

HAVE_LOCAL_PRANSWER = 4

have-local-pranswer

HAVE_REMOTE_PRANSWER = 5

have-remote-pranswer

class GstWebRTC.WebRTCStatsType(value)

Bases: GObject.GEnum

See <https://w3c.github.io/webrtc-stats/#dom-rtcstatstype>

CODEC = 1

codec

TRANSPORT = 10

transport

CANDIDATE_PAIR = 11

candidate-pair

LOCAL_CANDIDATE = 12

local-candidate

REMOTE_CANDIDATE = 13

remote-candidate

CERTIFICATE = 14

certificate

INBOUND_RTP = 2

inbound-rtp

OUTBOUND_RTP = 3

outbound-rtp

REMOTE_INBOUND_RTP = 4

remote-inbound-rtp

REMOTE_OUTBOUND_RTP = 5

remote-outbound-rtp

CSRC = 6

csrc

PEER_CONNECTION = 7

peer-connection

DATA_CHANNEL = 8

data-channel

STREAM = 9

stream