GstWebRTC Enumerations

GstWebRTCICECandidateStats

Members

ipaddr (gchar *) –
No description available
port (guint) –
No description available
stream_id (guint) –
No description available
type (const gchar *) –
No description available
proto (const gchar *) –
No description available
relay_proto (const gchar *) –
No description available
prio (guint) –
No description available
url (gchar *) –
No description available
_gst_reserved (gpointer *) –
No description available

Since : 1.22

GstWebRTC.WebRTCICECandidateStats

Members

ipaddr (String) –
No description available
port (Number) –
No description available
stream_id (Number) –
No description available
type (String) –
No description available
proto (String) –
No description available
relay_proto (String) –
No description available
prio (Number) –
No description available
url (String) –
No description available
_gst_reserved ([ Object ]) –
No description available

Since : 1.22

GstWebRTC.WebRTCICECandidateStats

Members

ipaddr (str) –
No description available
port (int) –
No description available
stream_id (int) –
No description available
type (str) –
No description available
proto (str) –
No description available
relay_proto (str) –
No description available
prio (int) –
No description available
url (str) –
No description available
_gst_reserved ([ object ]) –
No description available

Since : 1.22

Methods

gst_webrtc_ice_candidate_stats_copy 

GstWebRTCICECandidateStats *
gst_webrtc_ice_candidate_stats_copy (GstWebRTCICECandidateStats * stats)

Parameters:

stats –

The GstWebRTCICE

Returns ( [transfer: full]) –

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:

The GstWebRTC.WebRTCICE

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:

The GstWebRTC.WebRTCICE

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

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 audio

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 audio

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 audio

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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