| Top |  |  |  |  | 
| GstWebRTCPeerConnectionState | connection-state | Read | 
| GstWebRTCICEConnectionState | ice-connection-state | Read | 
| GstWebRTCICEGatheringState | ice-gathering-state | Read | 
| GstWebRTCSessionDescription * | local-description | Read / Write | 
| GstWebRTCSessionDescription * | remote-description | Read / Write | 
| GstWebRTCSignalingState | signaling-state | Read | 
| gchar * | stun-server | Read / Write | 
| gchar * | turn-server | Read / Write | 
| GstWebRTCBundlePolicy | bundle-policy | Read / Write | 
| GstWebRTCICETransportPolicy | ice-transport-policy | Read / Write | 
| void | add-ice-candidate | Action | 
| GstWebRTCRTPTransceiver* | add-transceiver | Action | 
| void | create-answer | Action | 
| void | create-offer | Action | 
| void | get-stats | Action | 
| GArray* | get-transceivers | Action | 
| void | on-ice-candidate | Run Last | 
| void | on-negotiation-needed | Run Last | 
| void | set-local-description | Action | 
| void | set-remote-description | Action | 
| gboolean | add-turn-server | Action | 
| GstWebRTCDataChannel* | create-data-channel | Action | 
| void | on-data-channel | Run Last | 
| void | on-new-transceiver | Run Last | 
| GstWebRTCRTPTransceiver* | get-transceiver | Action | 
GObject ╰── GInitiallyUnowned ╰── GstObject ╰── GstElement ╰── GstBin ╰── GstWebRTCBin
| plugin | webrtc | 
| author | Matthew Waters <matthew@centricular.com> | 
| class | Filter/Network/WebRTC | 
“connection-state” property  “connection-state”         GstWebRTCPeerConnectionState
The overall connection state of this element.
Flags: Read
Default value: GST_WEBRTC_PEER_CONNECTION_STATE_NEW
“ice-connection-state” property  “ice-connection-state”     GstWebRTCICEConnectionState
The collective connection state of all ICETransport's.
Flags: Read
Default value: GST_WEBRTC_ICE_CONNECTION_STATE_NEW
“ice-gathering-state” property  “ice-gathering-state”      GstWebRTCICEGatheringState
The collective gathering state of all ICETransport's.
Flags: Read
Default value: GST_WEBRTC_ICE_GATHERING_STATE_NEW
“local-description” property  “local-description”        GstWebRTCSessionDescription *
The local SDP description to use for this connection.
Flags: Read / Write
“remote-description” property  “remote-description”       GstWebRTCSessionDescription *
The remote SDP description to use for this connection.
Flags: Read / Write
“signaling-state” property  “signaling-state”          GstWebRTCSignalingState
The signaling state of this element.
Flags: Read
Default value: GST_WEBRTC_SIGNALING_STATE_STABLE
“stun-server” property“stun-server” gchar *
The STUN server of the form stun://hostname:port.
Flags: Read / Write
Default value: NULL
“turn-server” property“turn-server” gchar *
The TURN server of the form turn(s)://username:password@host:port. This is a convenience property, use #GstWebRTCBin::add-turn-server if you wish to use multiple TURN servers.
Flags: Read / Write
Default value: NULL
“bundle-policy” property  “bundle-policy”            GstWebRTCBundlePolicy
The policy to apply for bundling.
Flags: Read / Write
Default value: GST_WEBRTC_BUNDLE_POLICY_NONE
“add-ice-candidate” signalvoid user_function (GstWebRTCBin *object, guint mline_index, gchar *ice-candidate, gpointer user_data)
| object | the GstWebRtcBin | |
| mline_index | the index of the media description in the SDP | |
| ice-candidate | an ice candidate | |
| user_data | user data set when the signal handler was connected. | 
Flags: Action
“add-transceiver” signalGstWebRTCRTPTransceiver* user_function (GstWebRTCBin *object, GstWebRTCRTPTransceiverDirection direction, GstCaps *caps, gpointer user_data)
| object | the GstWebRtcBin | |
| direction | the direction of the new transceiver | |
| caps | the codec preferences for this transceiver. | [allow none] | 
| user_data | user data set when the signal handler was connected. | 
Flags: Action
“create-answer” signalvoid user_function (GstWebRTCBin *object, GstStructure *options, GstPromise *promise, gpointer user_data)
| object | the GstWebRtcBin | |
| options | create-answer options | |
| promise | a GstPromise which will contain the answer | |
| user_data | user data set when the signal handler was connected. | 
Flags: Action
“create-offer” signalvoid user_function (GstWebRTCBin *object, GstStructure *options, GstPromise *promise, gpointer user_data)
| object | the GstWebRtcBin | |
| options | create-offer options | |
| promise | a GstPromise which will contain the offer | |
| user_data | user data set when the signal handler was connected. | 
Flags: Action
“get-stats” signalvoid user_function (GstWebRTCBin *object, GstPad *pad, GstPromise *promise, gpointer user_data)
The promise
 will contain the result of retrieving the session statistics.
The structure will be named 'application/x-webrtc-stats and contain the
following based on the webrtc-stats spec available from
https://www.w3.org/TR/webrtc-stats/.  As the webrtc-stats spec is a draft
and is constantly changing these statistics may be changed to fit with
the latest spec.
Each field key is a unique identifer for each RTCStats (https://www.w3.org/TR/webrtc/rtcstats-dictionary) value (another GstStructure) in the RTCStatsReport (https://www.w3.org/TR/webrtc/rtcstatsreport-object). Each supported field in the RTCStats subclass is outlined below.
Each statistics structure contains the following values as defined by the RTCStats dictionary (https://www.w3.org/TR/webrtc/rtcstats-dictionary).
"timestamp" G_TYPE_DOUBLE timestamp the statistics were generated "type" GST_TYPE_WEBRTC_STATS_TYPE the type of statistics reported "id" G_TYPE_STRING unique identifier
RTCCodecStats supported fields (https://w3c.github.io/webrtc-stats/codec-dict*)
"payload-type" G_TYPE_UINT the rtp payload number in use "clock-rate" G_TYPE_UINT the rtp clock-rate
RTCRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/streamstats-dict*)
"ssrc" G_TYPE_STRING the rtp sequence src in use "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream "codec-id" G_TYPE_STRING identifier for the associated RTCCodecStats for this stream "fir-count" G_TYPE_UINT FIR requests received by the sender (only for local statistics) "pli-count" G_TYPE_UINT PLI requests received by the sender (only for local statistics) "nack-count" G_TYPE_UINT NACK requests received by the sender (only for local statistics)
RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/receivedrtpstats-dict*)
"packets-received" G_TYPE_UINT64 number of packets received (only for local inbound) "bytes-received" G_TYPE_UINT64 number of bytes received (only for local inbound) "packets-lost" G_TYPE_UINT number of packets lost "jitter" G_TYPE_DOUBLE packet jitter measured in secondss
RTCInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/inboundrtpstats-dict*)
"remote-id" G_TYPE_STRING identifier for the associated RTCRemoteOutboundRTPStreamStats
RTCRemoteInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/remoteinboundrtpstats-dict*)
"local-id" G_TYPE_STRING identifier for the associated RTCOutboundRTPSTreamStats "round-trip-time" G_TYPE_DOUBLE round trip time of packets measured in seconds
RTCSentRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/sentrtpstats-dict*)
"packets-sent" G_TYPE_UINT64 number of packets sent (only for local outbound) "bytes-sent" G_TYPE_UINT64 number of packets sent (only for local outbound)
RTCOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/outboundrtpstats-dict*)
"remote-id" G_TYPE_STRING identifier for the associated RTCRemoteInboundRTPSTreamStats
RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/remoteoutboundrtpstats-dict*)
"local-id" G_TYPE_STRING identifier for the associated RTCInboundRTPSTreamStats
| object | the GstWebRtcBin | |
| pad | [nullable] | |
| promise | a GstPromise for the result | |
| user_data | user data set when the signal handler was connected. | 
Flags: Action
“get-transceivers” signalGArray* user_function (GstWebRTCBin *object, gpointer user_data)
Flags: Action
“on-ice-candidate” signalvoid user_function (GstWebRTCBin *object, guint mline_index, gchar *candidate, gpointer user_data)
| object | the GstWebRtcBin | |
| mline_index | the index of the media description in the SDP | |
| candidate | the ICE candidate | |
| user_data | user data set when the signal handler was connected. | 
Flags: Run Last
“on-negotiation-needed” signalvoid user_function (GstWebRTCBin *object, gpointer user_data)
Flags: Run Last
“set-local-description” signalvoid user_function (GstWebRTCBin *object, GstWebRTCSessionDescription *desc, GstPromise *promise, gpointer user_data)
| object | the GstWebRtcBin | |
| desc | a GstWebRTCSessionDescription description | |
| promise | a GstPromise to be notified when it's set. | [nullable] | 
| user_data | user data set when the signal handler was connected. | 
Flags: Action
“set-remote-description” signalvoid user_function (GstWebRTCBin *object, GstWebRTCSessionDescription *desc, GstPromise *promise, gpointer user_data)
| object | the GstWebRtcBin | |
| desc | a GstWebRTCSessionDescription description | |
| promise | a GstPromise to be notified when it's set. | [nullable] | 
| user_data | user data set when the signal handler was connected. | 
Flags: Action
“add-turn-server” signalgboolean user_function (GstWebRTCBin *object, gchar *uri, gpointer user_data)
Add a turn server to obtain ICE candidates from
| object | the GstWebRtcBin | |
| uri | The uri of the server of the form turn(s)://username:password | |
| user_data | user data set when the signal handler was connected. | 
Flags: Action
“create-data-channel” signalGstWebRTCDataChannel* user_function (GstWebRTCBin *gstwebrtcbin, gchar *arg1, GstStructure *arg2, gpointer user_data)
Flags: Action
“on-data-channel” signalvoid user_function (GstWebRTCBin *object, GstWebRTCDataChannel *candidate, gpointer user_data)
| object | the GstWebRtcBin | |
| candidate | the new GstWebRTCDataChannel | |
| user_data | user data set when the signal handler was connected. | 
Flags: Run Last
“on-new-transceiver” signalvoid user_function (GstWebRTCBin *object, GstWebRTCRTPTransceiver *candidate, gpointer user_data)
| object | the GstWebRtcBin | |
| candidate | the new GstWebRTCRTPTransceiver | |
| user_data | user data set when the signal handler was connected. | 
Flags: Run Last
“get-transceiver” signalGstWebRTCRTPTransceiver* user_function (GstWebRTCBin *object, gint idx, gpointer user_data)
| object | the GstWebRtcBin | |
| idx | The index of the transceiver | |
| user_data | user data set when the signal handler was connected. | 
Flags: Action
Since: 1.16