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 See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information. none actpass sendonly recvonly new closed failed connecting connected Close the @channel. a #GstWebRTCDataChannel Send @data as a data message over @channel. a #GstWebRTCDataChannel a #GBytes or %NULL Send @str as a string message over @channel. a #GstWebRTCDataChannel a string or %NULL Close the @channel. a #GstWebRTCDataChannel Signal that the data channel reached a low buffered amount. Should only be used by subclasses. a #GstWebRTCDataChannel Signal that the data channel was closed. Should only be used by subclasses. a #GstWebRTCDataChannel Signal that the data channel had an error. Should only be used by subclasses. a #GstWebRTCDataChannel a #GError Signal that the data channel received a data message. Should only be used by subclasses. a #GstWebRTCDataChannel a #GBytes or %NULL Signal that the data channel received a string message. Should only be used by subclasses. a #GstWebRTCDataChannel a string or %NULL Signal that the data channel was opened. Should only be used by subclasses. a #GstWebRTCDataChannel Send @data as a data message over @channel. a #GstWebRTCDataChannel a #GBytes or %NULL Send @str as a string message over @channel. a #GstWebRTCDataChannel a string or %NULL Close the data channel the #GError thrown a #GBytes of the data received the data received as a string a #GBytes with the data the data to send as a string a #GstWebRTCDataChannel a #GBytes or %NULL a #GstWebRTCDataChannel a string or %NULL a #GstWebRTCDataChannel 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 See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate> none ulpfec + red RTP component RTCP component See <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate> new checking connected completed failed disconnected closed See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate> new gathering complete controlled controlling GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information. https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind Kind has not yet been set Kind is audio Kind is audio See <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate> new connecting connected disconnected failed closed 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 See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype> An object to track the receiving aspect of the stream Mostly matches the WebRTC RTCRtpReceiver interface. The transport for RTP packets The transport for RTCP packets without rtcp-mux An object to track the sending aspect of the stream Mostly matches the WebRTC RTCRtpSender interface. Sets the content of the IPv4 Type of Service (ToS), also known as DSCP (Differentiated Services Code Point). This also sets the Traffic Class field of IPv6. a #GstWebRTCRTPSender The priority of this sender The priority from which to set the DSCP field on packets The transport for RTP packets The transport for RTCP packets without rtcp-mux Unused The priority of the stream (Since: 1.20) Mostly matches the WebRTC RTCRtpTransceiver interface. Direction of the transceiver. the mline number this transceiver corresponds to The media ID of the m-line associated with this transceiver. This association is established, when possible, whenever either a local or remote description is applied. This field is NULL if neither a local or remote description has been applied, or if its associated m-line is rejected by either a remote offer or any answer. Indicates whether or not sending and receiving using the paired #GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled, either due to SDP offer/answer The #GstWebRTCRTPSender object responsible sending data to the remote peer The #GstWebRTCRTPReceiver object responsible for receiver data from the remote peer. The transceiver's desired direction. The transceiver's current direction (read-only) A caps representing the codec preferences (read-only) Type of media (Since: 1.20) none inactive sendonly recvonly sendrecv 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 See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate> See <http://w3c.github.io/webrtc-pc/#rtcsdptype> offer pranswer answer rollback the string representation of @type or "unknown" when @type is not recognized. a #GstWebRTCSDPType See <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class> the #GstWebRTCSDPType of the description the #GstSDPMessage of the description a new #GstWebRTCSessionDescription from @type and @sdp a #GstWebRTCSDPType a #GstSDPMessage a new copy of @src a #GstWebRTCSessionDescription Free @desc and all associated resources a #GstWebRTCSessionDescription See <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate> stable closed have-local-offer have-remote-offer have-local-pranswer have-remote-pranswer codec inbound-rtp outbound-rtp remote-inbound-rtp remote-outbound-rtp csrc peer-connectiion data-channel stream transport candidate-pair local-candidate remote-candidate certificate the string representation of @type or "unknown" when @type is not recognized. a #GstWebRTCSDPType