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. 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> Direction of the transceiver. 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 <https://www.w3.org/TR/webrtc/#rtcdatachannel> <https://www.w3.org/TR/webrtc/#rtcdtlstransport> <https://www.w3.org/TR/webrtc/#rtcicetransport> <https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface> <https://www.w3.org/TR/webrtc/#rtcrtpsender-interface> <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class> <https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface> the string representation of @type or "unknown" when @type is not recognized. a #GstWebRTCSDPType