See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information. none balanced max-compat max-bundle 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 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 See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate> new connection open closing closed 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 See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information. all relay 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 See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype> very-low low medium high The DTLS transport for this receiver 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 DTLS transport for this sender Caps representing the codec preferences. The transceiver's current directionality, or none if the transceiver is stopped or has never participated in an exchange of offers and answers. To change the transceiver's directionality, set the value of the direction property. Direction of the transceiver. The kind of media this transceiver transports 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. none inactive sendonly recvonly sendrecv See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate> new connecting connected closed 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