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 @data as a data message over @channel. TRUE if @channel is open and data could be queued a #GstWebRTCDataChannel a #GBytes or %NULL Send @str as a string message over @channel. a #GstWebRTCDataChannel a string or %NULL Send @str as a string message over @channel. TRUE if @channel is open and data could be queued 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> connecting open closing closed See <https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype> for more information. data-channel-failure dtls-failure fingerprint-failure sctp-failure sdp-syntax-error hardware-encoder-not-available encoder-error invalid-state (part of WebIDL specification) GStreamer-specific failure, not matching any other value from the specification invalid-modification (part of WebIDL specification) type-error (maps to JavaScript TypeError) none ulpfec + red The #GstWebRTCICE The #GstWebRTCICEStream The ICE candidate A #GstPromise for task notifications (Since: 1.24) The #GstWebRTCICEStream, or %NULL The #GstWebRTCICE The session id FALSE on error, TRUE otherwise The #GstWebRTCICE URI of the TURN server The #GstWebRTCICETransport, or %NULL The #GstWebRTCICE The #GstWebRTCICEStream The #GstWebRTCICEComponent FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream Get HTTP Proxy to be used when connecting to TURN server. URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=<alpn>] Get HTTP Proxy to be used when connecting to TURN server. The #GstWebRTCICE TRUE if set as controller, FALSE otherwise The #GstWebRTCICE FALSE on failure, otherwise @local_stats @remote_stats will be set The #GstWebRTCICE The #GstWebRTCICEStream A pointer to #GstWebRTCICECandidateStats for local candidate pointer to #GstWebRTCICECandidateStats for remote candidate URI of the STUN sever The #GstWebRTCICE URI of the TURN sever The #GstWebRTCICE The #GstWebRTCICE TRUE to enable force relay Set HTTP Proxy to be used when connecting to TURN server. The #GstWebRTCICE URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=<alpn>] The #GstWebRTCICE TRUE to set as controller FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream ICE username ICE password The #GstWebRTCICE The #GstWebRTCICEOnCandidateFunc callback function User data passed to the callback function a #GDestroyNotify when the candidate is no longer needed FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream ICE username ICE password The #GstWebRTCICE URI of the STUN server The #GstWebRTCICE The #GstWebRTCICEStream ToS to be set The #GstWebRTCICE URI of the TURN sever The #GstWebRTCICE The #GstWebRTCICEStream The ICE candidate A #GstPromise for task notifications (Since: 1.24) The #GstWebRTCICEStream, or %NULL The #GstWebRTCICE The session id FALSE on error, TRUE otherwise The #GstWebRTCICE URI of the TURN server The #GstWebRTCICETransport, or %NULL The #GstWebRTCICE The #GstWebRTCICEStream The #GstWebRTCICEComponent FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=<alpn>] Get HTTP Proxy to be used when connecting to TURN server. The #GstWebRTCICE TRUE if set as controller, FALSE otherwise The #GstWebRTCICE List of local candidates The #GstWebRTCICE The #GstWebRTCICEStream List of remote candidates The #GstWebRTCICE The #GstWebRTCICEStream FALSE on failure, otherwise @local_stats @remote_stats will be set The #GstWebRTCICE The #GstWebRTCICEStream A pointer to #GstWebRTCICECandidateStats for local candidate pointer to #GstWebRTCICECandidateStats for remote candidate URI of the STUN sever The #GstWebRTCICE URI of the TURN sever The #GstWebRTCICE The #GstWebRTCICE TRUE to enable force relay Set HTTP Proxy to be used when connecting to TURN server. The #GstWebRTCICE URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=<alpn>] The #GstWebRTCICE TRUE to set as controller FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream ICE username ICE password The #GstWebRTCICE The #GstWebRTCICEOnCandidateFunc callback function User data passed to the callback function a #GDestroyNotify when the candidate is no longer needed FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream ICE username ICE password The #GstWebRTCICE URI of the STUN server The #GstWebRTCICE The #GstWebRTCICEStream ToS to be set The #GstWebRTCICE URI of the TURN sever Maximum port for local rtp port range. min-rtp-port must be <= max-rtp-port Minimum port for local rtp port range. min-rtp-port must be <= max-rtp-port Add a local IP address to use for ICE candidate gathering. If none are supplied, they will be discovered automatically. Calling this signal stops automatic ICE gathering. whether the address could be added. The local IP address A copy of @stats The #GstWebRTCICE Helper function to free #GstWebRTCICECandidateStats The #GstWebRTCICECandidateStats to be free'd The #GstWebRTCICEStream, or %NULL The #GstWebRTCICE The session id The #GstWebRTCICETransport, or %NULL The #GstWebRTCICE The #GstWebRTCICEStream The #GstWebRTCICEComponent FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream The #GstWebRTCICE The #GstWebRTCICEStream The ICE candidate A #GstPromise for task notifications (Since: 1.24) FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream ICE username ICE password FALSE on error, TRUE otherwise The #GstWebRTCICE The #GstWebRTCICEStream ICE username ICE password FALSE on error, TRUE otherwise The #GstWebRTCICE URI of the TURN server The #GstWebRTCICE TRUE to set as controller TRUE if set as controller, FALSE otherwise The #GstWebRTCICE The #GstWebRTCICE TRUE to enable force relay The #GstWebRTCICE URI of the STUN server URI of the STUN sever The #GstWebRTCICE The #GstWebRTCICE URI of the TURN sever URI of the TURN sever The #GstWebRTCICE The #GstWebRTCICE URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=<alpn>] URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=<alpn>] Get HTTP Proxy to be used when connecting to TURN server. The #GstWebRTCICE The #GstWebRTCICE The #GstWebRTCICEStream ToS to be set The #GstWebRTCICE The #GstWebRTCICEOnCandidateFunc callback function User data passed to the callback function a #GDestroyNotify when the candidate is no longer needed FALSE on failure, otherwise @local_stats @remote_stats will be set The #GstWebRTCICE The #GstWebRTCICEStream A pointer to #GstWebRTCICECandidateStats for local candidate pointer to #GstWebRTCICECandidateStats for remote candidate 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 Callback function to be triggered on discovery of a new candidate The #GstWebRTCICE The stream id The discovered candidate User data that was set by #gst_webrtc_ice_set_on_ice_candidate controlled controlling the #GstWebRTCICETransport, or %NULL the #GstWebRTCICEStream The #GstWebRTCICEComponent FALSE on error, TRUE otherwise the #GstWebRTCICEStream the #GstWebRTCICETransport, or %NULL the #GstWebRTCICEStream The #GstWebRTCICEComponent FALSE on error, TRUE otherwise the #GstWebRTCICEStream the #GstWebRTCICETransport, or %NULL the #GstWebRTCICEStream The #GstWebRTCICEComponent FALSE on error, TRUE otherwise the #GstWebRTCICEStream 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 video 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 An object to track the receiving aspect of the stream Mostly matches the WebRTC RTCRtpReceiver interface. The DTLS transport for this receiver 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 DTLS transport for this sender Mostly matches the WebRTC RTCRtpTransceiver interface. 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 See <https://w3c.github.io/webrtc-stats/#dom-rtcstatstype> codec inbound-rtp outbound-rtp remote-inbound-rtp remote-outbound-rtp csrc peer-connection 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> See the [specification](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