See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information. none balanced max-compat max-bundle client server unknown none actpass sendonly recvonly new closed failed connecting connected Close the @channel. a #GstWebRTCDataChannel Send @data as a data message over @channel. Use gst_webrtc_data_channel_send_data_full() instead 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. Use gst_webrtc_data_channel_send_string_full() instead 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 Use gst_webrtc_data_channel_send_data_full() instead a #GBytes with the data Use gst_webrtc_data_channel_send_string_full() instead 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 Invoke the close procedure as specified in https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close. The #GstWebRTCICE a #GstPromise to be notified when the task is complete. 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 Use gst_webrtc_ice_transport_get_selected_candidate_pair(). 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 Invoke the close procedure as specified in https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close. The #GstWebRTCICE a #GstPromise to be notified when the task is complete. 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 Use gst_webrtc_ice_transport_get_selected_candidate_pair(). 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 String carrying the candidate-attribute as defined in section 15.1 of RFC5245 The assigned network component of the candidate (1 for RTP 2 for RTCP). The media stream "identification-tag" defined in [RFC5888] for the media component this candidate is associated with. The index (starting at zero) of the media description in the SDP this candidate is associated with. The #GstWebRTCICECandidateStats associated to this candidate. A copy of @candidate The #GstWebRTCICECandidate to be copied Helper function to free #GstWebRTCICECandidate The #GstWebRTCICECandidate to be free'd A copy of @pair The #GstWebRTCICE Helper function to free #GstWebRTCICECandidatePair The #GstWebRTCICECandidatePair to be free'd A string containing the address of the candidate. This value may be an IPv4 address, an IPv6 address, or a fully-qualified domain name The network port number used by the candidate A string that uniquely identifies the object that is being monitored to produce this set of statistics The candidate type A string specifying the protocol (tcp or udp) used to transmit data on the @port The candidate's priority, corresponding to RTCIceCandidate.priority For local candidates, the url property is the URL of the ICE server from which the candidate was received A copy of @stats The #GstWebRTCICE Helper function to free #GstWebRTCICECandidateStats The #GstWebRTCICECandidateStats to be free'd The candidate is a host candidate, whose IP address as specified in the RTCIceCandidate.address property is in fact the true address of the remote peer. The candidate is a server reflexive candidate; the ip and port are a binding allocated by a NAT for an agent when it sent a packet through the NAT to a server. They can be learned by the STUN server and TURN server to represent the candidate's peer anonymously. The candidate is a peer reflexive candidate; the ip and port are a binding allocated by a NAT when it sent a STUN request to represent the candidate's peer anonymously. The candidate is a relay candidate, obtained from a TURN server. The relay candidate's IP address is an address the TURN server uses to forward the media between the two peers. 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 The #GstWebRTCICE a #GstPromise to be notified when the task is complete. 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 An "active" TCP candidate is one for which the transport will attempt to open an outbound connection but will not receive incoming connection requests. A "passive" TCP candidate is one for which the transport will receive incoming connection attempts but not attempt a connection. An "so" candidate is one for which the transport will attempt to open a connection simultaneously with its peer. Value used for non-TCP candidate type. See also https://w3c.github.io/webrtc-pc/#dom-rtcicetransport-getselectedcandidatepair A #GstWebRTCICECandidatePair ICE Transport See also https://w3c.github.io/webrtc-pc/#dom-rtcicetransport-getselectedcandidatepair A #GstWebRTCICECandidatePair ICE Transport A #GstWebRTCICECandidatePair ICE Transport 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