6.4 KiB
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-24section-4.1.1
for more information.
Feature: v1_16
GST_WEBRTC_DTLS_SETUP_NONE: none GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
Implements
GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
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</ulink>
Feature: v1_16
none
ulpfec + red
Feature: v1_14_1
GST_WEBRTC_ICE_COMPONENT_RTP, GST_WEBRTC_ICE_COMPONENT_RTCP,
GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
See http://w3c.github.io/webrtc-pc/dom-rtciceconnectionstate</ulink>
GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
See http://w3c.github.io/webrtc-pc/dom-rtcicegatheringstate</ulink>
GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
Implements
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24section-4.1.1
for more information.
Feature: v1_16
GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
See http://w3c.github.io/webrtc-pc/dom-rtcpeerconnectionstate</ulink>
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</ulink>
Feature: v1_16
Implements
Implements
Implements
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</ulink>
Feature: v1_16
GST_WEBRTC_SDP_TYPE_OFFER: offer
GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
GST_WEBRTC_SDP_TYPE_ANSWER: answer
GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
See http://w3c.github.io/webrtc-pc/rtcsdptype``</ulink>
See https://www.w3.org/TR/webrtc/rtcsessiondescription-class</ulink>
type_
a WebRTCSDPType
sdp
a gst_sdp::SDPMessage
Returns
a new WebRTCSessionDescription from type_
and sdp
Returns
a new copy of self
Free self and all associated resources
GST_WEBRTC_SIGNALING_STATE_STABLE: stable
GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
See http://w3c.github.io/webrtc-pc/dom-rtcsignalingstate</ulink>
GST_WEBRTC_STATS_CODEC: codec GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp GST_WEBRTC_STATS_CSRC: csrc GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion GST_WEBRTC_STATS_DATA_CHANNEL: data-channel GST_WEBRTC_STATS_STREAM: stream GST_WEBRTC_STATS_TRANSPORT: transport GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate GST_WEBRTC_STATS_CERTIFICATE: certificate