GST_WEBRTC_DTLS_SETUP_NONE: none
GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
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_FEC_TYPE_NONE: none
GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red
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 <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">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 <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink>
GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
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 <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink>
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 <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink>
the string representation of @type or "unknown" when @type is not
recognized.
a #GstWebRTCSDPType
See <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
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
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 <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">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
the string representation of @type or "unknown" when @type is not
recognized.
a #GstWebRTCSDPType