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-24#section-4.1.1
for more information.
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 @channel.
a #GstWebRTCDataChannel
Signal that the data channel reached a low buffered amount. Should only be used by subclasses.
a #GstWebRTCDataChannel
Signal that the data channel was closed. Should only be used by subclasses.
a #GstWebRTCDataChannel
Signal that the data channel had an error. Should only be used by subclasses.
a #GstWebRTCDataChannel
a #GError
Signal that the data channel received a data message. Should only be used by subclasses.
a #GstWebRTCDataChannel
a #GBytes or %NULL
Signal that the data channel received a string message. Should only be used by subclasses.
a #GstWebRTCDataChannel
a string or %NULL
Signal that the data channel was opened. Should only be used by subclasses.
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
a #GstWebRTCDataChannel
a #GBytes or %NULL
a #GstWebRTCDataChannel
a string or %NULL
a #GstWebRTCDataChannel
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>
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
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
for more information.
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
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>
An object to track the receiving aspect of the stream
Mostly matches the WebRTC RTCRtpReceiver interface.
The transport for RTP packets
The transport for RTCP packets without rtcp-mux
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 transport for RTP packets
The transport for RTCP packets without rtcp-mux
Unused
The priority of the stream (Since: 1.20)
Mostly matches the WebRTC RTCRtpTransceiver interface.
Direction of the transceiver.
the mline number this transceiver corresponds to
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.
Indicates whether or not sending and receiving using the paired
#GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
either due to SDP offer/answer
The #GstWebRTCRTPSender object responsible sending data to the
remote peer
The #GstWebRTCRTPReceiver object responsible for receiver data from
the remote peer.
The transceiver's desired direction.
The transceiver's current direction (read-only)
A caps representing the codec preferences (read-only)
Type of media (Since: 1.20)
none
inactive
sendonly
recvonly
sendrecv
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>
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
the string representation of @type or "unknown" when @type is not
recognized.
a #GstWebRTCSDPType