9.2 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
none
actpass
sendonly
recvonly
Implements
new
closed
failed
connecting
connected
Feature: v1_18
Implements
Close the self.
Feature: v1_18
Signal that the data channel reached a low buffered amount. Should only be used by subclasses.
Feature: v1_18
Signal that the data channel was closed. Should only be used by subclasses.
Feature: v1_18
Signal that the data channel had an error. Should only be used by subclasses.
Feature: v1_18
error
a glib::Error
Signal that the data channel received a data message. Should only be used by subclasses.
Feature: v1_18
data
a glib::Bytes or None
Signal that the data channel received a string message. Should only be used by subclasses.
Feature: v1_18
str
a string or None
Signal that the data channel was opened. Should only be used by subclasses.
Feature: v1_18
Send data as a data message over self.
Feature: v1_18
data
a glib::Bytes or None
Send str as a string message over self.
Feature: v1_18
str
a string or None
Close the data channel
error
the glib::Error thrown
data
a glib::Bytes of the data received
data
the data received as a string
data
a glib::Bytes with the data
data
the data to send as a string
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
Feature: v1_16
none
ulpfec + red
Feature: v1_14_1
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
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
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
Feature: v1_16
Implements
Implements
Implements
Direction of the transceiver.
Feature: v1_18
Direction of the transceiver.
Feature: v1_18
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
Feature: v1_16
See http://w3c.github.io/webrtc-pc/`rtcsdptype`
offer
pranswer
answer
rollback
See https://www.w3.org/TR/webrtc/`rtcsessiondescription`-class
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
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