GstWebRTC - GstWebRTCBin Properties
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| GstWebRTC | ||||||||
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| WebRTC Fundamentals | ||||||||
| GstWebRTC Basics | ||||||||
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| Evaluating GstWebRTC | ||||||||
| Getting the code | ||||||||
| Building GstWebRTC | ||||||||
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Properties
name
The name of the object. For this property "webrtcbin0" is the default value.
parent
The parent of the object.
async-handling
The bin will handle Asynchronous state changes. Default: false
message-forward
Forwards all children messages. Default: true
signaler
If a predefined signaler is going to be used, this property is in charge of setting the desired predefined signaler to use in the application. Each signaler has its own properties. See Predefined Signalers.
signaler-obj
This property is used to set a custom signaler. See [Signalers] The custom signaler object to use (Must only be called on NULL state). Leave it NULL if you want to use a predefined signaler.
stun-server
Property used to set the STUN server address and port in case you want to change to use your own or a different one. Usage:
<stun-server-address>:<stun-server-port>
STUN Server IP Address:
address:port
Default:
"webrtc.ridgerun.com:3478"
turn-server
Property used to set the TURN server address and port in case you want to change to use your own or a different one. Usage:
<user>:<password>@<turn-server-address>:<stun-server-port>
TURN Server IP Address:
'user:password@address:port(?transport=[udp|tcp|tls])'
Default:
"ridgerun:Gh3tVhVZam3SSqb@webrtc.ridgerun.com:3478:transport=tcp"
certificate-pem
PEM file name containing the certificate, if PEM file is not set or is not found, autogenerated certificate will be used. With this property you are able to set you own certificate and private key (in PEM format) for the DTLS negotiation. Default: null
start-call
It's used to set which endpoint responsible of starting the call and sending the initial offer SDP to the peer. It only makes sense to have this property set to TRUE in one of the endpoints. Default value: false
data-port
This property is used to specify the desired SCTP port to use for data transfer. Its value must be defined in the range: [0 - 65535]. Default: 5000.
data-channel-id
It is used to indicate an identifier for each data channel stream. Default: A random number in the range [0 - 65535].
rtcp-mux
Enable Multiplex RTP and RTCP in a single port (Endpoint will reconfigure if needed). Default: true Write only
ice-trickle
This property is used to enable the sending of a single ICE candidates group when they become available. Default value: false
enable-data
Used to enable the negotiation of the WebRTC data channel for an endpoint (required to use the new_data and on_new_data callbacks without data-pads). Default: false
enable-rtcp-timeout
This property enables Monitor RTCP Feedback, if there are no messages within 30 seconds we will trigger a signal. Default: false
candidates-filter
Remove desired ICE candidates in the negotiation by entering candidate types separated by commas. Filters available: srflx, prflx, relay, host, UDP, TCP, ipv4, ipv6. Default: null
rtp-stats-interval
Interval in milliseconds to get RTP Stats as Gstreamer Messages. Range: 50 - 4294967295 Default: 1000
data-channel-status
Connection status indicating if the data channel is currently connected. Default: false
Signals
"on-new-data"
This signal is emitted once a data has arrived on data channel.
"on-rtcp-bitrate"
This signal is emitted when a REMB RTCP packet has arrived.
"on-rtcp-timeout"
This signal is emitted when it stops receiving RTCP feedback.
Actions
"new-data"
Action to call to send new data