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GstWebRTC - GstWebRTCBin: Difference between revisions
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If you require a custom signaler use signaler-obj property. According to the selected signaler different properties will be available. These properties can be accessed using the "signaler::<property>" syntax. The following list details the properties of the different signalers: | If you require a custom signaler use signaler-obj property. According to the selected signaler different properties will be available. These properties can be accessed using the "signaler::<property>" syntax. The following list details the properties of the different signalers: | ||
'''GstOwrSignaler:''' | |||
*'''session-id:''' Session Identification. Default: "ridgerun" | |||
*'''server-url:''' URL Server Connection. Default: "http://localhost:8080" | |||
*'''api-token:''' API Token ID. Default: "RRGstWebRTC" | |||
'''GstPubnubSignaler:''' | |||
*'''publish-key:''' Key to Publish Messages (only set in NULL state). Default: "pub-c-561a7378-fa06-4c50-a331-5c0056d0163c" | |||
*'''subscribe-key:''' Key to Subscribe Messages (only set in NULL state). Default: "pubsub.pubnub.com" | |||
*'''user-channel:''' Username Channel (only set in NULL state). Default: "gstwebrtc" | |||
*'''peer-channel:''' Only Incoming Messages from this peer number will be accepted if set, If it is not set, incoming messages from any peer will be accepted. If start call is true, the offer SDP will be sent to this peer number, so that it is required. (only set in NULL state). Default: "(null)" | |||
*'''origin-url:''' PubNub Signaler Origin URL (only set in NULL state). Default: "pubsub.pubnub.com" | |||
'''GstApprtcSignaler:''' | |||
*'''session-id: Session Identification. Default: "ridgerun" | |||
*'''server-url: URL Server Connection. Default: "http://localhost:8080" | |||
'''Enum "GstWebRtcSignalers"''' Default: 1, '''"GstPubnubSignaler"'''<br> | |||
'''(0): GstOwrSignaler''' - Open WebRTC signaler<br> | |||
'''(1): GstPubnubSignaler''' - Pubnub signaler<br> | |||
'''(2): GstApprtcSignaler''' - AppRTC signaler<br> | |||
===signaler-obj=== | |||
Custom signaler object to use (Must only be called on NULL state). Leave this NULL if you want to use a predefined signaler. | |||
===stun-server=== | |||
STUN Server IP Address: address:port. Default: "webrtc.ridgerun.com:3478" | |||
===turn-server=== | |||
TURN Server IP Address: <pre>'user:password@address:port(?transport=[udp|tcp|tls])'</pre>. Default: <pre>"ridgerun:Gh3tVhVZam3SSqb@webrtc.ridgerun.com:3478:transport=tcp"</pre> | |||
===certificate-pem=== | |||
PEM file name containing the certificate, if PEM file is not set or is not found, autogenerated certificate will be used. 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=== | |||
The SCTP port for data transfer. Range: 0 - 65535 Default: 5000 | |||
===data-channel-id=== | |||
The ID of the data channel stream (random number if 0 is set). Range: 0 - 65535 Default: 13118 | |||
===rtcp-mux=== | |||
Enable Multiplex RTP and RTCP in a single port (Endpoint will reconfigure if needed). Default: true Write only | |||
===ice-trickle=== | |||
Send single ICE candidates when they become available. Default: false | |||
===enable-data=== | |||
Include the negotiation of the WebRTC data channel for this endpoint (required to use the new_data and on_new_data callbacks without data-pads). Default: false Write only | |||
===enable-rtcp-timeout=== | |||
Monitor RTCP Feedback, if there are no messages within 30 seconds we will trigger a signal. Default: false | |||
===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"=== | |||
void user_function (GstElement* object, guint arg0, gchararray arg1, gpointer user_data); | |||
==="on-rtcp-bitrate"=== | |||
void user_function (GstElement* object, guint arg0, guint arg1, guint arg2, guint64 arg3, gpointer user_data); | |||
==="on-rtcp-timeout"=== | |||
void user_function (GstElement* object, gpointer user_data); | |||
==Actions== | |||
==="new-data"=== | |||
gboolean user_function (GstElement* object, gchararray arg0, guint arg1); | |||
|keywords=Gstreamer WebRTC Basics,Plugin Overview,WebRTC Basics,Gstreamer WebRTC Plugin Overview,GstRrWebRTCBin element,GstRrWebRTCBin}} | |keywords=Gstreamer WebRTC Basics,Plugin Overview,WebRTC Basics,Gstreamer WebRTC Plugin Overview,GstRrWebRTCBin element,GstRrWebRTCBin}} | ||