Difference between revisions of "GstWebRTC - AppRTC Audio Examples - x86"
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| − | |keywords=GstWebRTC Examples,WebRTC Examples,GstWebRTC GStreamer pipelines,WebRTC GStreamer pipelines,AppRTC signaler,AppRTC Node Server, websocket server, AppRTC Signaler Example, AppRTC Audio Example, GstWebRTC AppRTC gstreamer Audio pipeline,GstWebRTC audio example}} | + | |keywords=GstWebRTC Examples,WebRTC Examples,GstWebRTC GStreamer pipelines,WebRTC GStreamer pipelines,AppRTC signaler,AppRTC Node Server, websocket server, AppRTC Signaler Example, AppRTC Audio Example, GstWebRTC AppRTC gstreamer Audio pipeline,GstWebRTC audio example,signaling}} |
Revision as of 13:30, 14 May 2019
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This page presents some GstRrWebRTC audio examples on x86 platform using AppRTC.
Server Setup
To run the examples first enable the Websocket server:
$GOPATH/bin/collidermain -port=8089 -tls=false
Then, enable the AppRTC Node server in a different terminal window:
cd <PATH>/apprtc-node-server
node ./bin/www
Note: Make sure you previously install dependencies needed for enable the servers, if you didn't follow this link [AppRTC Node Server with our websocket server]
Opus
Unidirectional elements
Example
In this example, we use rrwebrtcbin to send an audio stream and rrwebrtcbin to receive the audio stream.
Send Pipeline
gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \
signaler::session_id=1234ridgerun name=web audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink
Receive Pipeline
gst-launch-1.0 rrwebrtcbin start-call=false signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \
signaler::session_id=1234ridgerun name=web web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink
Bidirectional elements
Example
In this example we use two rrwebrtcbins, each send an audio stream and receives each other audio stream.
Send-Receive Pipeline
gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \
signaler::session_id=1234ridgerun name=web web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink
Send-Receive Pipeline
gst-launch-1.0 rrwebrtcbin start-call=false signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \
signaler::session_id=1234ridgerun name=web web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink
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