GstWebRTC - AppRTC Audio Examples - x86

GstWebRTC
WebRTC Fundamentals
GstWebRTC Basics
Signaling 1.Predefined Signaling 2.Custom Signaling 3.API Reference 4.Advance Settings Plugin Elements 1.GstWebRTCBin 2.GstWebRTCBin Properties 3.GstWebRTCBin Features 1.Data Channel
Evaluating GstWebRTC
Getting the code
Building GstWebRTC
x86 Platforms Jetson Platforms: TX1, TX2 and Xavier Yocto Platforms: i.MX6 and i.MX8
Examples
x86 OpenWebRTC PubNub AppRTC MediaSoup iMX6 OpenWebRTC PubNub AppRTC iMX8 OpenWebRTC PubNub AppRTC TX1/TX2/Xavier OpenWebRTC PubNub AppRTC
MCU Demo Application
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This page presents some GstRrWebRTC audio examples on x86 platform using AppRTC.

Problems running the pipelines shown on this page? Please see our GStreamer Debugging guide for help.

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:

Setup 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