Tester le WebRTC avec AppRTC

The video chat demo app based on WebRTC


AppRTC parameters (https://apprtc.appspot.com/params.html)

A number of settings for the AppRTC video chat application can be changed by adding URL parameters.

For example: https://appr.tc/?hd=true&stereo=true&debug=loopback

The file using the parameters is apprtc.py. More Google-specific parameters are available from the MediaConstraints interface.

For more information see AppRTC : Google's WebRTC test app and its parameters.

hd=trueUse HD camera resolution constraints, i.e. minWidth: 1280, minHeight: 720
stereo=true&audio=echoCancellation=falseTurn on stereo audio
debug=loopbackConnect to yourself, e.g. to test firewalls
ts=[turnserver]Set TURN server different from the default
audio=true&video=falseAudio only
audio=falseVideo only
audio=echoCancellation=falseDisable all audio processing
audio=googEchoCancellation=falseDisable echo cancellation
audio=googAutoGainControl=falseDisable gain control
audio=googNoiseReduction=falseDisable noise reduction
asc=ISAC/16000Set preferred audio send codec to be ISAC at 16kHz (use on Android)
arc=opus/48000Set preferred audio receive codec Opus at 48kHz
vsc=VP8Set preferred video send codec to VP8
vrc=H264Set preferred video receive codec to H264
dscp=trueEnable DSCP
ipv6=trueEnable IPv6
arbr=[bitrate]Set audio receive bitrate, kbps
asbr=[bitrate]Set audio send bitrate
vsbr=[bitrate]Set video receive bitrate
vrbr=[bitrate]Set video send bitrate
videofec=falseTurn off video FEC
opusfec=falseTurn off Opus FEC
opusdtx=trueTurn on Opus DTX
opusmaxpbr=8000Set the maximum sample rate that the receiver can operate, for optimal Opus encoding performance

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