Tester le WebRTC avec AppRTC

Modifié le Lun, 12 Août, 2019 à 3:45 H

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.

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

hd=true	Use HD camera resolution constraints, i.e. minWidth: 1280, minHeight: 720
stereo=true&audio=echoCancellation=false	Turn on stereo audio
debug=loopback	Connect to yourself, e.g. to test firewalls
ts=[turnserver]	Set TURN server different from the default
audio=true&video=false	Audio only
audio=false	Video only
audio=echoCancellation=false	Disable all audio processing
audio=googEchoCancellation=false	Disable echo cancellation
audio=googAutoGainControl=false	Disable gain control
audio=googNoiseReduction=false	Disable noise reduction
asc=ISAC/16000	Set preferred audio send codec to be ISAC at 16kHz (use on Android)
arc=opus/48000	Set preferred audio receive codec Opus at 48kHz
vsc=VP8	Set preferred video send codec to VP8
vrc=H264	Set preferred video receive codec to H264
dscp=true	Enable DSCP
ipv6=true	Enable 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=false	Turn off video FEC
opusfec=false	Turn off Opus FEC
opusdtx=true	Turn on Opus DTX
opusmaxpbr=8000	Set the maximum sample rate that the receiver can operate, for optimal Opus encoding performance