Hi,
We are new to WebRTC and have several questions to ask, thanks a lot for helping!
1. What’s the best practice to monitor WebRTC quality?
We are monitoring these params to check whether we are in good condition currently.
inboundRTP:
fractionLost > 0.1
jitter > 0.03
roundTripTime > 1
outboundRTP:
framesPerSecond < 20
targetBitrate < 1000000
qualityLimitationReason != “none”
2. How do we control/reduce/monitor the WebRTC delays?
Sometimes we will see lag for video/audio for remote person, what’s the best way to debug on iOS to check WebRTC delays? bandwidth may affect the video quality and will it causes seconds delay?
Additionally, how do we recover from delays to ensure that both users see/hear everything that each other says/does while keeping them in sync?
3. We are using Mediasoup framework to wrap WebRTC, we don’t need to have STUN, but do we need to use TURN? Will use TURN increase our connection success rate?
There’s a small chance that transport is stuck at checking state, does this mean we need to use TURN
For this checking state, what’s the best way to restart if this happens? Will restartICE help? Should I create a new room to get all new params from the server to connect?
5. For WebRTC default AudioDevice on iOS, what’s the structure of that Device? How WebRTC control the noise and audio processing?
Here is our custom device, there are two problems with our custom audio device
Problems:
There’s some extra noise compared with the default WebRTC audio device module
The audio will be processed like a Robot