How to fix unreliable WebRTC calling?

[Nail Shakirov]

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WebRTC calls are not reliable in our application. Sometimes we see the black screen, sometimes we don’t see call start at all and sometimes there are seen huge delays or de-sync in audio/video. Setup: Google’s public STUN server stun:stun.l.google.com:19302; TURN server is Coturn hosted in azure in docker; Signaling server is a customized https://github.com/andyet/signalmaster that using express as a web server On client JS is used, with simplewebrtc as client; For iOS support used Cordova plugin - https://github.com/eface2face/cordova-plugin-iosrtc Almost 100% reproduced issue is calling from one client on LTE to another on Wi-Fi. In this case we see black screen on both devices, however, default bg-color is white, so at least something happens on WebRTC side. What was done to address issues: Examined Coturn logs... Sometimes we see "Unauthorized" errors there, but it's hard to say if they affect anything; Checked Coturn's traffic: in Wi-Fi to Wi-Fi scenarios it is low, so peer-to-peer connection is really made. If there is LTE, we see around 40–120KiB/sec load (Isn't that too low for audio/video?), so TURN seems to work; Checked client app logs, nothing special; Please, suggest any possible way of research or fix to make WebRTC as much reliable as possible.

[shakeeb nazmus]

1. Can you please repeat your test for an audio only call?  

2. Is it possible to use a turn server which is near( hop count distance is less and latency is less) to your WiFi network?

3. Use full cone NAT wifi router and avoid double natting to ensure P2P call. Does the issue happen with the P2P call? if not then the issue is related to turn server. 

The best option will be "enable log" in the client side and investigate the log to pinpoint the issue. I have assumed that you have no issue with signaling as your description indicates. 

Thanks,

Shakeeb