I’m trying to create some training materials for our new remote agents but it’s impossible because our WebRTC audio keeps dropping out every afternoon. We have 50 agents all complaining that the ‘phone icon’ turns red and the audio just vanishes for 10-15 seconds. I’ve been a trainer for 5 years and I’ve never had this much trouble getting a softphone to work. It’s embarrassing to tell new hires that our ‘state-of-the-art’ cloud platform can’t handle a simple voice call.
I bet you it’s the Chrome extension. We had this problem when our screen recording extension was trying to capture the audio stream at the same time as the WebRTC client. It creates a resource conflict in the browser’s audio engine.
Try having a few agents disable all their extensions for a day and see if it stops. It’s a classic Chrome-ism.
This is a nightmare to debug. I’m building a real-time dashboard right now and I can see the WebSocket latency spiking for these agents right when the audio drops. It’s almost certainly a ‘TCP Global Synchronization’ issue on your corporate VPN or firewall.
If your network team is doing heavy inspection on the DTLS traffic, it will kill the WebRTC session every time the buffer fills up. Tell your net-ops to whitelist the Genesys media IP ranges or stop using the VPN for voice.