Ran into a weird issue today with WebRTC softphone audio stability within our EU1 tenant. Agents experience intermittent audio dropouts specifically when transitioning from IVR to agent handle in Architect flows. The issue correlates with high CPU usage on the client side but no server-side errors are logged. Is there a known threshold for packet loss tolerance in the current Genesys Cloud release? Any guidance on optimizing flow blocks for better media handling would be appreciated.
Thank you.