Trying to understand why our Genesys Cloud WebRTC softphone clients are experiencing intermittent audio dropouts specifically during the 10:00 AM CST shift change window. The issue affects roughly 15% of agents logging in for the mid-day shift. We are using the latest Genesys Cloud desktop client version 5.12.0.
The dropouts manifest as 200-500ms silence gaps, not complete disconnections. Network traces show stable jitter and packet loss (<1%) for all affected agents, ruling out local connectivity issues. The problem seems isolated to the WebRTC stream initialization phase. When agents log in, the softphone connects, but the audio channel fails to stabilize until a manual ‘Refresh’ or re-login is forced.
We have verified that the WFM schedule publishing is successful and agent statuses update correctly in real-time. However, the latency between status change and full media readiness appears to be the culprit. Is there a known configuration in the Architect or Org settings that delays WebRTC media negotiation for agents transitioning from ‘Not Ready’ to ‘Available’ states? We suspect a race condition between the WFM status update and the softphone media handshake.