Why does this setting in the Admin > Telephony > Telephony providers configuration appear to cause a discrepancy in the ‘Active Talk Time’ metric for agents using the WebRTC softphone in our EU-West BYOC environment?
We have recently updated our Architect flow (v6.1) to include a new digital-to-voice handoff sequence. Since the deployment, agents using the WebRTC softphone are reporting intermittent connection drops during the initial media stream establishment. The issue is not present for agents using desk phones or the desktop application.
“WebRTC connections are established directly between the agent’s browser and the Genesys Cloud media servers. If the connection fails to establish within the timeout period, the call is dropped and no media is exchanged.”
The specific error observed in the browser console is RTCError: connection-failure with a timeout value of 15 seconds. However, the Agent Performance dashboard continues to register these calls as ‘Handled’ with an ‘Active Talk Time’ of 0 seconds, but the ‘Wrap-up Time’ is inflated by the duration of the timeout.
This is causing a significant skew in our Service Level calculations, as the calls are counted as handled but with zero talk time, artificially inflating the average handle time when wrap-up is included. We are using Genesys Cloud version 2023-12 GA. The issue is reproducible in Chrome 119 and Edge 119.
We have verified that the network conditions are stable and that the WebRTC settings in the Admin portal are configured correctly. The issue seems to be related to the timeout setting in the Telephony provider configuration, but we are unsure how to adjust it without affecting other aspects of the system.
Can anyone provide guidance on how to properly configure the WebRTC timeout settings to prevent this discrepancy in the Agent Performance metrics? We need to ensure that the ‘Active Talk Time’ and ‘Wrap-up Time’ metrics accurately reflect the actual call handling process.