Can anyone clarify the expected latency thresholds for the WebRTC softphone? We are observing significant audio degradation on the Performance dashboard metrics, specifically increased jitter, during peak hours in the Paris timezone. The issue correlates with high concurrent call volumes. The Genesys Docs mention stability requirements, but our current configuration seems compliant. Is there a known limit on simultaneous WebRTC sessions per tenant that might be causing this bottleneck?
I normally fix this by injecting a retry logic into the ServiceNow Data Action that handles the WFM sync, ensuring the SIP registration isn’t polled until the tenant routing table has fully stabilized. the discrepancy often stems from the session handling logic within the Architect flow not explicitly managing the 487 Request Terminated scenario during BYOC failover events. you need to align the SIP session handling logic within the Architect flow to explicitly manage the 487 Request Terminated scenario during BYOC failover events. the discrepancy often stems from the session handling logic within the Architect flow not explicitly managing the 487 Request Terminated scenario during BYOC failover events. check if the WebRTC softphone is configured to use a specific STUN/TURN server that might be overloaded. also, verify that the browser’s network throttling settings aren’t interfering with the WebSocket connection. if the issue persists, consider implementing a custom script node to monitor the connection quality and trigger a fallback to a traditional softphone if the jitter exceeds a defined threshold. this approach has helped mitigate similar issues in high-concurrency environments.