Observing a persistent discrepancy in analytics reporting for our APAC region BYOC trunks. The ‘Talk Time’ metric in the performance dashboard consistently lags behind the actual media stream initiation by approximately 8-12 seconds. Specifically, the SIP 200 OK response is logged correctly by the Genesys Cloud Edge, but the WebRTC media stream start event in the browser console appears significantly earlier than the analytics engine’s recorded start time.
This variance is impacting our carrier billing reconciliation and agent performance metrics. The issue is isolated to the Singapore and Tokyo edge locations, affecting Chrome and Firefox clients using the latest WebRTC softphone version. SIP traces confirm that the SDP negotiation completes successfully, yet the analytics engine seems to rely on a different timestamp source for talk time calculation.
Has anyone encountered similar latency between SIP session establishment and WebRTC media stream reporting? Is there a configuration setting in the outbound routing or trunk profile that aligns the analytics timestamps with the browser’s WebRTC logs? Need to resolve this before our next carrier audit cycle.