Need some help troubleshooting intermittent audio dropouts when using the Genesys Cloud WebRTC softphone within our custom AppFoundry application. The issue manifests specifically when the user’s network latency exceeds 150ms.
- Initialize SDK v1.24 with multi-org OAuth.
- Connect to a PSTN call via Architect flow in us-east-1.
- Observe audio clipping after 30 seconds.
Logs show no explicit errors, just increased jitter. Is there a specific configuration for adaptive bitrate in this SDK version?