WebRTC Audio Degradation on AP-Southeast-1 BYOC Trunks

Can anyone clarify the impact of carrier-specific jitter buffers on WebRTC softphone clients? Our 15 BYOC trunks in AP-Southeast-1 are showing 400ms+ jitter spikes during peak hours. The issue correlates with packet loss visible in the Genesys Cloud analytics dashboard. We are using SDK v2.1.0 for our custom softphone integration. Is there a specific configuration to mitigate this without increasing latency?