Can anyone clarify the specific sequence required for WebRTC softphone initialization when migrating from the Zendesk Talk plugin to the Genesys Cloud Engage SDK? We are currently transitioning our digital channel agents and have hit a wall with the softphone connectivity. In Zendesk, the audio driver was largely abstracted within the widget, but here we need explicit permission handling in the browser. The client-side logs show a successful register call to /api/v2/fax/registrations, but the subsequent WebRTC offer/answer exchange times out with a 408 Request Timeout from the media relay servers. This happens consistently on Chrome 120+ in the EU-West region. The Architect flow is set to Voice with WebRTC preferred, yet the agent interface remains in a Connecting state indefinitely. We have verified that the firewall allows UDP ports 50000-60000, mirroring the open policy we had for Zendesk’s STUN/TURN servers. The error payload in the browser console references ICE candidate gathering failed, which suggests a NAT traversal issue rather than an authentication failure. Given our background in Zendesk’s simpler connectivity model, the explicit STUN server configuration in Genesys feels like a steep learning curve. Are there specific DNS requirements or STUN server URLs that must be whitelisted explicitly for the Engage SDK to function correctly, or is this a known issue with the current SDK version 2.14.0?