What is the correct way to handle WebRTC handshake failures in apac-1 via CLI?

What is the standard approach to handle WebRTC handshake failures when deploying softphone settings via GC CLI?

Environment: apac-1, GC CLI 2.2.0. The deploy script completes without error, but agents report ‘Connection Refused’ on initial login. No logs in the console.

The issue only affects new users provisioned through our GitHub Actions pipeline. Manual config works fine.

Checking the network, ports seem open. Any known CLI flags to force a handshake retry or validate the STUN config during the apply phase?

Have you tried validating the STUN server configuration in your CLI payload? In Zendesk, network settings were often implicit, but Genesys Cloud requires explicit STUN server definitions for WebRTC in apac-1. Check if your script includes the correct region-specific STUN servers.

"stunServers": ["stun.apac-1.pure.cloud:3478"]

Missing this usually causes the handshake timeout for new users.