WebRTC softphone drops during chat-to-voice transfer

The softphone drops every time a chat transfer triggers a voice flow. Browser console shows WebRTC negotiation failed: ice_gathering_timeout on Genesys Web SDK 3.12.4. We’ve tried the STUN server workaround from the routing thread, but the mic stays hot and audio cuts out anyway. Environment details:

  • Chrome 124 on macOS
  • Architect flow version 18
  • Queue capacity set to voice only
    The STUN fix didn’t hold past the handoff.