Has anyone hit a wall with WebRTC softphone initialization timing out when migrating from Zendesk Talk to Genesys Cloud? We are in the final stages of moving a client over, and while the digital channels (chat/email) are mapped perfectly, the voice component is giving us headaches.
In Zendesk, the agent interface just loaded seamlessly. Now, with Genesys Cloud, our agents are seeing a 408 Request Timeout error when the softphone widget tries to register with the signaling server. This happens specifically during the first login after a browser refresh.
Here are the details:
- Environment: Genesys Cloud EU West (Frankfurt)
- SDK: Genesys Cloud Web SDK v1.2.4
- Error:
WebSocket connection failed: 408 Request Timeout in the browser console.
- Architect: The flow is simple-just a standard IVR routing to a skill-based group. No complex logic yet.
We have checked the firewall rules, and ports 443 and 3478 are open for STUN/TURN. The issue seems intermittent but frequent enough to block our UAT sign-off. I am used to Zendesk’s simpler setup where you just plug in the widget, but GC’s WebRTC requirements feel much stricter regarding network stability.
Is there a specific configuration in the Organization Settings or Architect flow that needs to be adjusted for WebRTC stability? Or is this a known issue with the current SDK version? Any tips on debugging the WebSocket handshake would be greatly appreciated. We want to ensure the migration doesn’t leave agents stuck in a loop of reconnection attempts.
Ah, I hit this last week when we were shifting traffic from a legacy Zendesk setup to our APAC BYOC trunks. The 408 isn’t usually a Genesys Cloud internal timeout; it’s almost always the WebRTC client failing to complete the SIP INVITE handshake with your specific carrier edge before the browser drops the WebSocket connection.
Since you are migrating from Zendesk, check your outbound routing logic. Genesys expects a stricter SIP registration state for WebRTC than Zendesk’s proprietary stack. If your agents are assigned to a trunk group that has aggressive failover or if the carrier requires specific authentication headers that aren’t being passed through the WebRTC signaling, the timeout triggers.
Try this config adjustment on your BYOC trunk profile:
- Disable “Early Media” if it is enabled. WebRTC clients often choke on early media responses during the initial handshake.
- Increase the
SIP_TIMEOUT in your custom domain settings if you have access to edge configuration, but realistically, you need to fix the signaling path.
Also, verify that your agents’ network allows UDP ports 50000-59999. Genesys WebRTC uses STUN/TURN heavily. If your corporate firewall is blocking the initial ICE candidate exchange, the client waits until it hits the 408.
We resolved this by forcing our agents to use the TURN server explicitly in the softphone settings, bypassing the STUN discovery which was timing out against our carrier’s NAT. Check your browser console for iceconnectionstate failures. If you see them, it’s a network path issue, not a Genesys bug.