WebRTC Softphone Latency Spikes in EU-FR Region During Peak Hours

Voice & Telephony
1

Struggling to figure out why the WebRTC softphone interface is experiencing significant audio latency and occasional dropouts specifically within the EU-FR environment during our peak operational hours (09:00 - 11:00 CET). The standard web softphone is being utilized by a cohort of approximately 50 agents who are handling high-volume inbound calls routed through a complex Architect flow involving IVR and skill-based routing.

The issue is not universal; it appears to be correlated with specific network conditions or tenant load. When the latency exceeds 200ms, the conversation quality degrades to the point where agents are unable to effectively assist customers, leading to increased handle times and customer dissatisfaction. We have verified that the agents’ local network connections are stable and meet the recommended bandwidth requirements.

The error logs from the browser console show intermittent ICE connection failures and re-negotiation events. A sample of the error captured during a recent incident is provided below:

WebSocket connection to 'wss://api.eu-fr-1.genesis.com/v1/websocket' failed: ICE_FAILED
Error: Connection terminated with error code ICE_FAILED after 30 seconds of active conversation.

We are currently using the latest version of the Genesys Cloud web client. The Architect flow itself is functioning correctly, as calls are being answered and routed as expected, but the media path seems to be compromised. We have checked the Network Dashboard and do not see any widespread outages or degradation for the EU-FR region.

Is there a known issue with WebRTC media routing in the EU-FR region that could cause this behavior? Are there specific metrics in the Performance dashboard or Conversation Detail views that we should be monitoring to diagnose this issue further? We need to understand if this is a tenant-specific configuration issue or a broader platform problem affecting our region.

2

Check the WebRTC configuration in Admin > Communications > WebRTC. The latency spikes in EU-FR often stem from default STUN/TURN server selections that are suboptimal for local traffic. Switch to the regional TURN servers explicitly.

{
 "stunServers": ["stun:stun.l.google.com:19302"],
 "turnServers": [
 {
 "urls": ["turn:eu-fr-turn.genesys.cloud:443?transport=udp"],
 "username": "agent_user",
 "credential": "turn_secret"
 }
 ]
}

In Zendesk, we relied on their global CDN for audio, so this level of granular control wasn’t necessary. Genesys Cloud requires you to pin agents to the correct edge for low latency. Also, ensure the “Use TURN for all media” toggle is OFF unless NAT traversal fails. For 50 agents, the default STUN should suffice if the network allows UDP. If latency persists, verify the Architect flow isn’t adding unnecessary recording steps that buffer audio before playback. This is a common migration gotcha when moving from Zendesk’s simpler voice setup.