Does anyone know why Genesys Cloud WebRTC returns a 403 Forbidden when failing over to secondary BYOC trunks in AP1?
Background
Running 15 BYOC trunks via AWS SIP Media Application. Primary trunks handle ~80% volume; secondary trunks are configured for failover in Architect.
Issue
When primary trunks hit capacity, calls routing to secondary trunks fail immediately with SIP 403 in CDRs. Direct dial to secondary trunks works fine.
Troubleshooting
Verified SIP credentials and outbound routing rules. Checked AWS logs; no rejections from carrier side. Issue isolated to Genesys Cloud failover logic.
The quickest way to solve this is to verify that the secondary BYOC trunk configuration explicitly inherits the correct WebRTC security policies from the primary trunk, rather than relying on default regional settings which often differ between AP1 zones. In my experience with automated ticketing workflows, misaligned security contexts are the primary cause of 403 errors during failover events. The secondary trunk must have identical WebRTC domain configurations and certificate bindings as the primary. Check the Trunk Settings in Admin to ensure the ‘WebRTC Enabled’ toggle is active on the secondary SIP trunk and that the associated WebRTC domain matches the primary exactly.
Next, inspect the Data Actions or Webhook payload structure if you are using automated routing logic. A common oversight is that the failover trigger might not pass the correct conversationId or mediaType headers required by the secondary trunk’s WebRTC endpoint. Ensure your Architect flow does not strip these headers during the failover transition. You can validate this by capturing the raw SIP INVITE and comparing the headers against the primary trunk’s successful calls. If the secondary trunk is configured with a different WebRTC domain, the 403 error will persist regardless of network connectivity.
Finally, cross-reference the Genesys Cloud documentation for WebRTC security restrictions in AP1. There are known latency issues with certificate validation in this region that can cause timeouts interpreted as 403 Forbidden errors. Implementing a retry mechanism in your failover logic or adjusting the timeout thresholds in the trunk settings can mitigate this. If the issue persists, enable detailed SIP debugging on the secondary trunk and compare the TLS handshake logs with the primary trunk. This usually reveals whether the issue is a certificate mismatch or a permission scope error in the WebRTC domain configuration.