Can anyone clarify the expected behavior for WebRTC softphone registration stability when leveraging BYOC trunks in the Singapore region? We are managing 15 distinct BYOC trunks for our APAC operations, and while SIP trunk registration remains stable, agent softphones are experiencing intermittent 408 Request Timeout errors during peak load hours.
The environment utilizes Genesys Cloud version 2023.4.0. The SIP credentials are verified as valid, and outbound routing logic is correctly mapped to the primary carrier. However, the failover logic does not seem to trigger for these specific softphone sessions, leading to a degraded experience for agents attempting to initiate outbound calls. The 408 Request Timeout appears to originate from the initial WebSocket handshake rather than the SIP INVITE stage.
We have reviewed the analytics reporting for the affected trunks and noticed no significant jitter or packet loss that would justify these timeouts. The issue seems isolated to the WebRTC client’s ability to maintain a persistent connection with the media servers when the backend is routing through our specific carrier configurations. Is there a known limitation with BYOC trunk concurrency that impacts softphone registration timeouts, or should we be adjusting the keep-alive intervals in the Architect flow?
check your websocket connection settings in the agent ui, since zendesk didn’t handle real-time media streams this way. the 408s usually mean the browser is dropping the keep-alive packets before the sip stack notices, so forcing a shorter keepalive interval in the softphone config might stabilize it.
It varies, but usually the 408 errors in this context are less about the SIP trunk itself and more about how the WebRTC client handles the WebSocket keep-alive under load. Since we are looking at APAC BYOC trunks, latency plays a bigger role here. The suggestion above about keep-alive intervals is correct, but there is a specific configuration tweak that helps stabilize the connection during high concurrency tests.
When running load tests with JMeter or similar tools to simulate peak hours, the browser’s WebSocket connections can drop if the ping/pong interval is too aggressive or too lax. Try adjusting the following settings in your agent desktop configuration or via the Admin API if you are provisioning these agents in bulk:
- WebSocket Ping Interval: Set this to
25000 ms (25 seconds). The default is often 30 seconds, which can be too long for high-latency APAC routes.
- WebSocket Ping Timeout: Set this to
5000 ms (5 seconds). This ensures the client reacts quickly to a missed pong.
- Jitter Buffer: Increase the initial jitter buffer to
200 ms. This helps absorb the packet loss that often accompanies the timeout errors.
Here is a sample JSON payload for updating the agent settings via the Admin API:
{
"settings": {
"webRtc": {
"pingInterval": 25000,
"pingTimeout": 5000,
"jitterBufferInitial": 200
}
}
}
We saw similar 408 timeouts in our EST environment when testing 500 concurrent sessions. Reducing the ping interval helped the system detect dropped connections faster and re-establish them before the SIP stack timed out. Make sure to monitor the WebSocket reconnection rate after applying these changes. If the 408s persist, check the browser console for any certificate errors or CORS issues that might be blocking the WebSocket handshake.