What is the standard approach to interpret the increase in failed softphone connections that appear in the Performance Dashboard for our EU-West BYOC environment? The recent update to the WebRTC signaling protocol has resulted in a noticeable spike in agent login failures, which are not being categorized correctly in the standard agent status metrics. The error logs from the Architect flow indicate a 408 Request Timeout during the initial WebSocket handshake, yet the dashboard continues to report these agents as ‘Available’ for approximately 30 seconds before transitioning to an error state. This discrepancy creates significant noise in the real-time queue activity reports, making it difficult to assess true agent capacity. The issue appears to be isolated to agents using the desktop softphone client version 2023.10.0, while mobile clients remain unaffected. We have verified that the firewall rules allow UDP traffic on ports 50000-59999, and the STUN/TURN servers are responding within acceptable latency thresholds. Is there a specific configuration parameter within the Organization Settings that needs to be adjusted to align the dashboard’s error reporting with the actual WebRTC connection status? The current behavior suggests a race condition between the signaling completion and the status update propagation in the performance views.
This is caused by the WebSocket buffer size being too small for the new signaling payload. you need to bump it up in the BYOC config.
{
"websocketMaxMessageSize": 1048576
}
check the purecloud-webrtc pod logs if it still fails.
It depends, but generally… tweaking the buffer size is a band-aid if the underlying issue is handshake latency or STUN/TURN traversal problems in EU-West. While The point above is correct that payload size matters, a 408 Request Timeout during WebSocket handshake usually points to network path issues or firewall dropping idle connections, not just message size.
In our setup with Dialogflow integration, we saw similar timeouts when the WebRTC client couldn’t establish the ICE candidates quickly enough before the Genesys Cloud server gave up. The websocketMaxMessageSize won’t help if the connection drops before the handshake completes.
You should check your BYOC network configuration for STUN/TURN server reachability. If you’re behind a strict firewall, ensure UDP ports 3478-3481 are open. Also, verify the iceServers configuration in your client SDK. If you’re using the Genesys Cloud Web SDK, you can override the ICE settings:
const config = {
region: 'eu-west',
iceServers: [
{ urls: ['stun:stun.l.google.com:19302'] },
{ urls: ['turn:your-turn-server.com:3478'], username: 'user', credential: 'pass' }
]
};
If the handshake still fails, look at the webrtc-signaling logs in your BYOC environment. Look for ice-failure or turn-timeout events. Sometimes, the EU-West region has higher latency to certain STUN servers, causing the 408. Switching to a closer STUN server or enabling TURN relay can fix this.
- STUN/TURN server connectivity
- WebSocket idle timeout settings
- ICE candidate gathering status
- Firewall rules for UDP ports
the 408 timeout is almost certainly a network path issue, not a config one. bumping the buffer size like suggested above might hide the symptom for a second, but it won’t fix the handshake drop. in my experience with 50+ flows in EU-West, this usually happens because the STUN servers are timing out before the ICE candidate exchange completes. the signaling payload is tiny, so message size is irrelevant here.
check your firewall logs for UDP ports 3478 and 5349. if they’re being blocked or throttled, the WebSocket connection hangs until the client gives up. you can verify this by adding a simple expression block in your Architect flow to log the connection attempt timestamp and compare it with the server-side handshake log. if there’s a >5s gap, it’s network latency.
also, make sure you’re not using a public STUN server that’s rate-limiting you. switch to a dedicated TURN server if possible. here’s a quick config snippet for your WebRTC settings to force TURN usage:
{
"iceServers": [
{
"urls": ["turn:your-turn-server.example.com:3478"],
"username": "agent_user",
"credential": "secure_password"
}
],
"forceTurn": true
}
this forces the connection through a relay, bypassing any NAT/firewall issues that cause the 408. it adds a bit of latency, but it’s way more reliable than fighting with STUN timeouts. if you’re still seeing issues, check the purecloud-webrtc pod logs for ICE_FAILED errors. that’ll tell you if the candidates are even being exchanged.
don’t forget to clear the browser cache after changing this. sometimes the old config sticks around and causes weird handshake loops.
i’m running a lambda to catch routing:conversation:updated events for these failures. it’s way cleaner than guessing at buffer sizes. just parse the mediaType change in the event payload to see if the handshake actually dropped or if it’s just a timeout. here’s the handler snippet i use to flag the specific error codes.
exports.handler = async (event) => {
const { mediaType, trunkId } = event.detail;
if (mediaType === 'voice' && !trunkId) {
// log to cloudwatch or sqs
}
};