WebRTC Softphone Latency Spikes During Peak Schedule Publishing

Voice & Telephony
1

Hey everyone, I’ve run into a really strange issue with our WebRTC softphone performance right now. We are using the standard Genesys Cloud Web SDK v3.12.5 in our custom agent desktop wrapper. Every time the WFM service pushes the final schedule publication for the week (usually around 4 PM CT on Fridays), we see a massive spike in audio latency for agents logged in via the softphone. The latency jumps from a healthy 120ms to over 600ms, causing significant talk-over issues during live calls. This only happens when the schedule sync is actively writing to the database, suggesting a resource contention issue between the WFM API threads and the media processing nodes.

We have verified that the network path is stable and that the issue correlates perfectly with the POST /api/v2/wfm/schedules/publications endpoint execution. Disabling the softphone and switching to PSTN routing eliminates the lag, which points to a client-side or edge media handling bottleneck rather than a general network outage. Has anyone seen this specific interaction between WFM schedule publishing and WebRTC media streams? Are there any known mitigations or SDK flags to prioritize media traffic during high-WFM-load windows?

2

Check your JMeter script configuration for how you are handling the WebSocket connection lifecycle during the schedule publication window. The latency spike isn’t actually a WebRTC audio issue; it is a thread contention problem in your load test setup. When the WFM API pushes the schedule update, it triggers a burst of configuration events. If your JMeter script is trying to reconnect or validate the WebSocket session simultaneously across 200+ threads, you are creating a bottleneck that mimics high latency.

The standard Web SDK handles the schedule update gracefully, but under heavy concurrent load, the local event loop gets blocked. You need to stagger the schedule polling. Instead of having all threads hit the /api/v2/wfm/schedules endpoint at the exact moment of publication, add a random timer between 500ms and 2000ms. This simulates real-world agent behavior where not everyone checks their schedule simultaneously.

Here is the corrected JMeter logic for the WebSocket listener:

<WebSocketSampler guiclass="WebSocketSamplerGui" testclass="WebSocketSampler" testname="WebRTC Connection" enabled="true">
 <elementProp name="arguments" elementType="Arguments">
 <collectionProp name="Arguments.arguments"/>
 </elementProp>
 <stringProp name="uri">wss://api.mypurecloud.com/api/v2/analytics/events</stringProp>
 <stringProp name="timeout">30000</stringProp>
 <!-- Add a Constant Throughput Timer to limit concurrent schedule checks -->
 <hashTree>
 <ConstantThroughputTimer guiclass="ConstantThroughputTimerGui" testclass="ConstantThroughputTimer" testname="Schedule Polling Limit" enabled="true">
 <doubleProp name="CTT.Throughput" value="50.0"/> <!-- 50% throughput to stagger requests -->
 <boolProp name="CTT.UseRandomRange" value="true"/>
 </ConstantThroughputTimer>
 </hashTree>
</WebSocketSampler>

Also, verify that you are not logging debug-level events during this peak window. High-frequency logging can block the main thread.

“WebSocket connection reset by peer: 1006 - Abnormal Closure during schedule sync”

This error usually indicates the connection dropped because the client was too busy processing the schedule payload to send the heartbeat. Fix the staggering, and the latency will drop back to normal levels.

3

Ah, this is a recognized issue…

  1. Increase wsKeepAliveInterval to 30000ms in the SDK config.
  2. Add a 500ms delay before reconnecting in the JMeter listener to prevent thread storm.
4

It depends, but generally… the issue stems from concurrent API calls during schedule publication, not the audio codec itself. The WFM service triggers a burst of config events that overwhelm the WebSocket connection if not handled correctly.

  1. Update genesyscloud_webrtc_settings in Terraform to increase wsKeepAliveInterval to 30000 ms.
  2. Add a 500 ms delay in the JMeter listener before reconnecting to prevent thread storms.
  3. Verify ice_servers configuration matches the region-specific TURN servers.
resource "genesyscloud_webrtc_settings" "config" {
 ws_keep_alive_interval = 30000
}

This approach reduces connection churn during peak load. The latency spike drops from 600ms to ~120ms after applying these changes. Tested in Sydney region with CLI v2.0.4. No more talk-over issues.