WebRTC softphone dropping ICE candidates mid-shift breaking historical wrap-up metrics for Erlang forecast

The genesys-cloud-webphone-sdk v3.14.2 on the agent desktop keeps throwing ICE_CANDIDATE_FAILURE around the 45 minute mark of extended queue sessions. It’s happening right during the peak seasonal volume window. RTCPeerConnection state flips to failed before the agent hits wrap-up. Missing those final disposition timestamps wrecks the seasonal pattern baseline. The Erlang C model expects consistent ACD handling times to calculate shrinkage and service levels. We’ve got a Python ingestion job hitting /api/v2/interaction/analytics/conversations/details/query every 15 minutes. WebRTC gaps leave 12 to 14 percent of the daily dataset null.

Edge deployment runs v23.11.4 on EKS. BYOC network policy allows outbound 443 and 3478 for STUN. TURN relay times out on the secondary AZ. Checked the Architect flow routing logic. Everything routes to the correct skill group. Softphone config JSON has enableIceRestart set to false to prevent mid-call audio glitches. Changing that to true doesn’t stabilize the connection without introducing latency spikes that throw off the average speed of answer calculations.

Network traces point to a NAT timeout on the corporate firewall. Genesys WebRTC diagnostics page reports healthy STUN connectivity. Doing jack all about the packet loss in the middle of the work phase. Logs show a burst of candidate-gathering-done events right before the drop. Does the SDK cache failed ICE candidates locally before flushing to the telemetry endpoint? Manual curl against the interaction query endpoint returns full JSON payloads for non-WebRTC channels.

Forecasting pipeline needs those exact timestamp deltas to adjust the seasonal index for Q4. Wrap-up state machine gets stuck in pending when the peer connection drops.

Looking at the raw STUN binding responses now. Timeout values seem off by a few seconds.

Problem

The 45-minute drop aligns with the default WebRTC signaling timeout in your org config. When the softphone SDK misses a keepalive heartbeat, the STUN server drops the ICE candidates. You need to bump the iceTransportPolicy and adjust the session keepalive interval via the configuration API before pushing it through your backup pipeline.

Code

PUT /api/v2/organizations/{organizationId}/settings
Content-Type: application/json
Authorization: Bearer <token>

{
 "webRTC": {
 "iceTransportPolicy": "all",
 "keepaliveIntervalMs": 15000,
 "maxIceGatheringTimeoutMs": 300000
 }
}

Run this against staging first. If you’re managing it through Terraform, map it to genesyscloud_organization_settings and force a plan refresh. It’s hitting the timeout wall because the default interval is just too aggressive for your current network latency.

Error

Watch for 403 Forbidden if your OAuth token lacks the organization:write scope. The backend rejects payloads missing the webRTC root key, so double-check the schema validation before you merge. Make sure you don’t overwrite the production backup snapshot during the apply phase.

Question

The payload structure looks clean on paper. Check the rollout script logs. The agent desktop might be caching an older config snapshot. Run a trace on the next shift.

Bumping the keepalive interval fixes the signaling handshake, but you’ll want to watch the audio buffer flush timing. When RTCPeerConnection flips to failed at that 45 minute mark, the browser dumps the local media stream before the Recording API can latch onto it. The final disposition audio gets truncated. If you’re piping that stream into a third-party voice biometric engine for continuous enrollment, the fingerprint breaks completely.

Check your mediaStreamTrackSettings in the admin UI under Telephony > Softphone. The audioBufferFlushInterval defaults to 2000ms. Drop it to 800ms and toggle forceLocalEchoCancellation off. This forces the SDK to write smaller chunks to the local cache before the STUN timeout kills the connection.

The workaround isn’t pretty, but it keeps the raw PCM data intact long enough for the recording webhook to fire. You’ll also want to monitor the iceRestartPolicy flag. Genesys is slowly moving toward WebTransport for media, so hardcoding ICE workarounds won’t save you.

Just watch the browser memory leak on Chromium builds past 120. The audio context hangs. No clean shutdown event fires.

{
 "action": "REST",
 "url": "/api/v2/interactions/web/instances/{{interaction.id}}",
 "method": "PATCH",
 "body": { "state": "completed", "dispositionCode": { "id": "wrapup-id" } }
}

genesys-cloud-webphone-sdk kills the peer connection, so force the disposition via a Studio REST Proxy call before the timeout hits. Catch the ICE_CANDIDATE_FAILURE event in your error handler to trigger this patch immediately.