Decoding WebRTC Diagnostic Logs via the API for Troubleshooting

How-To-Articles
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Decoding WebRTC Diagnostic Logs via the API for Troubleshooting

Executive Summary & Architectural Context

When a contact center agent working from home reports, “My softphone just disconnected,” IT support teams are often flying blind. Relying on the agent to run a speed test or describe their Wi-Fi signal strength is an unreliable diagnostic methodology.

Because Genesys Cloud relies on WebRTC (Web Real-Time Communication) for browser-based audio, the system silently records hundreds of micro-telemetry points during every active call-including jitter, packet loss, codec negotiation, and ICE candidate selection.

While the basic UI provides a high-level MOS (Mean Opinion Score), deep network engineering requires extracting the raw WebRTC diagnostic logs. This masterclass details how to utilize the Diagnostics API to extract these logs programmatically, parse the JSON payloads, and definitively prove whether a dropped call was caused by the agent’s ISP, a corporate firewall, or a browser crash.

Prerequisites, Roles & Licensing

  • Licensing: Available on all Genesys Cloud CX tiers.
  • Roles & Permissions:
    • OAuth Client with telephony:readonly and diagnostics:readonly.
  • Platform Dependencies:
    • The exact conversationId of the dropped interaction.

The Implementation Deep-Dive

1. Understanding the WebRTC Object Model

A single phone call (Conversation) contains multiple Participants (Customer, IVR, Agent). The WebRTC telemetry is exclusively attached to the Agent’s Media Session (the specific SIP endpoint running inside their Chrome/Edge browser).

2. Extracting the Conversation Metadata

Before pulling diagnostics, you must find the specific participant ID of the agent’s WebRTC session.

  1. Execute: GET /api/v2/conversations/{conversationId}
  2. Parse the JSON response. Look at the participants array.
  3. Find the participant where purpose == "agent" or purpose == "user".
  4. Inside that participant object, look at the calls array. Extract the id of the call segment (this is the sessionId).

3. Querying the Diagnostics API

With the conversationId and the sessionId, you can extract the raw WebRTC payload.

  1. Endpoint: GET /api/v2/diagnostics/webrtc/conversations/{conversationId}/participants/{participantId}
  2. The Response Payload: The API will return an array of diagnostic events.
{
  "entities": [
    {
      "type": "ICEConnectionStateChange",
      "timestamp": "2026-05-14T10:05:22Z",
      "details": { "state": "disconnected" }
    },
    {
      "type": "RtcStats",
      "timestamp": "2026-05-14T10:05:20Z",
      "details": {
        "packetsLost": 1450,
        "jitterBufferDelay": 1.45,
        "roundTripTime": 0.85
      }
    }
  ]
}

4. Interpreting the Telemetry (Root Cause Analysis)

Scenario A: Massive Packet Loss (ISP Issue)
If the RtcStats log shows packetsLost skyrocketing, and roundTripTime exceeding 0.15 (150ms), the agent’s local network is congested. If they are on Wi-Fi, microwave interference or bandwidth saturation (e.g., someone streaming 4K video on the same network) is destroying the UDP stream.

Scenario B: ICE Candidate Failures (Firewall Issue)
Before audio flows, WebRTC uses STUN/TURN servers to punch through NAT routers (ICE Negotiation). If the logs show ICEConnectionStateChange rapidly transitioning from checking → failed, the audio never connected. This definitively proves the agent’s corporate VPN or home router firewall is blocking outbound UDP traffic on the required WebRTC ports (16384-32768).

Scenario C: Browser Resource Exhaustion
If the logs show the audioLevel randomly dropping to 0, but packet loss is 0, the browser tab likely ran out of RAM, or the Windows OS forcibly muted the microphone due to driver conflicts.

Validation, Edge Cases & Troubleshooting

Edge Case 1: Ephemeral Data Limits

WebRTC diagnostic logs are highly volatile and consume massive amounts of storage on the Genesys backend.

  • The Trap: You cannot query WebRTC diagnostics for a call that happened 30 days ago. Genesys Cloud purges this granular telemetry rapidly (usually within 7-14 days).
  • Solution: If your organization requires long-term tracking of agent network health, you must write a cron job that automatically queries the Diagnostics API every night for all calls that dropped below a MOS of 3.5, extracts the JSON, and stores it in your own AWS S3 bucket for historical trend analysis.

Edge Case 2: TURN Server Fallbacks

If an agent’s network blocks direct UDP, WebRTC will attempt to fall back to a TURN server (relaying audio over TCP port 443).

  • Troubleshooting: If a call connects but the agent complains of a severe 2-second delay, check the diagnostics API for the selectedCandidatePair. If the protocol is TCP instead of UDP, the call fell back to a TURN relay. TCP is disastrous for real-time audio. The IT team must fix the UDP firewall block.

Official References