WebRTC stats API — extracting real-time MOS score from RTCPeerConnection

We need to extract real-time MOS scores from the browser’s WebRTC stats API to display call quality indicators on our custom agent desktop.

Is there a way to access the RTP statistics during a live call, or do we have to wait for the call to end?

Yes! The RTCPeerConnection.getStats() API provides real-time RTP statistics during the call.

async function getMOSScore(peerConnection) {
  const stats = await peerConnection.getStats();
  let jitter, packetsLost, roundTripTime;
  stats.forEach(report => {
    if (report.type === 'inbound-rtp' && report.kind === 'audio') {
      jitter = report.jitter;
      packetsLost = report.packetsLost;
    }
    if (report.type === 'candidate-pair' && report.nominated) {
      roundTripTime = report.currentRoundTripTime;
    }
  });
  return calculateMOS(jitter, packetsLost, roundTripTime);
}

The MOS calculation from raw stats uses the E-model (ITU-T G.107).

The simplified formula: R = 93.2 - (jitter_ms * 0.5) - (packet_loss_pct * 2.5) - (rtt_ms * 0.1). Then convert R to MOS: MOS = 1 + 0.035*R + R*(R-60)*(100-R)*7e-6. This gives you a MOS score from 1.0 to 4.5.

We display the MOS score as a color-coded indicator on our custom desktop.

Agents see the indicator in real-time and can alert their supervisor if quality degrades mid-call.

From a compliance perspective, we log every MOS reading to prove call quality meets our SLA thresholds.

Our contract guarantees MOS ≥ 3.5 for 95% of calls. We sample MOS every 10 seconds during each call, store the readings in Elasticsearch, and generate monthly SLA compliance reports.

For remote agents, correlate MOS drops with VPN reconnection events.

We overlay the MOS timeline with the agent’s network event log. Every VPN reconnect creates a 3-5 second MOS dip. If an agent has 10+ VPN reconnects per shift, their average MOS will be significantly lower than office-based agents.