CALLTYPES

🎧 Call Quality Score Calculator

Enter your network's latency, jitter, and packet loss to estimate a VoIP call's MOS score and get a Poor, Fair, Good, or Excellent rating, using the simplified ITU-T G.107 E-model.

📶 Estimate VoIP Call Quality

Estimated with the simplified ITU-T G.107 E-model. Indicative only — measured MOS depends on codec and endpoints.

🎧 Estimated call quality

R-factor
90
MOS score
4.34 / 4.5
Rating
Excellent

What is a Call Quality Score Calculator?

It translates raw network measurements into a voice-quality verdict. Feed in the one-way latency, the jitter, and the packet-loss percentage, and it computes the transmission rating factor R, converts it to a Mean Opinion Score between 1.0 and 4.5, and labels the result Poor through Excellent.

Network engineers and IT teams use it to predict how a VoIP or video call will sound before rolling out a deployment, to diagnose a link that users complain about, or to set the thresholds a monitoring system should alert on. The estimate follows the simplified E-model — measured MOS also depends on the codec and endpoints.

❓ Frequently Asked Questions

What is a MOS score?

Mean Opinion Score (MOS) rates voice quality on a 1-to-5 scale, where 4.3+ is excellent and anything below about 3.6 sounds noticeably degraded. This calculator estimates MOS from network conditions using the simplified ITU-T G.107 E-model rather than from human listeners, so treat it as an engineering estimate.

How do latency, jitter, and packet loss affect a call?

Latency is the one-way delay; above roughly 150 ms it causes people to talk over each other. Jitter is variation in that delay, which the jitter buffer smooths at the cost of extra latency. Packet loss drops audio outright — each 1% of loss knocks about 2.5 points off the R-factor — so all three degrade the score.

What formula does the calculator use?

It computes an effective latency (latency + 2×jitter + 10 ms), derives the transmission rating factor R from it, subtracts 2.5 R per percent of packet loss, then maps R to MOS with the standard E-model polynomial. It is the same simplified model used by many VoIP quality tools and is documented in the page's code.

What network conditions give good call quality?

As a rule of thumb aim for latency under 150 ms, jitter under 30 ms, and packet loss under 1% for good VoIP quality; under 100 ms latency with negligible jitter and loss is excellent. Use the calculator to see how close your measured figures land to those bands.