What Is MOS? VoIP Call Quality Scores Explained
MOS (Mean Opinion Score) is the standard way to express voice call quality as a single number from 1 (bad) to 5 (excellent). It started as a literal opinion score: panels of listeners rated calls, and the average was the MOS. Today almost all VoIP MOS figures are estimated from network measurements using the ITU-T E-model.
The MOS scale
| MOS | R-factor | Quality | User satisfaction |
|---|---|---|---|
| 4.3 – 4.5 | 90 – 100 | Excellent | Very satisfied |
| 4.0 – 4.3 | 80 – 90 | Good | Satisfied |
| 3.6 – 4.0 | 70 – 80 | Fair | Some users dissatisfied |
| 3.1 – 3.6 | 60 – 70 | Poor | Many users dissatisfied |
| 1.0 – 3.1 | 0 – 60 | Bad | Nearly all dissatisfied |
In practice, narrowband VoIP tops out at about 4.4. Even a perfect G.711 call can't reach 5, because the codec limits audio bandwidth to traditional telephone quality. A MOS of 4.0 or more is a good target for business calls.
R-factor and the E-model
The E-model (ITU-T G.107) starts with a perfect score and subtracts "impairments". The result is the R-factor, from 0 to 100, which converts to MOS with a standard formula. The main impairments are:
- Delay (Id): mouth-to-ear delay, the time from one person speaking to the other hearing it. Below about 150 ms it barely matters. Beyond about 177 ms the penalty rises steeply, because people start talking over each other.
- Equipment impairment (Ie-eff): how much the codec degrades audio, and how badly it copes with packet loss. G.711 starts at 0; G.729A starts at 11.
Our MOS calculator uses a simplified E-model, so you can see how each factor moves the score.
What affects call quality
Latency
The one-way network delay plus codec and jitter buffer delay. Aim for under 150 ms mouth-to-ear. Local calls are typically 20–60 ms. Calls between South Africa and Europe add roughly 80–100 ms of network delay each way, and to the US more still. That's why picking a server or carrier region close to your users matters.
Jitter
Variation in packet arrival times. Phones smooth it out with a jitter buffer, but a buffer adds delay. Keep jitter under 30 ms. High jitter almost always means congestion somewhere on the path, commonly the office upload link.
Packet loss
The most damaging factor. With packet loss concealment, G.711 tolerates around 1% loss with little audible effect, but 3–5% is clearly noticeable. Wi-Fi, saturated links and poor ISP routes are the usual causes.
Codec choice
Compressed codecs save bandwidth but lower the best possible score, and are often more sensitive to loss. Wideband codecs like G.722 and Opus sound noticeably better than narrowband ones. They're measured on an extended scale that this classic MOS range doesn't capture.
How to improve MOS
- Fix packet loss first. Move phones off Wi-Fi where possible, replace bad cables, and check for a saturated uplink.
- Enable QoS. Mark RTP as DSCP EF (46) and SIP as CS3 or AF31, and prioritise them on your router, particularly on the upload side.
- Shorten the path. Use a carrier or hosted PBX with a nearby point of presence.
- Use G.711 or Opus where bandwidth allows, and enable PLC on endpoints.
- Measure under load. Quality problems usually appear at peak times. Test then, not at 7am.
Try the MOS Call Quality Estimator
Estimate MOS and R-factor from latency, jitter, packet loss and codec, with tips to fix bad calls.