Call Quality Threshold Reference

Updated at July 24th, 2026

A quick reference for diagnosing MOS, latency, jitter, and packet loss on voice quality issues.

When a customer reports choppy audio, dropped words, or "bad call quality," the four measurements below are usually enough to confirm whether the network is the cause and how severe the impact is. Use this page alongside the raw numbers from the customer's router, SBC, or trace file.

Latency (one way, mouth to ear)

This is how long it takes a word to travel from the speaker's mouth to the listener's ear. Think of it like a walkie-talkie delay. A little delay is unnoticeable, but once it creeps past 300 ms, people start talking over each other because neither side realizes the other is still speaking. It is a timing problem, not a clarity problem.

Range Rating
Less than 150 ms Excellent
150 to 200 ms Acceptable
200 to 300 ms Noticeable degradation
300 ms and above Poor, unusable

Packet Loss

Every call is broken into hundreds of tiny data packets per second. Packet loss is the percentage of those packets that never arrive. A useful analogy: imagine someone reading you a sentence over a bad radio signal, and every so often, a word just disappears. Your brain fills in small gaps easily; that is what "concealment" does, but once too many words vanish, the sentence stops making sense.

Range Rating
0% Excellent
Up to 1% Good, imperceptible with concealment
1 to 3% Noticeable degradation
3 to 5% Poor
5% and above Generally unusable

Jitter

Jitter measures how consistently packets arrive, not whether they arrive at all. Picture a bucket brigade passing water buckets down a line. If everyone passes at a steady rhythm, the water flows smoothly. If some buckets arrive early and some arrive late, the flow gets uneven, and some buckets bunch up and spill. On a call, this unevenness is what causes that robotic or choppy sound, even when no data was actually lost.

Range Rating
Less than 20 ms Excellent
20 to 30 ms Acceptable
30 ms and above Audible choppiness likely, even with jitter buffers

MOS Score

MOS is the "report card" number. Instead of measuring a single technical factor, it is a 1-to-5 score intended to reflect what a real person would say if you asked them to rate the call. It is calculated from a combination of latency, jitter, and packet loss, so it is the fastest way to answer "how bad is this, really" without digging into three separate numbers.

Range Rating
4.3 to 5.0 Excellent (toll quality)
4.0 to 4.3 Good, typical enterprise Hosted Voice target
3.5 to 4.0 Fair, users start noticing
Below 3.5 Poor, complaints likely
Rule of thumb: when latency stays at or below 150 ms, packet loss stays at or below 1%, and jitter stays at or below 20 to 30 ms, MOS typically lands in the 4.0 to 4.3+ range. If a ticket's numbers fall inside those three limits but the customer still reports a poor experience, look beyond the network path (handset, headset, or local Wi Fi are common culprits).

How do they relate to each other?

If you see... It usually explains...
High latency alone Awkward pauses, people talking over each other
High packet loss Words or syllables cut out, garbled speech
High jitter Choppy or robotic-sounding audio
Low MOS A summary that something above is off, check the other three to find which one

Notes for Diagnosis

  • Burst loss versus scattered loss. Consecutive lost packets degrade MOS far more than the same percentage spread evenly across a call. Ask the customer, or check the trace, for whether the loss was continuous or scattered before rating severity.
  • Round-trip latency above 300 ms. This causes talk-over and interruptions regardless of what the MOS math says, because it breaks natural conversational turn-taking. Call this out separately from the MOS rating whenever it is reported.
  • Codec matters. G.711 tolerates loss worse than Opus or G.729 with good packet loss concealment (PLC). Confirm which codec is in use on the call before comparing the numbers against these thresholds.
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