# What determines VoIP call quality?

> VoIP call quality is decided by four network impairments — latency, jitter, packet loss, and the codec — summarised by a single Mean Opinion Score (MOS). Because voice is carried as RTP packets over IP, delayed, out-of-order, or dropped packets degrade the audio the listener hears. Managing these with QoS, jitter buffers, and the right codec is what keeps calls clear at scale.

- Canonical URL: https://www.broadnet.me/answers/voip-quality
- Language: en
- Last reviewed: 2026-07-22
- Publisher: BroadNet (https://www.broadnet.me)

## How VoIP quality is measured: MOS

Mean Opinion Score (MOS) rates call quality from 1 (bad) to 5 (excellent); 4.0+ is toll quality, below 3.5 is noticeably degraded. Originally a human listening test, MOS is now estimated automatically (per ITU-T models) from measured latency, jitter, and packet loss. It is the single number carriers and platforms track to compare routes and diagnose a bad call path. See /answers/what-is-mos.

## The impairments that lower quality

Latency (one-way delay above ~150ms makes conversation feel like a walkie-talkie); jitter (variation in packet arrival that garbles audio unless a jitter buffer smooths it); and packet loss (missing RTP packets heard as clipping or dropouts — above ~1% is audible). These accumulate across every hop, which is why direct, well-provisioned routes sound better than long ones. See /answers/what-is-jitter and /answers/what-is-rtp.

## Codecs and the quality-vs-bandwidth trade-off

The codec sets the ceiling. G.711 is uncompressed toll-quality at ~64 kbit/s; Opus adapts bitrate and is resilient to loss; low-bitrate codecs save bandwidth but cap MOS. Transcoding between codecs along the path costs quality and adds delay, so matching codecs end to end matters. See /answers/what-is-voice-codec and /answers/what-is-sip.

## How to protect quality

Prioritise voice with QoS/DiffServ so RTP is not queued behind bulk data; size jitter buffers to trade a little delay for smoother audio; pick loss-resilient codecs like Opus on lossy paths; and choose direct routes with few hops. Continuous MOS, ASR, and PDD monitoring surfaces a degrading route before customers complain. See /answers/what-is-asr, /answers/what-is-pdd, and /products/voice.

## Where poor quality actually comes from

Most bad VoIP is not the codec — it is the network path. Grey or over-hopped wholesale voice routes add latency and loss at each transit carrier, congested access links starve RTP of bandwidth, and unmanaged Wi-Fi introduces jitter. Diagnosing quality means correlating MOS with the route and the KPIs (ASR, ACD, PDD) that reveal a weak carrier leg. See /answers/programmable-voice and /platforms/signaling-cdr.

## What is a good MOS score for VoIP?

A MOS of 4.0 or above is considered toll quality — indistinguishable from a landline to most listeners. 3.5-4.0 is acceptable, 3.0-3.5 is noticeably degraded, and below 3.0 frustrates callers. G.711 on a clean path tops out near 4.4; the goal of QoS, jitter buffers, and direct routing is to keep the delivered MOS as close to that ceiling as the network allows. See /answers/what-is-mos and /products/voice.

## Related

- [All answers](https://www.broadnet.me/answers)
- [Glossary](https://www.broadnet.me/answers)
- [What is MOS (Mean Opinion Score)?](https://www.broadnet.me/answers/what-is-mos)
- [What is jitter in VoIP?](https://www.broadnet.me/answers/what-is-jitter)
- [What are voice codecs (G.711, Opus)?](https://www.broadnet.me/answers/what-is-voice-codec)
- [What is RTP (Real-time Transport Protocol)?](https://www.broadnet.me/answers/what-is-rtp)
- [ASR — Answer-Seizure Ratio explained](https://www.broadnet.me/answers/what-is-asr)
- [What is PDD (Post-Dial Delay)?](https://www.broadnet.me/answers/what-is-pdd)
- [Voice](https://www.broadnet.me/products/voice)
- [Signaling & CDR Intelligence](https://www.broadnet.me/platforms/signaling-cdr)
- [Erlang calculator](https://www.broadnet.me/tools/erlang-calculator)