Glossary
What is packet loss?
Packet loss is the percentage of data packets that never reach their destination on a network. Every VoIP call is a stream of small audio packets sent over RTP; when packets are dropped, the receiving side has gaps to fill — heard as choppy, robotic, or missing audio.
Voice is uniquely sensitive to it. A file download just re-requests lost packets; a live call cannot wait, so anything lost is simply gone from the conversation.
What packet loss sounds like on a call
- Words or syllables cut out mid-sentence
- “Robotic” or underwater-sounding voice as the codec interpolates gaps
- One-way audio dropouts while the other side hears you fine
- In severe cases, calls disconnecting entirely
Acceptable packet loss for VoIP
The common engineering guideline is under 1% for good call quality; above roughly 3–5%, degradation becomes obvious and MOS scores fall fast. Burst loss (many consecutive packets) sounds far worse than the same percentage spread evenly, because codecs can conceal isolated gaps but not whole missing words.
What causes packet loss
- Network congestion — links saturated by other traffic; the classic office cause is large uploads competing with calls.
- Wi-Fi interference — wireless is the most common culprit in practice: distance, walls, and channel congestion all drop packets.
- Faulty hardware — failing routers, switches, cables, or NICs.
- Overloaded equipment — consumer routers buckling under many simultaneous streams.
- ISP or backbone issues — loss outside your network, visible in path tests.
How to fix packet loss
- Test first — run a sustained ping or a VoIP-specific quality test to localize whether loss is on Wi-Fi, the LAN, or beyond your ISP handoff.
- Prefer wired — Ethernet for desk positions that live on calls; modern Wi-Fi (6/6E) with good placement for everyone else.
- Turn on QoS — prioritize voice packets on your router so a file sync can’t starve a call.
- Right-size bandwidth — each call needs ~100 Kbps with G.711; the real killer is bufferbloat under load, not raw throughput.
- Replace suspect hardware — a single failing switch port can poison a whole office’s calls.
- Escalate with data — persistent loss beyond your network goes to the ISP with traceroute and loss-by-hop evidence.
Cloud platforms also adapt: modern codecs conceal isolated loss, and jitter buffers smooth uneven arrival — but no software fixes a network dropping 5% of everything.
Common questions
How much packet loss is acceptable?
Under 1% is the standard target for business voice; calls stay clean and concealment handles stray drops. Between 1–3% quality is noticeably degraded on bursts. Above 3–5%, expect complaints on every call. Zero sustained loss is achievable on a healthy wired network.
How do I test for packet loss?
The quick check is a sustained ping (200+ packets) to your VoIP provider’s edge or a public anchor, watching the loss percentage. Better: a VoIP quality test that measures loss, jitter, and latency together, run both wired and on Wi-Fi to isolate the wireless layer. Test during busy hours — loss that only appears under load is congestion, not hardware.
Does packet loss affect video calls too?
Yes, usually more visibly — frozen frames, pixelation, and audio desync. Video streams are larger, so the same congested link drops more of them. Fixing the network for voice fixes video alongside it.
Why do my VoIP calls sound choppy only on Wi-Fi?
Wireless adds its own loss layer: interference, distance, and airtime contention drop packets before they ever reach your router. Wired connections skip that layer entirely, which is why the first diagnostic step is always comparing the same call path over Ethernet.
See DialPhone call quality
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