Packet Loss
What is Packet Loss?
Packet loss is when data packets carrying a VoIP call's voice never arrive at their destination, forcing the receiving end to either drop that slice of audio or mask it, which is what callers hear as a gap, garble, or robotic-sounding word. It is typically caused by network congestion, a failing router, or an overloaded wireless link somewhere along the path, and is one of the three core metrics tracked alongside jitter and latency when diagnosing poor VoIP call quality.
More detail
Packet loss is usually reported as a percentage of packets dropped over a call or time window; even a loss rate as low as 1% can produce audible artifacts on a voice call, while ordinary data transfers tolerate far higher loss thanks to retransmission.
Codecs and jitter buffers can partially mask small amounts of packet loss through concealment techniques that interpolate the missing audio, but sustained or bursty loss above a few percent is usually audible as static, dropouts, or robotic-sounding speech.
Frequently asked
- What causes packet loss on a VoIP call?
- Packet loss usually comes from network congestion, an overloaded router or Wi-Fi link, or a route with too many hops between the two call endpoints — any point along the path that drops packets faster than it can forward them.
- How much packet loss is too much for a phone call?
- Voice quality tools generally treat sustained packet loss above roughly 1-3% as noticeable to callers, and loss above 5% as a clear, disruptive quality problem — well below the loss rates a typical data transfer or web page load would even register.
See also
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