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Random freezes, missing sound and display glitches usually trace back to one bad driver. Find and replace yours safely.Free scan · under a minuteLatency is the time between a defined event and a later event. For a network frame, that might mean the time it takes to cross a device; for a live documentary stream, it can mean the time between an event happening and its playback on your screen. Those measurements are related, but they are not interchangeable.
What does latency mean?
Latency is elapsed time, but the word alone does not say what started the clock, what stopped it, or where the measurement was taken. A useful latency figure identifies those boundaries, its direction, its units, how it was measured, and the conditions during the test.
For example, IETF RFC 8238 defines latency in its data-center benchmarking context as the time a frame takes to transit the device under test. The measured interval depends on which input and output bit events are selected. For the switch measurements described in that RFC, the required event pairing is FILO—First In, Last Out—which captures the full frame passage, including serialization delay. That device-specific definition should not be treated as a definition of every kind of latency.
How do network latency, ping, and round-trip time differ?
Network latency can refer to a packet’s delay between measurement points. A ping-style exchange, by contrast, sends a request and receives a response; the elapsed time for that exchange is round-trip time (RTT). RTT covers both directions and the responding endpoint’s part in the exchange, so it is not automatically a measurement of one-way latency.
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IETF RFC 8802 describes calculating RTT from PING exchanges. Its method cannot measure forward and reverse delays separately and uses an assumption that they are identical when estimating one-way latency. That is an explicit assumption, not a general property of network paths. Unless a one-way method and its assumptions are justified, report the ping result as RTT rather than dividing it by two and calling it measured one-way delay.
Why can a live documentary stream be delayed?
For streaming video, the relevant viewer-facing measure is often glass-to-glass latency: the time from a real-world event to suitable playback on an end-user device. IETF RFC 9317 describes this interval as including more than packet travel. Encoding, decoding, buffering and, commonly, ingest and distribution through an intermediate service such as a CDN can all contribute.
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That distinction matters when interpreting a visible delay. A packet’s network transit time is only one possible part of the stream’s end-to-end timing; it does not, by itself, establish why playback is behind the event.
RFC 9317 gives rough categories its authors found useful for streaming-media requirements: ultra-low latency is less than 1 second; low-latency live is less than 10 seconds; non-low-latency live is 10 seconds to a few minutes; and on-demand is hours or more. These are useful reference categories from that document, not universal labels or guarantees for every streaming service.
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Why can latency change under load?
A minimum-path result and a result measured under load answer different questions. IETF RFC 2215 defines minimum path latency as the smallest possible packet delay added by a network element, arising from propagation delay, processing limitations, or both; it excludes variable queuing delay. It is a baseline, not an upper bound on delay for best-effort traffic.
ITU-T Recommendation Y.1567, published in October 2023, addresses latency-under-load metrics and measurement methods. Its framing includes UDP at maximum IP-layer capacity load and TCP under maximum-throughput conditions. A minimum measurement should therefore not be presented as a prediction of the delay users will experience when a network is busy.
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How should a latency result be measured and reported?
Before comparing figures, make sure they describe the same interval and test conditions. IETF RFC 9439 identifies six useful components of a fine-grained network performance metric: its name, description, measurement or calculation method, units, measurement points and measurement timing.
- Define the interval: name the start and end events and the measurement points.
- State direction and metric: distinguish device transit, one-way delay, RTT and glass-to-glass latency.
- Describe the method and units: identify the probe or timestamp method, protocol where relevant, and units such as milliseconds, microseconds or nanoseconds.
- Record conditions: include load, packet or frame size, endpoints or route, and the sample window.
- Show variation: provide an appropriate distribution or percentiles when available, rather than relying on a single average.
For data-center device benchmarking specifically, RFC 8239 addresses packet sizes from 64 bytes to 9216 bytes and recommends reporting minimum, average and maximum latency in time units, along with a jitter histogram. Those are device-testing conventions, not universal requirements for measuring every app or stream.
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What should you compare when looking at latency figures?
| Measurement | What interval it describes | What to check |
|---|---|---|
| Device transit latency | A frame passing through a device under test | Event pairing, frame size, units and test conditions |
| Packet or path delay | A packet traveling between specified network measurement points | Direction, endpoints, method and whether the result is a minimum or includes load effects |
| Round-trip time (RTT) | A request-and-response exchange between endpoints | Probe or protocol, endpoints and whether anyone is making an assumption about one-way delay |
| Glass-to-glass latency | A real-world event through playback on an end-user device | Where timing begins and ends, and whether the figure includes encoding, distribution, decoding and buffering |
Two figures labeled only “latency” are not necessarily comparable. Match the interval, measurement points, direction, method and test conditions before using them to judge a connection, device or live stream.
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