Low latency means less delay between an event and the moment it reaches someone or produces a visible response—but there is no single threshold that makes every experience feel seamless. A packet’s network travel time, a live stream’s glass-to-glass delay, and the time from a user action to its visible effect measure different parts of the experience.
What is latency?
Latency is delay. The number is useful only when you know what starts the clock and what stops it. A network measurement may time packets crossing a particular segment; a user-experience measurement may include capture, processing, buffering, decoding, and display.
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MDN distinguishes network latency—the time a packet takes to travel from source to destination—from media codec latency, the time needed to encode or decode data: MDN Web Docs’ latency glossary, last modified July 11, 2025.
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One-way delay measures travel from sender to receiver. Round-trip time (RTT) measures a request and its reply. Many consumer ping readings are RTT measurements, so they cannot be compared directly with one-way figures. A ping result also does not, by itself, tell you how long a complete video or interactive application takes to respond.
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Distance imposes a physical floor, and real routes are not straight lines. Apple’s archived networking documentation gives illustrative New York–San Francisco propagation lower bounds of about 30–48 ms round trip over copper and about 48 ms over optical fiber. These are physics-based examples in documentation last updated July 19, 2012, not current measurements of a particular route: Apple’s archived latency guide.
Glass-to-glass latency: event to playback
For live media, glass-to-glass delay runs from an event happening in the real world until its video or audio is appropriately played on the viewer’s device. It can include capture, encoding, transport, ingest and distribution, receiver buffering, decoding, and display. It is therefore broader than network transit alone. The IETF explains this end-to-end scope in RFC 9317 (2022).
Delivery latency: encoder to decoder availability
Delivery latency covers the interval from encoder output until media is available to the decoder. Depending on the system, that interval may include transport, retransmission, forward error correction, and buffering. It describes a media-delivery segment rather than necessarily measuring what a viewer sees. The DASH-IF interoperability report distinguishes this from broader end-to-end and interaction measures.
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Round-trip interaction delay: action to visible effect
For a remote game, call, or control system, the meaningful question may be how long it takes from a person’s action until that person sees its effect. The path can include input capture and delivery, server or engine processing, frame generation, return delivery, decoding, and display. The DASH-IF report describes round-trip interaction delay as the sum of user interaction delay and the age of the content shown to the user.
What does low latency mean?
“Low” depends on the task and the measurement boundary. A delay suitable for broadcast-style live viewing may be far too long for a two-way conversation or remote control. Standards and technical reports use different categories because they address different experiences, not one universal promise of “seamless.”
| Figure or category | What it describes | How to interpret it |
|---|---|---|
| Under 1 second | IETF’s rough category for ultra-low-latency streaming media, based on glass-to-glass delay (RFC 9317, 2022). | A streaming category, not a guarantee for every service or network. The RFC notes that variation can make it difficult to achieve without visible artifacts. |
| Under 10 seconds | IETF’s low-latency live-media category for a broadcast-TV-like experience (RFC 9317, 2022). | This is not an appropriate sub-second target for a video call, game action, or hardware-control response. |
| On the order of no more than hundreds of milliseconds | IETF terminology for real-time media and interactive action/reaction (RFC 8825, 2020). | A broad scale, not a precise guarantee or threshold for every application. |
| 400 ms one-way | ITU-T G.114’s general network-planning guidance (2003 revision). | The recommendation says highly interactive tasks can be affected by much lower delay; 400 ms should not be treated as a responsive-conversation target. |
Sources: IETF RFC 9317, IETF RFC 8825, and the ITU-T G.114 recommendation summary.
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How does latency affect real-time interactions?
Delay makes feedback arrive later. In a conversation, replies can feel less immediate; in a remote game or control system, an action may appear to take effect after the person has already made another decision. The relevant measure is often the complete action-to-visible-response path, not a single network ping.
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Consistency matters alongside the average. A relatively steady delay can feel more predictable than a lower average that swings sharply. Jitter—variation in packet arrival time—can make media arrive unevenly, while packet loss or late frames can interrupt the experience. The effect depends on the application and how its receiver handles those conditions.
Why can reducing delay make playback less smooth?
A receiver can wait briefly before playing media so that packets arriving at uneven intervals have time to catch up. That buffer helps smooth playback, but the extra waiting makes the content older by the time it is heard or seen.
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Playing media sooner reduces added playout delay, but leaves less room to absorb network variation. Late packets may then contribute to visible artifacts, freezes, or dropped frames. The WebRTC playout-delay proposal describes this balance; it is an experimental proposal, not a finalized standard, and says the suggested minimum and maximum range is best effort, with the receiver deciding how to handle a late frame: WebRTC playout-delay proposal.
For a live broadcast, viewers may value continuity more than being as close as possible to the event. For a two-way interaction, stale content can be more damaging than occasional disruption. Low delay can also constrain flexibility in delivery and make transient network problems more visible; a low-latency label alone does not tell you the picture quality, resilience, or consistency to expect. RFC 9317 discusses these trade-offs for streaming delivery.
Where does delay accumulate?
Latency is accumulated across a chain; it is not one setting inside “the internet.” A live-media path can include capture, encoding, network transport, distribution, receiver buffering, decoding, and display. An interactive path may add input-device capture, server or engine work, frame generation, and a return path to the user.
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- Distance and routing: Signals take time to travel, and the route between endpoints may be longer than the straight-line distance.
- Congestion and queueing: Routers may hold packets while busy, adding delay.
- Loss and recovery: Packet loss, retransmission, and wireless errors can delay delivery or degrade playback.
- Processing and media handling: Encoding, decoding, server-side work, and frame generation all consume time.
- Buffering: A receiver can deliberately wait to smooth playback, increasing content age.
Apple’s archived guide explains how distance, router buffering under congestion, retransmission, and packet loss can add delay; its examples are explanatory principles, not a current benchmark for your connection: Apple’s archived networking documentation.
How to troubleshoot a latency problem
Start with the symptom and the measurement boundary. A high ping suggests delay on the measured network round trip, but does not isolate the cause of a stream’s glass-to-glass delay or an application’s action-to-visible response. If a device is on Wi-Fi and supports Ethernet, testing it over a wired connection can help assess the local wireless segment.
- Identify what feels late. Is the issue delayed live video, a slow response to an action, a lagging conversation, or an inconsistent connection?
- Check what the number measures. Note whether it is one-way or round-trip network time, glass-to-glass delay, delivery latency, startup delay, or action-to-visible-response time.
- Test the local connection where possible. If the device has an Ethernet port, connect it directly to the network and compare the same symptom under otherwise similar conditions. If it lacks a port, a compatible Ethernet adapter may be needed.
- Compare consistency as well as delay. Notice whether the problem is persistent or appears as jitter, freezes, artifacts, or late-frame drops.
- Keep the scope of any improvement clear. A wired local connection can help investigate Wi-Fi-related problems; it cannot control long geographic routes, congestion elsewhere on the internet, service-side processing, or buffering in a media service.
A Cat 6 Ethernet cable is one way to make a wired test when the device and network equipment support Ethernet. The cable does not guarantee lower latency or eliminate internet lag; compatibility depends on the device’s available ports and any required adapter.
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Before comparing services or devices, ask what the number includes. A network RTT is not equivalent to glass-to-glass delay, and neither necessarily equals the time from an action to its visible response. Also consider whether the figure describes a stable result or an average that hides variation.
Quick Recap
- Boundary: What are the exact start and end points?
- Direction: Is the figure one-way or round trip?
- Consistency: Does the claim address jitter or only an average delay?
- Resilience: What happens under packet loss or changing network conditions—retransmissions, artifacts, freezes, or dropped frames?
- Experience: Is the claim about live broadcast viewing, a call, remote gameplay, or interactive control?
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