Acceptable Latency Calculator

Whether you are on a VoIP call, playing online, streaming, or in a video meeting, the responsiveness of your connection depends on two key metrics: latency (often measured as RTT) and jitter. This guide explains what they are, which values are acceptable for each type of use, how to measure them, and how to improve your experience when they are too high.

Why use this calculator?

The calculator gives you recommended maximum RTT and jitter per application type. You can consult the reference values only, or enter your current RTT and jitter (from a speed test, game, or VoIP diagnostic) to see if your connection is within acceptable limits for VoIP, gaming, streaming, video conferencing, remote desktop, or general web use.

How the calculator works

Select an application type (e.g. VoIP, competitive gaming, video conferencing). The calculator shows the recommended maximum RTT and jitter in milliseconds for that use. If you enter your measured RTT and jitter, it compares them to these limits and indicates whether your values are acceptable or above recommended.

Real-time vs buffered applications

Benefits of checking latency

• Diagnose issues: Know if your RTT or jitter is above recommended limits for your use case.
• Set expectations: Understand what 'good' means for VoIP, gaming, or video calls.
• Choose server/region: Pick the closest game or VoIP server to stay within limits.
• Plan upgrades: Decide if you need a different connection or QoS to meet real-time requirements.

Limitations and considerations

• Recommended values are guidelines; some users tolerate slightly higher latency, while competitive use may require lower.
• Latency to a speed-test server can differ from latency to your actual game or VoIP provider—measure to the service you use.
• Packet loss is not included in this calculator; high loss degrades quality even with acceptable RTT and jitter.
• Wi-Fi, congestion, and time of day affect results; test under conditions similar to your real usage.

Quick reference (max RTT / jitter)

RTT and jitter are the two main metrics that define how responsive and stable your connection feels. Below we define them precisely and how they relate to packet loss and application type.

What is RTT (Round-Trip Time)?

RTT is the time in milliseconds for a packet to go from your device to the server (or peer) and back. It directly affects how responsive an application feels: voice calls, games, and video calls all need low RTT for a good experience. High RTT causes noticeable delay (e.g. in VoIP) or lag (e.g. in games).

RTT is often called 'ping' in games and network tools. One-way latency is roughly half of RTT (assuming symmetric paths); for real-time apps, the round-trip matters because your action must reach the server and the response must come back before you see the result.

What is Jitter?

Jitter is the variation in delay between packets. Even if average latency is acceptable, high jitter causes uneven delivery: some packets arrive late, leading to stutter, dropouts in voice, or unstable gameplay. Applications use jitter buffers to smooth this out, but too much jitter still degrades quality.

Jitter is typically expressed in milliseconds (ms). It can be calculated as the variation between successive RTT measurements or as the standard deviation of one-way delays. VoIP and video apps often report jitter in their statistics or diagnostics.

Packet loss and latency

Packet loss (packets that never arrive) also hurts real-time quality. High loss causes gaps in voice, frozen video, or retransmissions that increase effective latency. A good connection has low RTT, low jitter, and minimal packet loss (e.g. under 1%). This calculator focuses on RTT and jitter; if your app shows high packet loss, consider checking your link and router as well.

Why it matters by application

Real-time applications (VoIP, gaming, video conferencing) are sensitive to both RTT and jitter. On-demand video streaming can tolerate higher latency because of buffering; competitive gaming and live interaction need the lowest values.

Several factors influence RTT and jitter. Understanding them helps you choose the right improvements—from switching to Ethernet to choosing a closer server or enabling QoS.

• Distance and routing: Data travels at finite speed. The physical distance between you and the server, and the number of network hops, directly affect RTT. Choosing a server or region closer to you reduces latency.
• Wi-Fi vs Ethernet: Wi-Fi adds variable delay and contention with other devices, which increases both latency and jitter. Wired Ethernet usually gives lower and more stable RTT and jitter.
• Congestion: When the link or router is busy, packets wait in queues, increasing delay and variation. Heavy downloads or streaming on the same connection can raise latency and jitter for real-time traffic.
• Router and modem: Older or overloaded equipment can add processing delay and bufferbloat (large queues that cause spikes in latency). Upgrading or tuning the router (e.g. QoS, buffer limits) can help.
• ISP and backbone: Your Internet provider and the path to the service (game server, VoIP provider, etc.) determine the baseline RTT. Some ISPs or plans optimize for low latency; others may have more congested or indirect routes.
• Wireless and mobile: Cellular and long-range Wi-Fi typically have higher and more variable latency than fixed broadband. 4G/5G can be acceptable for casual use but often not ideal for competitive gaming or professional VoIP.

Use these limits as targets for your measured RTT and jitter. Staying at or below them typically gives a good experience for each application type.

• VoIP: RTT < 150 ms and jitter < 30 ms for clear, natural conversation. Above that, delay and choppiness become noticeable.
• Competitive gaming: RTT < 50 ms and low jitter (< 20 ms). Every millisecond counts for reaction time.
• Casual gaming: RTT < 100 ms is usually fine; jitter < 50 ms to avoid stutter.
• Video streaming (on-demand): Latency matters less once buffered; initial load and control (e.g. seeking) benefit from RTT < 200 ms.
• Live streaming / interactive: Similar to VoIP/video conferencing: RTT < 150 ms, jitter < 40 ms.
• Video conferencing: RTT < 150 ms, jitter < 30 ms for smooth audio and video sync.
• Remote desktop: RTT < 100 ms and low jitter for responsive cursor and input.
• General web: RTT < 200 ms is comfortable; higher values make pages feel slow.

Measure RTT and jitter toward the same service you use (game server, VoIP provider, etc.). Generic ping to a random host may not reflect the path to your application.

RTT is often reported as 'ping' in games or network tools. Use ping (ICMP) or dedicated speed/latency tests toward the service you use (e.g. game server, VoIP provider). Note: ping to a random host does not always reflect the path to your actual application server.

Tools and methods

• Ping (command line): On Windows open Command Prompt and run: ping -n 20 example.com. On Mac/Linux: ping -c 20 example.com. The result shows min/avg/max RTT in ms. Jitter can be estimated from the spread between min and max.
• Speed test websites: Sites like Speedtest.net or Fast.com often report latency (ping) and sometimes jitter. Run several tests at different times; latency to the test server may differ from latency to your game or VoIP service.
• In-game ping: Many games display ping or latency to the game server. This is the most relevant RTT for that application. Use it with this calculator by selecting the appropriate usage (e.g. gaming competitive or casual).
• VoIP and video app statistics: Apps like Zoom, Teams, Discord, or VoIP phones often show RTT, jitter, and packet loss in settings or during a call. These values reflect the path to the service's servers and are ideal for checking VoIP or video conferencing limits.
• Traceroute: Traceroute (tracert on Windows) shows each hop to a destination and the delay per hop. It helps identify where latency is introduced (e.g. at your ISP or a distant backbone).

For the most accurate picture, run tests during the same time of day and conditions as your real usage (e.g. while others use the network).

If your RTT or jitter is above the recommended values for your use case, try these steps in order. Start with Ethernet and reducing concurrent traffic; then adjust server choice and router settings.

1. Switch to wired Ethernet if you are on Wi-Fi. This alone often significantly reduces latency and jitter.
2. Stop or pause heavy downloads, streaming, or cloud backups on the same connection while you need low latency.
3. Restart your modem and router. Clear queues and temporary issues that can cause spikes.
4. Choose the closest server or region in your game or application (e.g. same country or continent).
5. Enable QoS (Quality of Service) on your router if available. Prioritize gaming, VoIP, or video conferencing so their packets are not delayed by other traffic.
6. Update router firmware and check for bufferbloat. Some routers have 'Smart Queue' or similar features that limit queue buildup and reduce jitter.
7. If you are on DSL or cable, ensure no line issues (noise, bad filters). Contact your ISP for a line check if latency is consistently high.
8. For gaming or VoIP, consider a connection with low-latency features (e.g. fiber, or an ISP known for gaming/VoIP performance). Not all plans are equal in terms of latency.

If problems persist, run ping and traceroute to the exact host your application uses and share the results with your ISP or support; they can often identify the problematic segment.

Latency and bandwidth are different: one is delay, the other is throughput. For real-time apps, low latency usually matters more than raw speed.

Bandwidth (e.g. in Mbps) is how much data you can send or receive per second. Latency (RTT) is how long it takes for a single packet to make a round trip. They are independent: you can have high bandwidth and high latency, or low bandwidth and low latency.

For real-time applications (VoIP, gaming, video calls), low latency is more important than raw bandwidth. A 20 Mbps connection with 30 ms RTT will feel much better for calls and games than a 100 Mbps connection with 150 ms RTT. For downloads and buffered streaming, bandwidth matters more once latency is reasonable.

When choosing an Internet plan or optimizing for VoIP/gaming, prioritize low and stable latency; use this calculator's limits as a target and check your actual RTT and jitter to the services you use.