Where latency comes from
- Distance. Light in fiber travels about 200 km per millisecond, so every 100 km of fiber adds about 1 ms of round trip. Richmond to the West Coast and back is roughly 60–80 ms on a good route.
- Queues. When a router is busy, packets wait in line. This is the part you can fix — see bufferbloat.
- Access technology. Fiber is lowest, cable and fixed wireless next, then DSL and cellular. Geostationary satellite adds about 500–600 ms round trip from physics alone.
- Wi-Fi. Interference and retries add both latency and jitter.
- The phones themselves. Codec packetization (usually 20 ms) and the jitter buffer (often 20–60 ms) add to mouth-to-ear delay.
Latency budget for a phone call
| One-way mouth-to-ear delay | ITU-T G.114 guidance |
|---|---|
| 0–150 ms | Acceptable for most user applications |
| 150–400 ms | Acceptable if administrators are aware of the impact |
| > 400 ms | Unacceptable for general network planning |
Your measured round trip to the nearest server is only one slice of that budget. That's why this site calls anything over 100 ms to a nearby server "fair" — there has to be room left for the carrier, the far end, and the phones.
Confirm it from a command prompt
Browsers can't send raw ICMP or UDP. To test the actual path to your VoIP carrier, use the SIP server address from your phone's settings (shown here as sip.example.com):
# Windows — 100 pings, then a per-hop loss report
ping -n 100 sip.example.com
pathping sip.example.com
# macOS / Linux
ping -c 100 sip.example.com
mtr -rwc 100 sip.example.com # install mtr first (brew install mtr / apt install mtr)Loss that starts at one hop and continues to the end of the path is real. Loss on a single middle hop that does not carry through to the last hop is usually just a router that de-prioritizes ping replies — not a problem.
Frequently asked questions
What is latency in VoIP?
Latency is the time it takes a voice packet to travel from the speaker to the listener. Ping tools report round-trip time — there and back. For calls, ITU-T G.114 recommends one-way mouth-to-ear delay under 150 ms. Above that, conversation starts to feel like a walkie-talkie.
What is a good ping for phone calls?
Under 100 ms round trip to a nearby server is good; 100–200 ms is fair; over 200 ms will usually be noticed on calls. Most fiber and cable connections in the U.S. ping a nearby edge server in 5–40 ms.
Is ping the same as latency?
Close. Ping is a tool that measures round-trip latency with ICMP echo packets. Latency is the general term for delay. This browser test measures round-trip latency with small HTTPS requests because browsers can't send ICMP.
What causes high latency?
Long distance or poor routing, congested queues in routers (bufferbloat), Wi-Fi interference, satellite or cellular links, VPNs that send traffic far away before it reaches the internet, and overloaded or failing equipment.
How do I reduce latency for VoIP?
Use wired Ethernet for phones, enable QoS or Smart Queue Management on the router, keep heavy uploads off the voice network, avoid sending voice through a distant VPN, choose a VoIP provider with servers near you, and replace aging modems or routers.
Does fiber internet have lower latency than cable?
Usually. Fiber access typically adds a few milliseconds; cable (DOCSIS) access commonly adds more, and its latency rises more when the upstream is busy. The route to your VoIP provider matters as much as the access type.
Why is my latency fine at night but bad during the day?
Daytime congestion. Either your own connection fills up with business traffic, or the provider's network is busy at peak hours. Run the Bufferbloat test during business hours to see which.
Sources & references
- ITU-T G.114 — One-way transmission time
- IETF RFC 3550 — RTP: A Transport Protocol for Real-Time Applications
- Bufferbloat.net project
Written by Eric · Reviewed November 2026