Glossary

Measured in milliseconds (ms). When you send a request, it travels from your device
through your router, your ISP's network, and reaches the server — then the server's reply
makes the same journey back. That total round-trip time is your ping.

Lower is better:
  < 20 ms — excellent (local fiber or cable)
  20–50 ms — good for most purposes
  50–100 ms — acceptable, some apps may feel sluggish
  > 150 ms — noticeable delay in voice/video calls

Analogy: ping is like the time between asking a question and hearing the first word
of the answer. Even if the answer is long, a slow ping makes the conversation feel awkward.

If your ping bounces between 20 ms and 80 ms every few packets, your jitter is high
(~60 ms). Consistent ping = low jitter. Jitter is what makes voice calls sound choppy —
not high latency itself, but unpredictable latency.

Real-time apps (Zoom, VoIP) use a "jitter buffer" to smooth this out, but a large buffer
introduces its own delay.

Acceptable jitter:
  < 10 ms — transparent, excellent
  10–30 ms — mostly fine, very occasional glitches
  > 30 ms — audible artifacts in voice calls, choppy video

Analogy: imagine a conveyor belt delivering packages. If they arrive at irregular intervals,
the worker at the end keeps stopping and waiting — that's jitter.

The internet breaks all data into small "packets". Packet loss means some of those
packets disappear in transit — dropped by an overloaded router, a bad cable, or wireless
interference. TCP connections will retransmit lost packets, but that causes visible delays.
UDP-based apps (calls, gaming) simply lose the data — no retransmit.

Impact by percentage:
  0% — perfect
  1–2% — barely noticeable for most uses
  3–5% — calls stutter, gaming lags
  > 5% — severe degradation, connections may drop

Packet loss is often intermittent and hard to diagnose. A test may show 0% right now
and 5% in peak hours when your ISP's network is congested.

When your connection is saturated (someone is downloading a large file), your router
buffers packets in a queue. Cheap or misconfigured routers use enormous buffers — instead
of dropping packets to signal congestion, they just hold them. The result: your ping jumps
from 20 ms to 400+ ms while downloading.

This is "bufferbloat" — your link isn't slow, it's just laggy when in use.

We measure it by comparing your idle ping to your ping while the download test is running.
The difference is your bufferbloat. We grade it A–F:
  A: < 5 ms increase — excellent router/modem firmware
  B: < 30 ms — good
  C: < 60 ms — moderate (common in consumer routers)
  D: < 200 ms — high, gaming/calls suffer when downloading
  F: > 400 ms — severe, essentially unusable under load

Fix: update router firmware, enable SQM/QoS, or get a router with CAKE/fq_codel.

Measured in Megabits per second (Mbps). 1 Mbps = 125 KB/s. Most everyday usage
is download-heavy: web pages, video streaming, software updates.

Rough requirements:
  1–5 Mbps   — basic web, SD video
  5–25 Mbps  — HD streaming, video calls
  25–100 Mbps — 4K streaming, faster downloads
  100+ Mbps  — multiple 4K streams, large file transfers

We measure download speed by fetching a large block of random data from the server
and measuring how many bits arrived per second, ignoring the first 2 seconds (TCP warm-up).

Often much lower than download on cable and DSL connections (asymmetric bandwidth).
Fiber connections are usually symmetric.

Upload matters more than people think:
  Video calls send your video upstream
  Cloud backup uploads your files
  Gaming sends your inputs to the server
  Sharing files or screen-sharing

Rough requirements:
  1 Mbps  — basic video calls (low quality)
  3–5 Mbps — HD video calls, screen share
  10+ Mbps  — 4K streaming to others, fast cloud uploads