"My speed test shows 500 Mb/s, and yet the video freezes." This situation is more common than you might think. A speed test measures how much data can pass per second, like the width of a highway. It says little about how steady the traffic is, about occasional jams, or about vehicles that never arrive. These three dimensions have names: latency, jitter and packet loss. Understanding them lets you make the right diagnosis and avoid buying a faster plan that will change nothing.
Three indicators, three different questions
On the Internet, all data travels cut into packets of a few hundred to about 1,500 bytes. Each of them travels a path through several routers before reaching its destination.
- Latency answers the question: how long does a packet take to make the round trip? It is measured in milliseconds (ms); it is the value shown by the
pingcommand, hence its nickname. - Jitter answers: is that time steady? A latency that swings between 10 and 150 ms has high jitter, even if its average looks fine.
- Packet loss answers: how many packets never arrive? It is expressed as a percentage of the number sent.
Reference values
| Indicator | Excellent | Acceptable | Problematic |
|---|---|---|---|
| Latency to a nearby server | under 20 ms | 20 to 60 ms | over 100 ms |
| Jitter | under 5 ms | 5 to 20 ms | over 30 ms |
| Packet loss | 0% | under 0.5% | 1% and above |
These thresholds are indicative. The access technology sets the floor: fiber generally offers the lowest latencies, ADSL and 4G add a few tens of milliseconds, and a geostationary satellite structurally exceeds 500 ms, because the signal travels about 36,000 km to the satellite, in each direction.
Why these indicators disrupt video
Most video streams on the Internet use TCP, the protocol that guarantees every packet arrives, resending it if necessary. This guarantee has a cost.
Losses make throughput collapse
When TCP detects a loss, it resends the missing packet and, above all, sharply reduces its sending rate, assuming the network is congested. It then gradually ramps back up. With 1 to 2% of regular losses, a 500 Mb/s connection may carry only a few megabits per second on a given stream. This is how a speed test, which opens several connections in parallel, can stay good while a single video stream collapses.
Latency limits a connection's throughput
The longer the round trip, the longer TCP takes to speed up and to recover from a loss. A server on the other side of the world, with 250 ms of latency, will have more trouble delivering a steady 4K stream than a server in your own country, at equal bandwidth.
Jitter drains the buffer
The player counts on a steady supply. If packets arrive in bursts separated by gaps, the reserve drains during the gaps. A larger buffer partly compensates: that is the subject of the article fixing buffering.
The case of MPEG-TS streams
Some live TV streams are sent as MPEG-TS, sometimes over UDP, a protocol that does not resend lost packets. A loss then shows immediately: colored blocks, a picture that tears, sound that crackles for a fraction of a second. This very particular symptom points to losses rather than bandwidth.
Bufferbloat: when the router itself creates latency
Bufferbloat is a little-known yet very widespread phenomenon. Routers and home gateways have queues to temporarily store packets when the link is full. If these queues are too large, they fill up during a big transfer and each packet waits its turn for hundreds of milliseconds.
The typical scenario: someone in the household starts a photo backup to the cloud, which saturates the upload bandwidth. Immediately, the latency of the whole household jumps from 15 to 400 ms. The acknowledgments sent by the TV get stuck in the queue, and the video, though being received, starts to cut out.
Detecting it
- Start a continuous
pingto a reliable server (see the next section) and note the idle latency. - While the ping runs, start a speed test on another device, or a large file upload.
- If latency is multiplied by ten during the test, your connection suffers from bufferbloat.
Several speed test sites now show "latency under load", which makes exactly this measurement automatically.
Fixing it
- Limit large uploads to off-peak hours: schedule cloud backups at night.
- Enable quality of service management if your router offers it; settings vary from one provider to another.
- For advanced users, a router that supports active queue management (algorithms such as
fq_codelorcake, often called SQM) dramatically reduces the problem, at the cost of a slight limit on maximum speed.
Measuring it yourself with system tools
There is no need to install software: the basic network commands are present on every computer.
The ping command
On Windows, open the command prompt and type:
ping -n 100 192.168.1.1
ping -n 100 un-serveur-connu.exempleOn macOS and Linux, the equivalent option is -c:
ping -c 100 192.168.1.1The first address is that of the router in most homes (it may be 192.168.0.1 or something else; it appears in the computer's network settings under the name "gateway"). The second is a server on the Internet, for example your provider's website. At the end of the measurement, the system shows the minimum, maximum, average and the percentage of lost packets.
Interpreting the results
| Ping to the router | Ping to the Internet | Diagnosis |
|---|---|---|
| Stable, 1 to 5 ms, 0% loss | Stable, 0% loss | Healthy connection: look at the source or the device |
| Uneven, losses | Uneven, losses | Local problem: Wi-Fi, cable or powerline |
| Stable, 0% loss | Uneven or losses | Problem on the line or at the provider |
This table is the key to diagnosis: comparing the router and the Internet clearly separates what happens in your home from what happens beyond. On Wi-Fi, a ping to the router that regularly exceeds 20 or 30 ms points to a weak signal or a crowded channel; the guide Wi-Fi, Ethernet or powerline offers solutions.
Following the path with tracert and pathping
The tracert command (Windows) or traceroute (macOS, Linux) shows every router crossed on the way to the destination, with its latency. On Windows, pathping goes further: for a few minutes, it measures losses at each hop. Beware of a common misreading: many routers answer tests with low priority. A loss shown at an intermediate hop is only significant if it carries through all the following hops to the destination.
The most common causes and their remedies
- Weak or crowded Wi-Fi: losses and jitter right from the ping to the router. Move closer, switch to 5 GHz, or use a cable.
- Damaged Ethernet cable: a connector with a broken clip or a cable crushed under furniture can cause intermittent losses. Replacing the cable costs little and removes the doubt.
- Powerline disturbed by an electrical appliance: losses appear when the washing machine or a charger is running.
- Upload saturation (bufferbloat): latency that soars during uploads.
- Degraded copper line: losses even over Ethernet, often worse in damp weather. Only the provider can intervene; keep your dated measurements to support your request.
- Distant or overloaded remote server: everything is healthy up to your provider, then latency climbs. Nothing to do at home except increase the player's buffer.
If the degradation persists and is down to your provider, first send them a written complaint. In France, Arcep, the telecommunications regulator, also collects user reports on its site J'alerte l'Arcep; this service does not handle individual disputes but helps spot recurring faults. Other countries have their own regulators and reporting tools.
In summary
- Latency, jitter and packet loss complement bandwidth and explain many dropouts.
- Landmarks: under 20 ms of latency, under 5 ms of jitter and 0% loss are excellent values.
- A few percent of loss are enough to collapse the throughput of a video stream over TCP.
- Compare the ping to the router and to the Internet to know whether the problem is in your home or beyond.
- Bufferbloat is detected by measuring latency during a large transfer.
Frequently asked questions
What latency do I need to watch videos?
Video on demand tolerates a latency of 50 to 100 ms well, because the player preloads several seconds. What bothers it most are jitter and packet loss, which make data arrive unevenly.
My ping to the router is 40 ms over Wi-Fi: is that normal?
No. To the router, on decent Wi-Fi, latency generally stays under 10 ms. A higher or highly variable value indicates a weak signal, a crowded channel or a badly placed extender.
Do online speed tests measure jitter?
Most display it, sometimes under the name jitter, and some also measure latency under load. Keep in mind that these tests target a nearby server: they do not necessarily reflect the path to the server delivering your video.
Do I absolutely need 0% loss?
An isolated loss now and then has no consequence. On the other hand, regular losses above 0.5 to 1% clearly degrade the video and justify looking for the cause.
