What Are Latency, Jitter and Packet Loss?

Three measures that determine how a call sounds

In a VoIP call, your voice is cut into tiny slices, usually 20 milliseconds each. Each slice is packed into a packet and sent across the network to the other side — about fifty packets every second, in each direction. On the other side the packets are put back together into continuous voice. As long as they arrive quickly, at a steady pace and with nothing missing, nobody feels that there is a network involved at all.

Latency is how long it takes a packet to travel from the speaker's mouth to the listener's ear. Jitter is how much that time varies from packet to packet: one arrives after 30 milliseconds, the next after 90, the third after 40. Packet loss is the percentage of packets that never arrived, or arrived too late and were discarded.

It is important to understand: these are not speed measures. A one-gigabit connection can have terrible jitter, and a modest 20-megabit connection can deliver perfect calls. How much room there is in the pipe is the bandwidth. How smooth the flow in it is — that is what this article is about.

What you hear, and what it means

Most problems sound like one of a few familiar patterns, and each one almost always has a different cause. When you identify the symptom correctly, you save a lot of guessing:

Note the last two rows. One-way audio — when you hear the customer and they can't hear you, or the other way around — almost never comes from slow internet. It is a problem with address translation in the router (NAT), with a setting called SIP ALG, or with a firewall that blocks the voice channel. The same goes for a consistent disconnect after the same number of seconds in every call. These are configuration problems, and you fix them in the settings, not with a faster internet plan.

How much is "good"? The numbers worth knowing

There are a few accepted thresholds in the telephony world. They are not a law of nature, but they give a clear direction:

• One-way latency — up to about 150 milliseconds, a call feels natural. This is also the recommendation of the International Telecommunication Union (ITU). Above 250–300, people start talking over each other.
• Jitter — up to about 30 milliseconds is usually not noticeable.
• Packet loss — below one percent you almost can't hear it. Above 2–3 percent the voice starts to sound robotic.

How does the device cope with jitter? Every IP phone and every app has a jitter buffer: it holds a few packets aside before playing them, in order to put them back in order at an even pace. This works well on small jitter. But the buffer has a price — every millisecond it waits is added to the latency. Large jitter forces it to choose between two bad options: increase the latency, or discard packets that are late. That is why high jitter ends up sounding like packet loss.

Another measure you will come across is MOS (Mean Opinion Score) — a score from 1 to 5 that tries to translate all these numbers into a single question: "How does it sound to a person?" Many systems calculate it automatically. A score above 4 is a very good call; below 3.5, people already complain.

What actually causes it

In most offices, the problem is a few meters from the phone, not on the other side of the world. The common causes, in the order in which it's worth suspecting them:

• Wi-Fi — the number one cause. A wireless network is shared by all devices, suffers from walls, microwaves and the neighbors' networks, and produces exactly what voice hates: jitter and packet loss.
• A choked connection — someone is uploading a backup to the cloud, downloading a big update or watching video, and has filled the pipe. Voice packets get stuck in line behind the files. It usually happens on the upload side, which is the narrower one.
• A weak router — a home router serving twenty employees overheats, freezes, and sometimes resets in the middle of the day.
• A damaged cable or jack — a network cable crushed under a chair can lose packets without the computer "noticing." A specific phone that always sounds bad while everything else is fine is a strong hint.
• The provider's line — an old copper connection that keeps resyncing, or evening congestion at the internet provider.

It is also worth remembering that every call has two sides. If the customer calls from a mobile phone in an area with weak reception, their voice will arrive broken even when your network is perfect. When only one call sounds bad, and not all calls, the problem is probably on the other side.

How to test it yourself in five minutes

Before you call support, a few simple checks will help narrow the search:

1. Is it everyone or just one? If only one extension sounds bad — suspect the cable, the jack or the device. If the whole office — the connection or the router.
2. Is it always or at certain times? A problem that appears only at noon, or only when a particular employee uploads files, points to congestion.
3. A speed test, with an eye on the ping. On a speed-test site, look also at the ping number (round-trip latency) and the jitter, if shown. Run it several times over the day.
4. A continuous ping test. From the computer, in the command window: ping -t followed by the address of a well-known site (on Windows) for a few minutes. If you see "Request timed out" lines or times that jump from 20 to 300, you've found the problem.
5. Take Wi-Fi out of the equation. Connect the problem phone or computer with a cable and test again.

Write down exactly when the bad calls happened, from which extension and to which number. These three details — time, extension and destination — are exactly what's needed to cross-check against the call data and find the cause.

The solutions, from cheapest to most expensive

Most problems are solved without changing providers and without buying expensive equipment. It's worth going in order:

• A network cable instead of Wi-Fi for every fixed desk. A network cable with PoE gives the phone both power and a stable connection.
• Turn off SIP ALG in the router — it solves a large part of one-way audio and disconnect problems.
• Priority for voice — set up QoS in the router and the switch, so heavy downloads don't push ahead of calls. Sometimes it is combined with separation into a VLAN.
• A business router instead of a home router that keeps resetting.
• Check the plan — mainly the upload speed. Sometimes moving to fiber optic makes the problem disappear entirely.
• A suitable codec — on especially weak connections, a more compressed codec uses less, and some codecs are more resilient to packet loss.

And for employees working from home: exactly the same rules. A mobile app on a congested cellular network, or a phone connected to Wi-Fi in a room far from the router, will sound bad even with an excellent PBX. When there is no choice, a SIM extension bypasses the internet entirely and works on the regular cellular network.

Field guide by symptom: what you hear, the cause, what to do

The table in the previous section gives a direction. Here we go into each symptom in depth — in the order in which it's worth checking, from cheapest to most expensive.

1. One-way audio (you hear them, they don't hear you, or the reverse). This is almost always the voice stream (RTP) failing to find its way while the signaling (SIP) works fine, which is why the call rings and gets answered. The causes, in order of how common they are: SIP ALG turned on in the router, which rewrites the addresses inside the messages; double address translation (a provider-supplied modem that also routes, followed by your own router); a firewall that blocks the voice port range going out. The fix: turn off SIP ALG, put the modem in bridge mode, and allow outbound traffic to the PBX's voice port range. If only outgoing calls suffer from this, suspect the firewall; if only incoming ones do, suspect the router's timeout.

2. Robotic or metallic voice, choppy words. Packet loss. Start with the question "one extension or all of them?" One extension: a bad cable, wall jack, or network port on the phone. Replace the cable before anything else. All of them: the line is saturated (someone is uploading), Wi-Fi, or a cheap switch that drops packets under load. The test: run a continuous ping during a bad call. If lines are lost, you've found it.

3. Echo. Here is a rule that saves hours: the person who hears the echo is not the one causing it. If you hear your own voice coming back, the echo is created on the other side: their speaker is close to their microphone, they have cheap headphones, or there is an old analog line there. If a customer complains about echo, the problem is on your side. Lower the handset or speaker volume, move the headset microphone away from your ear, and for a device connected through an analog adapter, check that echo cancellation is turned on. Echo is almost never a network problem.

Two more symptoms: noticeable delay, and disconnecting after 30 seconds

4. Delay: people talk over each other. When the other side's reply arrives late and each person keeps cutting the other off, the delay has crossed the threshold. Common causes in an office: Wi-Fi with power-saving mode (the phone or computer "falls asleep" between packets), a VPN connection that routes the voice to a distant country and back, a jitter buffer set too large on the phone, or a cellular backup line. The test: ping a nearby server. If the round-trip time is already over 100 milliseconds, the local network or the provider is at fault. If it is 20 and the call is still lagging, suspect the VPN and the device settings.

5. Disconnect after about 30 seconds, on every call. A clear sign of a configuration problem, and also a short explanation of why it is specifically 30. When a call is answered, the device sends a confirmation message. If the other side doesn't receive it within about 30 seconds, it assumes the call doesn't really exist and hangs up. That message gets lost when the router rewrites addresses (SIP ALG again) or when the firewall has closed the "door" that was opened. The same story appears in other versions: a disconnect at exactly 15 or 30 minutes is a periodic call refresh that didn't get through. The fix is always in the same place: SIP ALG off, a 180-second timeout for UDP connections in the router (the value we recommend), and a phone that sends a keep-alive every 20-30 seconds.

A test from a workstation, and what a healthy call looks like in numbers

Besides a continuous ping, every computer has two more tools that need no installation. tracert (on Windows) shows all the hops on the way to the destination and the time for each one. If the big jump in time is at the first hop, the problem is in your router or Wi-Fi. If it is farther along, it is at the provider or on the way. pathping does the same thing over several minutes and calculates the loss percentage for each hop, which is exactly the number that matters for voice. Run both from a computer connected by cable, otherwise you are measuring the wireless link and not the network.

A third tool is in the phone itself. Many IP phones can show a statistics screen during a call: codec, jitter, packet loss and round-trip time, as the device measures them at that moment. This is the most reliable data, because it is measured exactly where the problem is heard. When you call support, the numbers from this screen are worth more than any description.

Look at the "healthy" column: a 20 ms ping that jumps to 200 once every ten seconds is worse than a steady 80. Voice can live with a constant delay; it can't live with surprises.