What is VoIP?

What is VoIP

VoIP stands for Voice over Internet Protocol. IP is the protocol the Internet is built on — the system of rules that moves information from computer to computer. VoIP simply means: carrying voice the same way emails, websites and files are carried.

In everyday language it is also called "Internet telephony" or "IP telephony." For the person talking, it is a regular phone in every way: you pick up, dial, hear ringing and talk. The whole difference is in what happens along the way.

From a dedicated line to packets: the basic difference

In the old phone network, every call got its own circuit: a continuous electrical path between the two phones, reserved from the moment the call was answered until it ended. The path was occupied the whole time, even during seconds of silence. In this method, every call "holds" an entire line.

In VoIP there is no reserved path. The voice is chopped into small pieces called packets, and each packet is sent into the network with a destination address. Along the way they share the same connection with all other data: an email being sent, a software update, a website loading. During moments of silence almost nothing is sent, and the connection is free for other things.

That is the secret of VoIP: the same internet connection can carry several calls at once, together with all of the office's regular traffic. But it is also the source of its sensitivity. If the connection is too busy, voice packets may be delayed.

How it works, step by step

1. Capture. The microphone picks up the speech as a continuous sound wave.
2. Digitizing and compression. A codec samples the wave thousands of times a second, turns it into numbers and compresses them.
3. Packaging. The compressed voice is cut into packets, usually dozens of packets every second, and each one gets a sequence number and a timestamp.
4. Travel across the network. The packets pass through the router, the internet provider and the PBX to the other side.
5. Reassembly and playback. On the other side the packets are collected, put in order by sequence number, unpacked and played through the speaker.

The call itself (who calls whom, ringing, answering and hanging up) is managed by a separate protocol, SIP. The voice is carried by another protocol, called RTP. All of this happens in fractions of a second, so the call sounds continuous.

A short history of internet telephony

The idea of sending voice over a computer network is almost as old as computer networks themselves. As early as the 1970s, experiments were carried out sending speech over the American research network from which the internet grew. But the networks were slow, and the computers too weak to compress voice in real time.

The breakthrough to the public came in the mid-1990s. In 1995 an Israeli company released one of the first commercial programs for voice calls between computers over the internet. The quality was far from a phone call, and both sides had to be online at the same time, but the principle was proven.

In the late 1990s the standards arrived: first H.323, followed by SIP, which became the leading standard in the 2000s. At the same time, desk phones that connect to the network appeared, along with software PBXs and services that link the internet to the regular phone network, so that you can dial from VoIP to any phone in the world.

In the last decade VoIP is no longer an alternative but the foundation. Even calls that look "regular" today travel, for most of the way, as packets on the IP networks of the telecom companies.

Why organizations move to VoIP

• Internal calls at no cost between extensions, between branches and with employees at home.
• The number is not tied to an address. The same number and the same extension work anywhere there is internet.
• High audio quality: with a wideband (HD) codec the voice is fuller and clearer than on a regular line.
• Connection to other systems: recordings, call log, reports, email and CRM.
• Flexibility: an additional extension is a setting, not a new line that has to be run to the building.

What affects quality

A stable connection matters more than a fast one. A single call uses little bandwidth (tens of kilobits per second), and most internet connections today are enough for many calls at once.

What really hurts is latency and jitter: packets that arrive late, out of order, or not at all. The result sounds like choppy voice, a "robotic" voice, or echo. The common causes: a busy connection (for example, someone uploading large files), weak Wi-Fi, or a router that does not give priority to voice.

The solutions are well known: connect phones with a cable rather than Wi-Fi when possible, give priority to voice on the network (QoS), and sometimes separate the phones onto their own network (VLAN).

Common mistakes

• Upgrading speed when the problem is stability. A fast but unstable connection will give worse calls than a slow, stable one.
• A home router in a busy office. A router that was not designed for many calls may cause calls that are heard on one side only, or that drop after a few minutes. Usually this is a NAT problem or a router settings problem.
• Forgetting about power. An IP phone needs power, and so do the router and the switch. Without a UPS, a short power outage disconnects the whole office.

Numbers worth knowing

A few orders of magnitude help explain why VoIP behaves the way it does:

• Sampling. A regular phone codec samples the voice 8,000 times a second; a wideband (HD) codec, 16,000. That is the difference between a "phone" voice and a voice that sounds like it is in the room.
• Packets. In most setups each packet carries 20 milliseconds of voice, which means 50 packets per second in each direction. A five-minute call is about 30,000 packets.
• Bandwidth. An uncompressed codec uses about 64 kilobits per second for the voice itself, and about 80-90 with the network headers, in each direction. Ten simultaneous calls are less than one megabit. A compressed codec drops to a third of that.
• Latency. Up to about 150 milliseconds one way, the call sounds natural; above about 300, people start talking over each other, as in an old satellite call.
• Packet loss. One percent already sounds like slight breakups; above three to five percent the call is exhausting.
• Jitter buffer. The phone keeps a few dozen milliseconds of voice "in stock" to even out packets that arrive at an uneven pace. This adds a small delay, but prevents breakups.

The practical conclusion: you are not short of bandwidth. Even a modest connection is enough for dozens of calls. What is missing, when there is a problem, is order: packets that arrive on time and at an even pace. That is a matter of the router, the cables and prioritization, not of the package from the internet provider.

Troubleshooting: what you hear, what the cause is, what to do

Most VoIP problems can be heard. The table translates what you hear into the likely cause and the first action:

Two simple tools save guessing: a latency check (ping) to a fixed server, which shows whether the latency spikes; and swapping — the same phone on another jack, or another phone on the same jack. If the problem moves with the phone, it is the phone; if it stays at the jack, it is the network.

How a VoIP call reaches a phone that is not VoIP

A customer dialing from an old cell phone or from a copper line in a village does not speak VoIP. So how does the call still reach the cloud PBX? Along the way sits a gateway: a machine connected on one side to the telephone network and on the other to an IP network, which translates between the two — continuous voice becomes packets, and the dialing becomes a SIP message.

In a business, this gateway belongs to the provider, and the connection to it is called a trunk. In an office that wants to keep an old analog phone, a small gateway — an adapter — does the same translation in the opposite direction: the old phone gets a regular jack, and the adapter speaks SIP to the PBX.

What this means: from the caller's point of view, there is no "internet call" and "regular call". There is a phone number, and the networks translate between them along the way. That is why you can dial from VoIP to any phone in the world, and receive calls from any phone.