What is VoIP?

VoIP carries a voice conversation as data across an IP network, which means the same connection that handles your email and web traffic also handles your calls.[2] The traditional phone network, often called the PSTN, holds open a dedicated circuit for the length of a call. VoIP works the other way around: it samples your voice, compresses it, and breaks the result into small packets that share the network with everything else, then reassembles them at the far end.[2] Because the audio is just data, a VoIP call can start on a computer or a desk phone and still reach an ordinary landline or mobile number once it passes through a gateway that hands the call off to the public network.[1]

From sound waves to packets

An analog-to-digital converter samples the sound of your voice thousands of times per second and turns each sample into a number. A codec then compresses that stream so it uses less bandwidth, and the device wraps small slices of audio, typically about 20 milliseconds each, into packets addressed to the person you are calling.[7] Each packet finds its own way across the network, so a brief delay or a single lost packet affects only a sliver of speech rather than the whole call.

Why it reaches regular phones

Calls between two VoIP endpoints can stay entirely on IP networks from start to finish. To reach a number on the traditional network, the call passes through a media gateway that converts the IP audio back into a standard telephone signal.[1][2] That is why a service can offer to keep your existing phone number and still let you call anyone, regardless of what technology sits on the other end.

Two different jobs run in parallel on every VoIP call: one protocol sets up the conversation and another carries the sound. The setup work, called signaling, is handled most often by the Session Initiation Protocol, or SIP, which locates the other party, negotiates the audio formats both sides support, and then rings, connects, or ends the call.[3] Once both ends agree, the audio flows in a separate stream using the Real-time Transport Protocol, or RTP, which timestamps and sequences each packet so the receiver can play speech back in order and smooth over minor network hiccups.[4]

What a codec does

A codec decides how the audio is compressed, and that choice trades sound quality against bandwidth. The classic option, ITU-T G.711, sends uncompressed voice at 64 kilobits per second and delivers the familiar quality of a traditional phone line.[5] With the headers that IP, UDP, and RTP add to every packet, a single G.711 call actually uses closer to 85 to 87 kilobits per second in each direction.[7] G.729 compresses the same speech to about 8 kilobits per second for networks where bandwidth is tight, at a modest cost in fidelity.[7]

Newer codecs and wideband audio

The Opus codec, defined in 2012, adapts on the fly and ranges from low-bitrate narrowband speech at 6 kilobits per second up to high-quality stereo at 510 kilobits per second, which is why many softphones and browser-based calling tools now use it.[6] Wideband codecs like Opus carry a broader slice of the audio spectrum than G.711, so voices sound fuller and are easier to understand, especially on busy conference calls.

An analog telephone adapter, or ATA, lets you plug an ordinary corded phone or a fax machine into a VoIP service. The adapter sits between the handset and your internet connection, converting the analog signal from the phone into IP packets and back again. This form matters most during a transition, because it keeps equipment you already own working while the rest of the system moves to VoIP. A common use is a fax machine or an alarm panel that a business is not ready to replace. If you go this route, confirm the setup handles fax and alarm signals reliably, since heavily compressed voice codecs can distort those tones; an uncompressed path such as G.711 is generally the safer choice for them.[5]

An IP desk phone is a hardware phone that connects directly to your data network and runs VoIP without any adapter. It looks and behaves like a traditional office phone, but it registers to the phone system over the network using SIP and sends audio as RTP packets.[3][4] This form suits people who want a dedicated device on the desk rather than software on a computer, such as reception desks, contact centers, or anyone on calls for most of the day. Many models draw power over the same Ethernet cable that carries their data, using Power over Ethernet, so there is no separate power adapter at each desk. When you choose handsets, check that they support the codecs and the encryption your provider requires, because older models sometimes lack current security features.

A softphone is calling software that turns a laptop, tablet, or smartphone into a VoIP endpoint. It gives you a work number and full calling features on whatever device you already carry, which is why remote and hybrid teams depend on it for mobility.[2][12] Because the software travels with the user, a call can follow someone from a desk to a home office to an airport without changing their number. The trade-off is that call quality now depends on whatever network the device happens to be on, so a weak home Wi-Fi connection or a congested cellular link tends to show up as choppy audio. For staff who work in noisy or bandwidth-limited spots, pairing the app with a headset and a codec that tolerates packet loss, such as Opus, usually helps.[6]

SIP trunking is a VoIP connection that links an on-premises PBX, the private phone system many organizations already run, to the outside world over the internet instead of over legacy telephone circuits.[3] It replaces the older analog or digital trunk lines a business used to buy from a phone company, often at lower cost and with the ability to add or remove call capacity in software rather than by ordering physical lines.[2] This form fits organizations that have invested in a PBX and want to keep it while retiring aging circuits. A SIP trunk is measured in concurrent call paths, so a practical first step is to count how many simultaneous calls you actually make at your busiest hour and size the trunk to that number, with a little headroom, rather than to your total count of phones.

A hosted or cloud PBX moves the entire phone system into a provider's data center, so there is no call-routing hardware to run on site. You connect IP phones or apps to the provider over the internet, and the provider handles the servers, the upgrades, and the connection to the public phone network.[2] This form suits teams that would rather not buy and maintain their own equipment, and it is often sold as part of a broader unified communications package that adds video meetings, messaging, and voicemail-to-email in one place.[12] Because the system lives off site, its reliability depends on both your internet connection and the provider's uptime, so it is worth reviewing a provider's service-level commitments and how calls fail over if your primary internet link goes down.

US rules treat VoIP differently depending on whether it connects to the traditional phone network. The FCC defines 'interconnected VoIP' as a service that supports real-time two-way voice, requires broadband and IP-compatible equipment, and lets users both place calls to and receive calls from the public switched telephone network.[1] Services that meet that definition take on duties resembling those of a phone company, while VoIP that only connects app to app falls largely outside these rules.

911 and caller location

Interconnected VoIP providers must route 911 calls to the right emergency center and deliver the caller's call-back number and location, and they have to tell customers plainly where 911 might not work, such as during a power or internet outage.[8] Under Section 506 of the RAY BAUM'S Act, fixed interconnected VoIP services have had to supply an automated dispatchable location, meaning a validated street address plus details like a floor or suite, since January 6, 2021, with non-fixed services following on January 6, 2022.[9] A related measure, Kari's Law, requires multi-line phone systems to allow direct 911 dialing with no prefix and to notify a front desk or security team, for equipment put into service after February 16, 2020.[9]

Caller ID authentication

Since June 30, 2021, the FCC has required large voice providers to run STIR/SHAKEN in the IP portions of their networks, a framework that cryptographically signs caller ID so the receiving carrier can check that a number has not been spoofed.[10][11] Smaller providers with 100,000 or fewer lines were given until June 30, 2023.[11] The point of the framework is to cut down on the illegal robocalls that spoofed numbers make so easy.

VoIP depends on your internet connection, and that dependency creates a few limits a traditional line does not have. Naming them up front makes them easier to plan around.

Power and internet outages

A landline draws power from the phone company's central office, so it usually keeps working in a blackout. A VoIP phone relies on your local power and your internet link, so a router without a battery backup goes dark when the electricity does. Providers are required to warn you about this because it directly affects 911 access.[8]

Location for 911

Because a VoIP account can travel, the network does not always know where a call is physically coming from. Fixed services now supply an automated location, but a nomadic softphone may fall back on the address you registered, so keeping that address current matters for emergency calls.[8][9]

Call quality and the network

Voice is sensitive to delay, jitter, and packet loss, and a network that is fine for email can still produce choppy calls under load. Prioritizing voice traffic with quality-of-service settings, and keeping enough upstream bandwidth for the number of concurrent calls, is what keeps the audio clean.[7]

Security

Because calls are data, they can be intercepted or attacked like any other network traffic, so encrypting both the signaling and the media, alongside the caller ID authentication now required on public networks, has become part of a normal deployment.[10]

VoIP is a method for carrying calls, not a single product or brand. A few common mix-ups are worth clearing up.

It is not one app

Consumer apps that make internet calls are one way to use VoIP, but the term covers a whole category that also includes desk phones, carrier trunks, and full business phone systems.[2] Two VoIP services can share none of the same software and still both be VoIP.

It is not the same as your cellular voice

Mobile calls on a modern network often travel as Voice over LTE, an IP voice technology the mobile carrier runs inside its own network. It shares the packet idea with VoIP but is not the interconnected VoIP service the FCC rules describe, and you do not provision it the way you provision a business VoIP account.[1]

It is not the same as UCaaS

Unified Communications as a Service bundles calling with video meetings, chat, and file sharing. VoIP is the voice layer inside that bundle, so every UCaaS platform uses VoIP, yet plenty of VoIP is sold as plain calling without the wider collaboration tools.[2][12]

It is not automatically a phone company service

Whether a VoIP service carries phone-company duties depends on that interconnected VoIP test, so an app that only calls other users of the same app is not held to the same 911 and universal-service rules as a service that reaches the public network.[1]