Voice over Internet Protocol (VoIP) is a way of carrying phone calls as packets of data over IP networks, such as a company LAN, a private WAN or the public internet, instead of over the dedicated analog or digital circuits of the traditional phone network. Your voice is digitized, compressed by a codec, split into packets and reassembled at the other end. VoIP is the technology underneath most business phone services sold today, including IP PBXs, hosted PBX, UCaaS and SIP trunking.
At a glance
- VoIP turns calls into data packets, so voice can share the same network as other business traffic.
- A signaling protocol, usually SIP, sets up and ends calls; a codec turns audio into data and back.
- Call quality depends mainly on latency, jitter and packet loss across every network the call crosses.
- VoIP calls can reach ordinary phone numbers when the service connects to the public phone network.
- Emergency calling, power backup and analog devices (fax, alarms) need planning, because there is no copper line behind the phone.
What problem it solves
The traditional phone network dedicated a circuit to each call for its whole duration, and a business needed separate phone wiring, phone lines from the carrier, and phone-specific hardware at each site. Adding lines, moving desks or opening an office meant ordering circuits and waiting for installers.
VoIP lets voice run over the data network a business already has. Phones and apps can connect from most locations with a suitable connection, numbers need not be tied to a building, and sites can share one phone system. Carrier connections can be bought as SIP trunks or bundled into a hosted service, which in many cases costs less and scales more easily than analog or PRI lines. It also makes features such as voicemail-to-email, call recording and mobile apps easier to deliver.
How it works
Signaling. When you dial, the phone or app sends a request, most commonly using the Session Initiation Protocol (SIP), to the phone system, which locates the destination and sets up the call.
Media. Once the call is connected, audio is encoded with a codec and sent as a stream of small packets, typically using the Real-time Transport Protocol (RTP). Codecs trade bandwidth against audio quality; wideband codecs give clearer audio when both ends support them.
Quality. Voice is sensitive to delay, variation in delay (jitter) and lost packets. On networks you control, quality of service (QoS) settings can prioritize voice traffic. Over the public internet, no one guarantees end-to-end priority, so providers rely on good access connections, nearby points of presence and jitter buffers.
Reaching phone numbers. To call an ordinary number, a VoIP service hands the call to the public switched telephone network (PSTN) through a carrier interconnection. A business that runs its own phone system usually buys SIP trunks for this; hosted PBX and UCaaS providers include it.
Endpoints. VoIP can reach users on IP desk phones, conference phones, softphones on computers, mobile apps and browsers. Analog devices can connect through adapters, with the caveats above.
Caller ID and emergency calls. In the US, carriers are required to authenticate caller ID on calls that travel over IP networks, using a framework known as STIR/SHAKEN, and requirements differ by country. Because a VoIP phone can move, the service also needs each user’s location for E911.
When it matters for buyers
- When replacing an on-premises phone system or moving it to a hosted or UCaaS service.
- When analog or PRI lines are being repriced or retired by the carrier.
- When call quality complaints start, which often trace to the network more than the phone system.
- When staff work from home or travel and need their business number with them.
- When opening offices, since a new site can often use its network connection instead of ordering its own phone lines.
Questions to ask vendors
- What bandwidth, latency, jitter and packet loss does your service need per call, and do you test our sites before cutover?
- Which codecs do you support, and do calls stay on your network or cross the public internet?
- How do you monitor call quality, and what data can we see when users report problems?
- What happens to inbound calls when a site loses internet or power?
- How do you handle 911 location for desk, remote and mobile users?
- Which analog devices, such as fax, alarm panels and door phones, have you tested, and how?
- How do you authenticate caller ID, and how do you help keep our outbound calls from being labeled as spam?
How it differs from SIP
VoIP is the broad idea of carrying calls over IP networks. SIP is one protocol used within VoIP: it sets up, changes and ends calls, while other protocols carry the audio itself. Most business VoIP uses SIP, but some systems use proprietary or other protocols internally. “SIP trunking” is a carrier service that uses SIP to connect a phone system to the public network.
Our Unified Communications as a Service solution page covers how buyers compare cloud phone services built on VoIP.
