Reference

VoIP Glossary

By the VoIP Cost Guide Editorial Team  |  Last updated: October 2026

The working vocabulary of VoIP, defined the way the terms are used in this site's guides — short, precise, and linked to the guide that covers each topic in depth.

ATA (Analog Telephone Adapter)

A device that connects a traditional analog phone or fax machine to a VoIP network, converting between analog signals and SIP/RTP.

Bandwidth

The capacity of a connection in bits per second. A G.711 call needs roughly 87 kbps each way on Ethernet once all overhead is counted.

CCaaS

Contact Center as a Service: a cloud-hosted contact-center platform licensed per agent, covering queues, routing, recording and analytics.

Codec

The algorithm that encodes voice for transport. Common VoIP codecs include G.711 (uncompressed quality, high bandwidth), G.729 (compressed) and Opus (adaptive).

DID (Direct Inward Dialing)

A public phone number that routes directly to an extension, user or queue on a phone system without passing a receptionist.

DSCP

Differentiated Services Code Point: the marking in an IP packet header that tells network devices how to prioritize it. Voice media is normally marked EF (46).

DTMF

Dual-Tone Multi-Frequency: keypad tones used by IVR menus and voicemail. In VoIP it travels as RFC 2833/4733 events, SIP INFO, or inband audio.

E911

Enhanced 911: emergency calling that delivers the caller's number and registered location to the public safety answering point.

Echo

Hearing your own voice delayed. Causes include acoustic coupling at the far end, analog hybrid conversion, and excessive network delay.

Hosted VoIP

A phone system operated by a provider in the cloud and licensed per user, instead of a PBX installed on your premises.

ICE

Interactive Connectivity Establishment (RFC 8445): a framework that finds a working media path between endpoints using STUN and TURN candidates.

IP PBX

A private branch exchange that switches calls using IP and SIP rather than legacy TDM circuits — on-premise or cloud-hosted.

Jitter

Variation in packet arrival times. Excess jitter exhausts the receiver's jitter buffer and produces choppy, garbled audio.

Latency

One-way delay between mouth and ear. ITU-T G.114 guidance: keep it under about 150 ms for good conversational quality.

MOS

Mean Opinion Score: a 1–5 quality scale (5 = excellent). VoIP dashboards usually show an estimate derived from loss, jitter and delay via the E-model.

NAT

Network Address Translation: rewriting of private IP addresses at a router or firewall. NAT is the single most common cause of VoIP audio problems.

Packet loss

Packets that never arrive. Even 1–3% loss on a voice stream is audible as gaps, clicks or robotic artifacts.

PBX

Private Branch Exchange: the phone system that connects a business's extensions to each other and to the public network.

PoE

Power over Ethernet: delivering phone power through the network cable (802.3af/at). An undersized PoE budget reboots phones under load.

Porting (LNP)

Local Number Portability: moving your existing phone numbers from one provider to another, coordinated through an industry process.

PSTN

Public Switched Telephone Network: the global telephone network that VoIP calls still reach through gateways and SIP trunks.

QoS

Quality of Service: the set of marking, queuing and policing mechanisms that protect voice traffic from bulk data on a network.

RFC 3261

The IETF specification defining SIP — requests, responses, transactions and dialogs. The foundation document of modern VoIP signaling.

RTP

Real-time Transport Protocol (RFC 3550): carries the encoded voice itself, with sequence numbers and timestamps used to detect loss and jitter.

SBC

Session Border Controller: a device or service at a network edge that normalizes SIP, fixes NAT problems, enforces security and can transcode media.

SIP

Session Initiation Protocol: the signaling protocol that sets up, modifies and tears down VoIP calls and registrations.

SIP ALG

SIP Application Layer Gateway: a router/firewall helper that rewrites SIP messages. Frequently buggy; a leading cause of registration and audio faults.

SIP trunk

A virtual circuit from a provider that carries calls between a phone system and the PSTN, priced per channel, per minute, or unmetered.

SRTP

Secure RTP: encryption and authentication for voice media, usually keyed via DTLS or SDES alongside TLS-protected signaling.

STIR/SHAKEN

The caller-ID authentication framework US carriers use to sign calls and fight spoofing; attestation level affects how calls are labeled and delivered.

STUN

Session Traversal Utilities for NAT (RFC 8489): lets an endpoint learn its public IP/port mapping so media can find its way back.

T.38

The ITU-T standard for fax over IP networks, relaying fax tones as data instead of audio so they survive VoIP compression.

UCaaS

Unified Communications as a Service: cloud bundles combining phone, meetings, messaging and collaboration in one per-user license.

UDP timeout

How long a firewall keeps an idle UDP mapping open. Short timeouts close the pinholes RTP and registrations depend on.

VoIP

Voice over Internet Protocol: carrying telephone calls as data packets over IP networks instead of dedicated telephone circuits.

Looking for a term used in a specific fault? The SIP response code library decodes the numbers, and What Is SIP? and RTP Explained cover the two protocols everything else hangs from.