---
title: "VoIP Glossary - Know Your Terms"
slug: "voip-glossary-know-your-terms"
updated: 2025-08-12T15:04:14Z
published: 2025-08-12T15:04:14Z
canonical: "docs.ucstack.io/voip-glossary-know-your-terms"
stale: true
---

> ## Documentation Index
> Fetch the complete documentation index at: https://docs.ucstack.io/llms.txt
> Use this file to discover all available pages before exploring further.

# VoIP Glossary - Know Your Terms

**Analog audio signals**

Analog audio signals transmit voice data over telephone lines by modulating the frequency of sound waves to replicate pitch accurately. This same principle applies to radio wave transmissions.

**ATA (Analog Telephone Adaptor)**

An ATA is a hardware device that connects traditional telephones to the internet via a high-speed broadband connection. It converts analog voice signals into IP packets, supplies dial tone, and manages call setup.

**Audio encoding**

The ITU has standardized multiple audio codecs for H.323, all compatible with SIP (which is codec-agnostic). Key codecs include:

- G.711 – 3 kHz audio encoded at 64 kbps, using PCM (the standard for traditional telephone networks).
- G.722 – High-quality 7 kHz audio at 48, 56, or 64 kbps. Lower-bitrate variants include G.722.1 (24/32 kbps) and G.722.2 (~16 kbps).
- G.723.1 – Optimized for low-bitrate speech compression (5.3–6.3 kbps).
- G.728 – 3.4 kHz audio at 16 kbps, using small packet sizes (0.625 ms) to minimize delay.
- G.729 – A modern 8 kbps codec with 15 ms packet sizes. Variants G.729 and G.729A differ only in mathematical implementation.
- Speex – Open-source, patent-free, variable-bitrate (2.15–44.2 kbps).
- GSM6.10 – Open-source at 13.3 kbps, though currently under patent dispute.

**Audio Menu**

A recorded phone menu offering verbal choices, commonly used in automated attendants, IVRs, and fax-on-demand systems. Users respond via voice commands or touch-tone inputs.

**Audio Response Unit (ARU)**

A computer telephony system using voice store-and-forward technology. Passive ARUs play pre-recorded messages, while interactive ARUs respond dynamically to caller input.

**Audio Teleconferencing**

Originally reliant on PBX conferencing circuits, early teleconferencing suffered from voice degradation and limited capacity. Dedicated conference bridges improved quality but required operator intervention. Modern PC-based systems allow self-managed conferences via touch-tone phones, email scheduling, or web browsers. Emerging solutions enable endpoints to handle mixing locally, eliminating centralized servers.

**Bandwidth**

The data transmission capacity of a communication channel, measured in bits per second (bps) for digital signals or Hertz (Hz) for analog. It can also refer to the range of frequencies a signal occupies.

**Call Duration**

The elapsed time from when a phone goes off-hook to when it is hung up after a call.

**Circuit-Switched Networks**

Used since 1878, these networks establish dedicated point-to-point connections for each call, limiting efficiency since no other traffic can share active switches.

**Client (Softphone Client)**

Software installed on a user’s device to enable internet-based calling.

**Call Hunting**

A feature that redirects inbound calls past busy signals or through multiple numbers until answered.

**Class 5 (Telephony) Switch**

A local telephone exchange providing dial tone, calling features, and digital services to subscribers. In SIP/VoIP/IMS networks, IP-based Class 5 switches (or Class 5 Application Servers) play a key role.

**Clipping**

The loss of speech segments, causing parts of words to drop out.

**Cloud PBX**

A hosted PBX service delivered over the internet (e.g., via AWS, Azure). Features:

- Scalability: No on-premise hardware.
- Remote management: Administered via web portals.
- Integrates with SIP trunks, CRM tools, and UCaaS platforms.

**Codec**

Short for Compressor-Decompressor or enCOder/DECoder, a codec transforms data streams for transmission, storage, or encryption. Essential for videoconferencing and streaming media.

**Compression**

Reducing data size for efficient storage or transmission, particularly useful for large audio, video, and graphics files.

**Conference Bridge**

A device connecting multiple callers, with features like noise balancing, voice-activated muting, and touch-tone rejection. VoIP bridges use SIP/Megaco for control and RTP for media transport.

**Data Compression**

Shrinking files by identifying binary patterns, reducing bandwidth and storage needs. Effective compression can cut data to 40% of its original size (up to 90% for graphics).

**Dial-Tone Delay**

The lag (in milliseconds) between lifting a handset and hearing a dial tone.

**DNS (Domain Name System)**

Translates domain names into IP addresses. Recent additions like SRV/NAPTR and ENUM records support SIP/VoIP.

**Dual-Tone Multifrequency (DTMF)**

The touch-tone dialing system, assigning unique dual-frequency pairs to each key for easy detection.

**E.164**

The international numbering format for phone numbers (e.g., +1 555 123 4567). Critical for VoIP and PSTN interoperability.

**Echo Cancellation**

A feature to eliminate echo in voice calls caused by signal reflections. Critical for VoIP and audio conferencing.

**Emergency 911 Calls**

A North American emergency number routed to dispatchers for police, fire, or medical assistance.

**ENUM (E.164 Number Mapping)**

A DNS-based system that maps E.164 phone numbers to internet addresses (e.g., SIP URIs), enabling direct VoIP routing without PSTN gateways.

**Fax over IP (FoIP)**

Sending faxes over IP networks using protocols like T.38 to ensure reliability.

**Find-Me/Follow-Me**

A call-routing feature that rings multiple devices (e.g., cell, home, office) simultaneously.

**Frame Relay**

A packet-switching method using bandwidth on demand, capable of voice transmission with proper management.

**Full Duplex**

Simultaneous two-way communication without quality loss.

**Gateway (VoIP)**

Converts PSTN voice/fax calls to IP packets in real time.

**H.323**

An ITU standard for real-time voice/video conferencing over IP, supporting media gateways for packet conversion.

**High-Availability**

Designs ensuring constant system access, often using failover clusters.

**Interactive Voice Response (IVR)**

A phone-based system where callers interact via touch-tone or voice commands to query databases, hear synthesized responses, or update records.

**Internet**

The global network combining academic, government, and commercial subnetworks, serving as the backbone for the World Wide Web.

**Internet Congestion**

Slowdowns caused by excessive data on low-bandwidth or high-latency networks, leading to packet loss and degraded service.

**Internet Telephony (IP Telephony)**

Voice transmission over the internet, involving:

- Client-side voice digitization.
- Internet-based packet routing.
- IP address lookup for connection.
- PSTN gateways for phone-to-internet calls.

**IP (Internet Protocol)**

Defines data packet routing between sources and destinations.

**IP Address**

A unique identifier for devices on a network, either static or dynamically assigned.

**IP Mapping**

Geolocating IP addresses for network management.

**IP Phone (VoIP/SIP Phone)**

A device converting voice to digital packets for internet calls, supporting protocols like SIP/H.323 and codecs like G.711/H.261.

**ISP (Internet Service Provider)**

A business offering internet access, hosting, and related services.

**ITU (International Telecommunication Union)**

A UN-affiliated body setting global telecom standards.

**Jitter**

Signal timing fluctuations due to delayed/early packet arrivals.

**Kbps (Kilobits per Second)**

A data transfer rate metric (1 Kbps = 1,000 bits/sec).

**Lag**

Delays in data round-trip times, often from network or processing inefficiencies.

**Latency**

The delay between a data request and its fulfillment.

**Mean Opinion Score (MOS)**

A subjective 1–5 scale rating voice quality.

**Messaging**

Store-and-forward systems (e.g., voicemail, fax mail) for asynchronous communication.

**NANP (North American Numbering Plan)**

The system governing area codes and phone numbers in the U.S., Canada, and nearby regions.

**NAT (Network Address Translation)**

A method to map private IP addresses to a public IP (e.g., in home routers). Can cause issues for SIP/VoIP without STUN/TURN servers.

**Opus Codec**

A high-efficiency, low-latency audio codec for VoIP and streaming. Supports variable bitrates (6–510 kbps) and dynamic adjustment to network conditions.

**Packet**

A data unit containing payload, origin/destination details, and synchronization data, routed dynamically across networks.

**Packet Loss**

Data packets dropped due to congestion or errors.

**Packet Switching**

Efficient data transmission via routed packets, adaptable for voice/video on high-speed networks.

**PBX (Private Branch Exchange)**

A business phone system managing internal/external calls, evolving to support automated attendants.

**Peer-to-Peer (P2P)**

Decentralized networks where nodes act as both clients and servers.

**POP (Point of Presence)**

A carrier’s local network junction, or an email retrieval protocol.

**POTS (Plain Old Telephone Service)**

Basic analog phone service.

**Protocol**

Rules governing data transmission, including error handling and compression.

**PSTN (Public Switched Telephone Network)**

The global circuit-switched phone network.

**QoS (Quality of Service)**

Network performance metrics like delay and uptime.

**Real Time**

Minimally delayed communication (e.g., live calls).

**Router**

A device directing data packets between networks.

**Sample Rate**

Audio samples per second (Hz/kHz), determining bandwidth.

**Service Provider**

An entity offering and maintaining application services.

**SIP (Session Initiation Protocol)**

An IETF standard for interactive sessions (voice, video, messaging).

**SIP Phone**

A phone using SIP for internet calls, often with voicemail/blocking features.

**SIP Trunking**

A virtual alternative to traditional phone lines, using SIP to connect a PBX to the PSTN via the internet. Benefits:

- Cost-effective: Eliminates physical PRI lines.
- Flexible: Supports concurrent calls over a single connection.

**Softphone**

Software enabling calls via PCs/mobile devices with headsets.

**Softswitch**

Software replacing hardware switches at circuit-packet network junctions.

**STUN/TURN/ICE**

Protocols to help VoIP devices traverse NAT/firewalls:

- STUN: Discovers a device’s public IP.
- TURN: Relays media if direct connection fails.
- ICE: Combines STUN/TURN for optimal routing.

**TCP (Transmission Control Protocol)**

A core internet protocol ensuring reliable data delivery.

**Telephony**

Telephony refers to the technology and systems used for voice communication over distances, traditionally via analog signals transmitted through physical networks like the PSTN (Public Switched Telephone Network)

**UCaaS (Unified Communications as a Service)**

Cloud-delivered services combining VoIP, video conferencing, messaging, and collaboration tools (e.g., Microsoft Teams, Zoom Phone).

**Virtual Phone Number**

A telephone number not directly tied to a physical line. Features include:

- Cloud-based routing (e.g., forwarding to mobile, desk phone, or VoIP client).
- Global availability (e.g., local numbers in foreign countries).
- Used for IVR systems, call centers, and privacy protection.

**VoIP (Voice over Internet Protocol)**

Technology for transmitting voice calls over IP networks (e.g., the internet) instead of traditional PSTN. Key traits:

- Uses codecs (e.g., G.711, Opus) to compress/decompress audio.
- Relies on protocols like SIP, RTP, and WebRTC.
- Enables features like video conferencing, instant messaging, and call analytics.

**WebRTC (Web Real-Time Communication)**

An open-source framework enabling browser-based voice/video calls and data sharing without plugins. Integrates with SIP and VoIP systems.

**Wi-Fi (Wireless Fidelity)**

A wireless networking technology that allows devices to connect to a local area network (LAN) using radio waves. Commonly used for VoIP calls, internet access, and data transfer without physical cables. Operates under IEEE 802.11 standards (e.g., Wi-Fi 6/802.11ax).