27 Telephony Terms Explained

Like many specialized industries, business communications has developed its own vocabulary over time. Just as doctors, lawyers, and software developers speak in jargon that might sound like gibberish to outsiders, telecom professionals toss around language that can be equally baffling. If you’re not knee-deep in the world of switches, trunks, and protocols on a daily basis, understanding the lingo can feel like trying to crack a secret code.

To cut through the confusion, we’ve put together a clear and concise glossary of essential telephony terms and commonly used telephone acronyms. Whether you’re a seasoned IT manager, a small business owner evaluating phone service providers, or just someone curious about how telephone calls actually travel across modern networks, this guide is designed to demystify the terminology that powers everything from your office phone line to your unified communications platform.

ACD (Automatic Call Distribution)

The telephony acronym ACD is a smart telecommunications interface standard used in call centers and contact centers to efficiently manage incoming call traffic. Instead of sending calls to the next available agent at random, an ACD system routes incoming calls based on predefined rules like agent skillsets, caller history, or menu selections. Each inbound call reaches the most qualified contact center agent, improving response times and customer satisfaction.

ATA (Analog Telephone Adaptor)

An ATA is a hardware device that connects analog (non-IP enabled) telephones, PBX systems, fax machines, door alarms and similar devices to digital systems or an internet-based telephony network. ATAs are used to connect legacy hardware to more modern telecommunications equipment like VoIP systems and SIP trunks.

Bandwidth

Bandwidth refers to the capacity of a telecommunications network or internet connection to transmit data over a specific period of time. In digital environments, it’s typically measured in bits per second (bps), while analog systems use cycles per second, or Hertz (Hz). Higher bandwidth means more data can travel across your computer network or telephone network, making it essential for supporting high-quality voice calls, video conferencing, and other bandwidth-intensive applications.

Broadband Telephony

Broadband telephony is the delivery of voice calls over a high-speed internet connection instead of traditional telephone lines. Often used interchangeably with terms like VoIP (Voice over Internet Protocol), internet telephony, and IP telephony, this method allows users to make telephone calls using digital data signals rather than analog ones. Broadband telephony is a cornerstone of modern business communication, enabling features like unified communications, video conferencing, and seamless integration with private branch exchange (PBX) systems, all while reducing reliance on wired telecommunication networks.

CDR (Call Data Record)

A Call Data Record is a detailed log of a telephone call stored within a telecommunications network or VoIP system. It includes essential call metrics like the phone numbers of the calling and called parties, the start and end time of the call, call duration, and the originating and terminating switches. CDRs help businesses monitor network traffic, analyze voice call patterns, and manage billing or customer relationship management (CRM) integrations. They’re an important component of enterprise telephony systems and are especially useful for contact center analytics and compliance tracking.

Codec

A codec, a telephony term that stands for coder-decoder, converts an audio signal (your voice) into compressed digital form for transmission (VoIP) and then back into an uncompressed audio signal for replay. It’s the secret sauce of VoIP. Different codecs have different levels of compression. The highly compressed signals require less internet bandwidth, while less compression is associated with better voice quality.

CTI (Computer Telephony Integration)

Computer Telephony Integration is the technology that connects telephone systems with computer networks, allowing for powerful automation in contact centers and business telephone systems. CTI enables features like screen pops with caller information, automated call routing, call transfer, and CRM integration in real time. By linking voice calls with software applications, CTI enhances the efficiency of both customer service teams and office phone lines, creating a more unified communications experience.

DID (Direct Inward Dialing)

Direct Inward Dialing is a telephone service feature that allows external callers to reach specific extensions within a private branch exchange (PBX) system without going through a receptionist or automated menu. With DID, each user or department can have a unique phone number that routes calls directly to their VoIP phone or IP-enabled device.

E911

E911, short for Enhanced 911, is a telephony service that automatically transmits a caller’s physical address to emergency responders when 911 is dialed. Calls are quickly routed to the nearest Public Safety Answering Point (PSAP), significantly enhancing response times. Modern implementations often include Nomadic E911, which enables users to designate any valid address, which is ideal for businesses with mobile or remote workforces and individuals using VoIP phones.

IP (Internet Protocol)

Internet Protocol is a standard that defines how data is transmitted between the source device and the destination. It’s the network layer protocol in the TCP/IP communications protocol suite. Telecommunications hardware designed for use over the internet is commonly called “IP enabled.” IP-enabled phones may be referred to as “SIP Phones” or “VoIP Phones.”

IVR (Integrated Voice Response)

Interactive Voice Response is a telephony system feature that automates the handling of inbound calls through voice prompts and keypad inputs. Often used in call centers and business telephone systems, IVR can answer incoming calls, perform real-time database lookups, route callers to the appropriate department, or complete transactions without human intervention.

Jitter

Jitter is used to describe a short fluctuation in the transmission of a voice signal. In SIP trunking, it may result from an abrupt variation in signal characteristics, such as when a data packet arrives either ahead or behind a standard clock cycle.

Latency

Latency is the time between the moment a voice packet is transmitted and the moment it reaches its destination. This delay may be in nanoseconds, but it’s still used to judge the efficiency of networks. Latency in SIP trunking can lead to poor quality calls.

LEC (Local Exchange Carrier)

A Local Exchange Carrier is the regional telephone service provider responsible for managing and routing telephone calls within a specific geographic area. LECs form the backbone of wired telecommunication networks, handling everything from plain old telephone service (POTS) lines to digital subscriber line (DSL) internet access. In many cases, LECs interconnect with VoIP services and larger telecommunications networks to ensure seamless local and long-distance calling.

PBX (Private Branch Exchange)

A PBX is a private telephone network used within a business to manage internal and external phone calls. It includes both the hardware and software needed to handle call routing, voicemail, call transfer, and more. When connected to an internet connection, a traditional PBX becomes an IP PBX (Internet Protocol PBX), supporting VoIP phones and enabling advanced features like unified communications and CTI.

POTS (Plain Old Telephone Service)

Plain Old Telephone Service refers to the traditional analog phone line system that has been in place for decades. A typical POTS setup includes a single phone line connected to a single telephone number. It’s often used for home phones, fax machines, or as backup lines for business continuity. While still reliable, POTS is being steadily replaced by more flexible and scalable VoIP phone systems.

PRI (Primary Rate Interface)

Primary Rate Interface is a type of digital circuit used to connect business telephone systems to the Public Switched Telephone Network. It supports multiple voice channels over a single physical line and was once the standard for enterprise telephony. Today, PRI is increasingly being replaced by SIP (Session Initiation Protocol) trunking, which offers the same functionality using an internet connection and provides greater flexibility for scaling and managing phone service.

PSTN (Public Switched Telephone Network)

The PSTN is the global system of interconnected voice networks, including local exchange carriers, long-distance providers, and international operators, that enables traditional telephone calls. It encompasses everything from undersea telephone cables to mobile networks, allowing voice traffic to travel seamlessly from one part of the world to another.

QoS (Quality of Service)

The telephony term Quality of Service is a network configuration that prioritizes certain types of data, especially voice traffic, over others to maintain call quality. It’s especially important for VoIP phones and SIP trunking, where real-time voice data must be delivered with minimal latency or packet loss. Enabling QoS settings on your router helps ensure that your office phone line or contact center maintains crystal-clear audio, even when your network is handling heavy traffic.

SIP (Session Initiation Protocol)

SIP is an industry-standard signaling protocol used to initiate, manage, and terminate real-time communications between devices over a computer network. It’s the backbone of many unified communications platforms, enabling voice calls, video conferencing, instant messaging, and other multimedia services. SIP works hand-in-hand with Voice over Internet Protocol to deliver reliable, flexible, and scalable digital communications across private telephone networks and public internet infrastructure.

SIP Phone/IP Phone

SIP phones, also known as IP phones, are desktop handsets designed to work with internet-based telephone systems. Instead of using traditional telephone lines, these devices convert voice into data packets and transmit them over a broadband internet connection. SIP phones are a key component in VoIP services and are commonly used in modern office phone systems, enabling features like voicemail-to-email, call forwarding, and seamless call transfer across locations.

SIP Trunkinig

SIP Trunking is a voice service that uses Session Initiation Protocol to connect your IP-enabled PBX or VoIP gateway to the PSTN via the internet. Unlike traditional analog lines, SIP trunking leverages your existing internet connection to consolidate voice and data on a single telecommunications network. This setup allows for more efficient outbound and inbound calls, reduces costs, and scales easily.

SIP Channel

A SIP channel represents a single digital pathway that supports one concurrent telephone call—either inbound or outbound—over a SIP trunk. It’s the virtual version of a traditional telephone line. While a SIP trunk can support multiple SIP channels, each channel handles just one call at a time. Businesses can scale their SIP channels based on call volume, ensuring their phone system can handle peak network traffic without overpaying for unused capacity.

Soft Switch

A soft switch is the software equivalent of a physical telephone switchboard. Internet-based telephony and even some traditional telecommunication networks use soft switches to manage phone call connections.

Softphone

A softphone (also known as a soft client) is a software program for making telephone calls over the internet using a computing device rather than a traditional telephone handset. The application can be run on a desktop PC, laptop, tablet or cell phone.

Telephony

Telephony refers to any technology involving the development, application, and deployment of telecommunication services for the purpose of electronic transmission of voice, fax, or data between distant parties. The history of telephony is intimately linked to the invention and development of the telephone.

VoIP (Voice over Internet Protocol)

Voice over Internet Protocol is a technology that enables voice calls and multimedia communication to be transmitted over IP networks—most commonly, the internet. Instead of relying on traditional telephone lines, VoIP converts voice into digital data signals and sends them through your internet connection, offering a more flexible and cost-effective alternative to POTS. 

VoIP powers everything from business phone systems to unified communications platforms, with SIP trunking being one of the most widely used methods for delivering VoIP services. Other implementations include private telephone networks and direct point-to-point internet connections.

Speak the Language of Modern Business Communication

Understanding these essential telephony terms and acronyms isn’t just about sounding tech-savvy. It’s about making smarter decisions for your business communications. Whether you’re setting up a new VoIP system, optimizing your office phone line, or choosing a reliable service provider, having a clear grasp of these concepts gives you the confidence to ask the right questions and invest in the right solutions.

At SIP.US, we make it easy to modernize your communications with straightforward pricing, flexible control, and fast setup so you can focus on running your business, not decoding telecom jargon. We’ve designed our platform to put power in your hands, and if you ever need help, our support team has your back. Get started with SIP.US today to simplify your phone system and speak the same language as your tech team.