VoIP glossary: 100 terms explained.

A system that automatically routes incoming calls to the most appropriate agent or department based on rules like skill level, availability, or time of day. The backbone of most contact center platforms.

The technical mechanism that identifies the phone number of an incoming caller. Think of it as the infrastructure behind Caller ID. Often used in contact centers to pull up customer records automatically before the agent answers.

In VoIP, APIs let your phone system connect to other tools your team already uses, like CRMs, helpdesks, and ticketing systems. Platforms like RingCentral and Dialpad expose REST APIs that developers use to trigger calls, log activity, or build custom workflows.

A device that converts traditional analog phone signals into digital VoIP packets so you can use a legacy desk phone on a modern cloud phone system. Useful during migrations when you're not ready to replace physical hardware.

An automated system that answers incoming calls and presents a menu of options ("Press 1 for Sales…"). A standard auto-attendant uses DTMF tones (button presses); an AI receptionist understands natural language. Most VoIP platforms include at least basic auto-attendant at every tier.

The capacity of your internet connection to carry data. VoIP calls require roughly 100 kbps per concurrent call at standard quality. Insufficient bandwidth or unmanaged bandwidth causes choppy audio, dropped calls, and jitter.

A visual indicator on a desk phone or softphone that shows whether a colleague is available, on a call, or do-not-disturb. Essential for receptionists managing calls for a team of people.

A configuration where you connect your own SIP trunk or PSTN carrier to a UCaaS platform rather than using the provider's built-in calling plans. Often used to reduce per-minute costs or keep an existing carrier relationship. Microsoft's Direct Routing is the most common BYOC scenario.

A policy that allows employees to use personal devices (smartphones, laptops) to make and receive business calls through the company phone system via softphone apps. Most modern VoIP platforms support BYOD natively.

A SIP component that terminates an incoming call and initiates a new outgoing call, acting as both the endpoint and the originator simultaneously. Used in SBCs, hosted PBX systems, and call recording platforms to intercept, inspect, and relay call signaling. Most enterprise VoIP infrastructure includes B2BUA logic.

The configured path a call takes through your phone system, starting from the moment it's answered by the auto-attendant, through any menus or queues, to the final destination. Designing a good call flow is one of the most impactful things you can do for caller experience.

Redirecting incoming calls to a different number such as a mobile phone, another extension, or voicemail. Can be set unconditionally, on no-answer, on busy, or when out of coverage. A basic feature included on every VoIP platform.

Placing a call on hold in a shared "parking lot" so another employee can pick it up from a different phone or extension. Common in retail, hospitality, and healthcare environments where staff move around.

A virtual waiting line that holds incoming callers when all agents or employees are busy. Callers hear hold music or a message and are connected to the next available person. Queue management, including overflow rules and callback options, is a key differentiator between basic VoIP and full UCaaS platforms.

The automatic or on-demand recording of phone calls for quality assurance, compliance, training, or documentation purposes. Cloud-based recording is included on mid-tier and above plans for most UCaaS providers. Storage duration and access controls vary significantly.

The ability to see who is calling and decide how to handle it before answering. Options typically include sending to voicemail, answering, or forwarding. Some systems announce the caller's name and reason for calling to the receiving employee before connecting.

Cloud-based contact center software delivered on a subscription basis. Includes advanced ACD, agent dashboards, workforce management, quality monitoring, and omnichannel (voice, chat, email, social) capabilities. Distinct from UCaaS, though some providers like 8x8 and RingCentral offer both on one platform.

A data file that captures metadata about every call, including caller ID, called number, duration, time, and result. CDRs are used for billing reconciliation, usage analysis, and compliance reporting.

The phone number displayed to the person receiving your call. In VoIP, CLI can be set to show your main business number regardless of which extension or device is making the call, so your outbound identity stays consistent.

A Private Branch Exchange hosted in the cloud rather than on-premises hardware. The provider manages the infrastructure; you access it via the internet. Effectively synonymous with hosted VoIP for most business buyers. See also: Hosted PBX.

Software that compresses and decompresses audio for transmission over the internet. Different codecs trade off audio quality vs. bandwidth consumption. G.711 delivers high quality but uses more bandwidth; G.729 compresses more aggressively but can reduce quality on poor connections.

A cloud platform that lets developers embed real-time voice, SMS, video, and messaging capabilities directly into their own applications via APIs, without building telecom infrastructure. Twilio is the most well-known example. Different from UCaaS, which is a packaged end-user product.

The integration between a phone system and a computer application, most commonly a CRM like Salesforce. CTI enables click-to-dial, automatic screen pops (pulling up the caller's record when a call comes in), and automatic call logging after the call ends.

A feature that lets website visitors or CRM users initiate a phone call by clicking a button or phone number. No manual dialing required. Powered by WebRTC or a CTI integration. Widely used in sales workflows to increase outbound call volume and in customer support to reduce friction for inbound inquiries.

A dedicated dial-in number that allows multiple participants to join a single audio call simultaneously. Each participant dials the bridge number and enters a PIN or meeting code. Standard on most UCaaS platforms; some providers include unlimited conference bridges, others charge per use or require a higher tier.

A phone number that routes directly to a specific person, department, or extension without going through an operator or main number. Each employee can have their own DID while sharing the underlying phone system infrastructure.

A Microsoft Teams configuration that connects Teams Phone to an external SIP trunk via a certified Session Border Controller (SBC), allowing organizations to use their own PSTN carrier instead of Microsoft Calling Plans. Reduces per-user cost significantly but requires ongoing SBC management.

Identifies which number the caller dialed. Useful when one phone system handles multiple business numbers across different brands, departments, or campaigns. Allows the system to route the call or greet the caller differently depending on which number was dialed.

A status setting that silences incoming calls and sends them directly to voicemail (or another destination) without ringing the user's device. Standard on every VoIP platform. Can often be set per device or system-wide.

The technical name for the tones generated when you press a key on a phone keypad. Used to navigate phone menus (auto-attendants, IVRs). Each key produces a unique combination of two audio frequencies that the phone system recognizes.

The set of rules that govern how a phone system interprets and routes dialed digits, including which numbers go to which extensions, which require an outside line prefix, and how calls are routed based on time, location, or caller. A well-designed dial plan is invisible to users. A poorly designed one is a constant source of frustration.

The international standard format for phone numbers. It starts with a "+" followed by the country code and subscriber number (e.g., +18445062299). VoIP systems use E.164 formatting internally even when displaying numbers in local formats to users.

The US emergency calling standard that automatically provides a caller's location to emergency services. VoIP E911 compliance requires organizations to register location information for each user, especially important for remote workers. Misconfigured E911 is one of the most common and serious VoIP oversights.

An internal number used to reach a specific person or department within a phone system, typically 3 to 5 digits. External callers dial your main number; internal callers can dial extensions directly. Most VoIP platforms support unlimited extensions.

Automatic rerouting of calls to a backup system, carrier, or device when the primary path fails. Enterprise-grade VoIP platforms use geographic redundancy and multiple carrier relationships to ensure calls connect even when infrastructure goes down.

A feature that rings multiple devices or numbers sequentially or simultaneously when a call comes in. For example: office phone first, then mobile, then home office. Ensures callers reach a live person even when someone is away from their desk.

A port on a VoIP gateway or ATA that connects to the traditional PSTN (a telephone wall jack). FXO ports receive analog PSTN lines into a VoIP system. They are typically used during migrations from legacy to cloud systems.

A port on a VoIP gateway or ATA that connects to traditional analog devices such as desk phones, fax machines, and door phones. FXS ports provide the dial tone and power that analog devices expect, allowing them to work on a VoIP network.

Technology that transmits fax documents over an IP network rather than traditional telephone lines. The most common implementation uses the T.38 protocol to handle the real-time nature of fax signaling over VoIP. Many businesses migrating to cloud VoIP are surprised to discover their fax lines require special handling. Always confirm FoIP or analog ATA support before cutting over.

The most common VoIP audio codec. It delivers high-quality, uncompressed audio that closely matches PSTN call quality. Requires ~64 kbps per call. Best choice when bandwidth is ample; the default on most enterprise-grade systems.

A compressed VoIP audio codec that uses only ~8 kbps per call, making it 8x more efficient than G.711. Useful on constrained or high-congestion connections, but introduces slight audio degradation. Often used as a fallback codec.

Infrastructure design where a VoIP platform runs across multiple data centers in different geographic regions. If one data center goes offline, calls automatically route through another. Required for 99.999% uptime SLAs.

Audio quality that captures a wider frequency range than traditional telephone calls, making voices sound clearer, more natural, and less fatiguing. Requires both ends of the call to support it. Standard on all modern VoIP platforms; enabled by codecs like G.722 and Opus.

A business phone system where the PBX hardware and software are hosted and managed by a third-party provider in the cloud, as opposed to on-premises equipment in your office. Most cloud VoIP services are hosted PBX systems. Also called Cloud PBX or Virtual PBX.

A feature that lets any employee log into any shared desk phone and have it behave as their personal extension, complete with their number, voicemail, and settings. Common in hybrid workplaces where employees don't have assigned desks.

A set of extensions or users that incoming calls cycle through until someone answers. Can ring all simultaneously (blast), in order (linear), or using round-robin distribution. Used to ensure calls to a department are always answered by someone available. Also called a Ring Group.

A physical desk phone that connects to your network via Ethernet (or Wi-Fi) and makes calls over VoIP rather than a traditional phone line. Also called a VoIP phone or SIP phone. Hardware from Poly, Cisco, Yealink, and Grandstream is common in business deployments.

A Private Branch Exchange that uses IP (internet protocol) to route calls, either via on-premises hardware or a cloud-based system. Most modern business phone systems are IP PBXs. See also: Cloud PBX, Hosted PBX.

A system that interacts with callers through a combination of voice prompts and keypad input (DTMF). Traditional IVRs use menus; modern AI-powered IVRs understand natural speech. Used for self-service (account balance, appointment confirmation), call routing, and data collection before a live agent answers.

Inconsistency in the arrival timing of voice data packets. While some latency is tolerable, jitter is the variation in that latency and is what causes choppy or robotic-sounding audio. Fixed with a jitter buffer (software that smooths out irregular packet delivery). High jitter is usually a sign of network congestion or QoS misconfiguration.

A component in VoIP systems that temporarily stores arriving voice packets and releases them at a steady rate to compensate for jitter. A dynamic jitter buffer adjusts its size automatically based on network conditions. Reduces choppy audio on imperfect connections.

The delay between when audio is spoken and when it's heard on the other end of a VoIP call. Under 150ms is imperceptible; 150–400ms is noticeable but acceptable; over 400ms causes conversation difficulty. Caused by distance to servers, network congestion, or processing overhead.

The regulatory requirement that allows phone numbers to be transferred from one carrier to another so you keep the same number when you switch VoIP providers. The porting process typically takes 2–4 weeks. See also: Number Porting.