Glossary

Explore the terms that power better collaboration. This glossary demystifies the terminology behind Cisco, Webex, Zoom, and other UC and call analytics platforms—helping teams understand what drives smarter communication and better customer experiences.

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ACD (Automatic Call Distributor)

An Automatic Call Distributor is a telephony system that routes incoming calls to specific agents or groups based on rules such as skills, availability, or workload. ACDs are core to contact centers and help ensure calls are handled efficiently and fairly. Analytics on ACD performance can reveal bottlenecks and opportunities to improve service levels.

Abandon rate

Abandon rate is the percentage of calls that are terminated by callers before they are answered by an agent or endpoint. It is calculated as abandoned calls divided by total offered calls over a defined period. A high abandon rate often signals long wait times, poor routing, or understaffing.

Abandoned call

An abandoned call is a call where the caller hangs up or disconnects before reaching an agent or being answered. High abandoned call rates can indicate long queue times, poor IVR design, or insufficient staffing. Monitoring abandoned calls helps improve customer experience and optimize contact center resources.

Agent adherence

Agent adherence measures how closely contact center agents follow their scheduled work patterns, such as login times, breaks, and call handling tasks. Good adherence helps ensure proper coverage and consistent service levels. Poor adherence can lead to longer wait times, higher abandon rates, and reduced productivity.

Agent group

An agent group is a collection of contact center agents grouped together for routing, reporting, or management purposes. Groups may be organized by skill, department, language, or location. Analytics by agent group help supervisors understand workload distribution and performance trends.

Agent occupancy

Agent occupancy is the percentage of time agents spend actively handling calls or related work compared to their total logged-in time. High occupancy can indicate strong productivity but may also signal burnout risk if sustained too long. Low occupancy may suggest overstaffing, idle time, or inefficient call distribution.

Answer-seizure ratio (ASR)

Answer-seizure ratio is a telephony metric that compares answered calls to total call attempts. A high ASR indicates that most calls are successfully connected, while a low ASR may point to network issues, trunk problems, or blocked calls. ASR is often used for quality monitoring in carrier and enterprise voice environments.

Auto attendant

An auto attendant is an automated call handling system that answers calls, plays greetings, and offers menu options such as Press 1 for Sales before routing callers to destinations. It reduces the need for a live receptionist and helps callers quickly reach the right department or person. Well-designed auto attendants can improve caller experience and reduce misrouted calls.

Average call duration (ACD metric)

Average call duration is the average length of time calls last, from answer to hang-up, over a defined period. This metric helps organizations understand how complex or time-consuming typical interactions are. Changes in average call duration can indicate new customer issues, process changes, or policy shifts.

Average handle time (AHT)

Average handle time is the average amount of time an agent spends on a customer interaction, including talk time, hold time, and after-call work. It is a key efficiency metric in contact centers and is often balanced against quality and customer satisfaction goals. Extremely high or low AHT can both signal process or training issues.

Average speed of answer (ASA)

Average speed of answer is the average time callers wait in queue before their calls are answered by an agent or endpoint. Lower ASA values typically indicate better responsiveness and higher customer satisfaction. Monitoring ASA helps contact centers adjust staffing, routing, and priorities to meet service level targets.

Bandwidth utilization

Bandwidth utilization is the percentage of available network bandwidth that is currently being used. In VoIP and UC environments, high bandwidth utilization can negatively impact call quality if voice traffic competes with data applications. Analytics on bandwidth usage support capacity planning and QoS tuning.

Busy-hour call attempts (BHCA)

Busy-hour call attempts is the number of call attempts made during the busiest hour in a given period. It is used to size and design telephony systems, trunks, and gateways to handle peak loads. Tracking BHCA helps ensure that infrastructure can support growth and seasonal spikes without congestion.

Busy-hour call completions (BHCC)

Busy-hour call completions is the number of successfully completed calls during the busiest hour. This metric indicates how well the network and telephony systems handle peak demand. Comparing BHCC and BHCA can highlight capacity, congestion, or quality issues at busy times.

CUCM AXL API

The AXL API exposes CUCM configuration data for provisioning, reporting, and automation workflows.

CUCM CDR

CUCM CDRs provide metadata about every call, including duration, origin, destination, and outcome.

CUCM CMR

CUCM CMRs provide media-quality statistics essential for diagnosing call-quality issues.

CUCM SIP trunk

A SIP trunk in CUCM transports call signaling and media to carriers or external systems.

CUCM cluster

A CUCM cluster groups multiple servers to provide scalable, redundant call control.

CUCM device pool

Device pools standardize endpoint configurations by assigning shared settings such as region and SRST reference.

CUCM dial plan

The dial plan defines how CUCM interprets and routes dialed digits across patterns, partitions, and route lists.

CUCM extension mobility

Extension mobility allows users to log into any Cisco phone and temporarily adopt their extension and settings.

CUCM partition

Partitions restrict or allow access to directory numbers based on calling search spaces.

CUCM route group

A CUCM route group organizes trunks and gateways for outbound routing with load balancing or failover.

Call capacity planning

Call capacity planning is the process of forecasting voice traffic and ensuring there are enough trunks, gateways, licenses, and agents to handle demand. It uses historical analytics, trends, and growth assumptions to prevent congestion and call failures. Effective capacity planning helps avoid dropped calls and protects caller experience.

Call center reporting

Call center reporting involves collecting, visualizing, and analyzing key performance metrics such as call volume, service levels, handle time, and agent performance. Good reporting enables supervisors to make data-driven decisions about staffing, training, and processes.

Call concurrency

Call concurrency is the number of simultaneous calls in progress at a given time across trunks, gateways, or platforms. Monitoring concurrency helps ensure sufficient capacity and reveals peak usage patterns. High concurrency periods are critical for capacity planning and cost optimization.

Call detail record (CDR)

A call detail record is a log entry generated by telephony systems that describes the details of a call, such as time, duration, participants, and call result. CDRs are the foundation for call analytics, billing, compliance, and troubleshooting.

Call management record (CMR)

A call management record provides technical call-quality metrics such as jitter, packet loss, latency, and MOS. CMRs complement CDRs by explaining media-layer issues affecting voice clarity. Analyzing both CDR and CMR data provides a complete view of call performance.

Call quality analytics

Call quality analytics is the practice of measuring and analyzing voice quality metrics such as MOS, jitter, latency, and packet loss across calls and locations. It identifies network issues, codec problems, or configuration errors that impact voice performance.

Call quality statistics

Call quality statistics summarize numerical measures of voice performance like MOS, jitter, latency, and packet loss. Tracking these values helps organizations maintain high audio quality and quickly detect degradation.

Call queue

A call queue is a holding area for incoming calls waiting to be answered by agents or endpoints. Queue analytics reveal wait times, abandonment rates, and staffing gaps.

Call routing strategy

A call routing strategy defines how calls are directed to destinations based on rules like time of day, caller location, skills, or priority. Good routing improves efficiency and customer experience.

Call spoofing

Call spoofing is the falsification of caller ID information, often used for fraud, spam, or intrusion. Monitoring unusual call behaviors helps detect spoofing and protect users.

Call type

Call type categorizes calls by origin and destination—internal, external, inbound, or outbound. Understanding call types helps organizations analyze performance, cost, and user behavior.

Call volume

Call volume is the total number of calls handled over a selected time period. It is a core workload metric used in staffing, forecasting, and capacity planning.

Cisco

Cisco is a global networking and collaboration provider offering CUCM, Webex Calling, and contact center products. Many UC deployments rely on Cisco platforms for enterprise telephony. UC analytics tools integrate with Cisco systems to deliver visibility into calling behavior and performance.

Cisco CUCM

Cisco Unified Communications Manager is Cisco’s primary call-control platform for enterprise voice. It handles call setup, routing, device management, and generates CDR/CMR data for analytics.

Cisco CallManager

Cisco CallManager is the original branding for Cisco Unified Communications Manager. It manages call control, routing, and device registration for Cisco IP telephony environments.

Cisco Emergency Responder (CER)

Cisco Emergency Responder improves emergency call routing and ensures accurate location information is provided to first responders. Analytics validate correct behavior and compliance.

Cisco SIP trunk

A Cisco SIP trunk connects CUCM to carriers, SBCs, or PSTN providers. Monitoring SIP trunks ensures call quality and availability.

Cisco UCCX

Cisco Unified Contact Center Express supports small-to-mid contact centers with call routing, queueing, scripting, and reporting.

Cisco Unity Connection

Cisco Unity Connection provides voicemail and unified messaging services with full integration to CUCM.

Cisco device utilization

Cisco device utilization measures how often Cisco IP phones, soft clients, and other endpoints are used. Low utilization indicates underused assets or poor adoption.

Cisco hunt pilot

A Cisco hunt pilot is the entry point for a hunt group. It routes calls to a configured list of line members based on distribution rules.

Cisco line group

A Cisco line group organizes directory numbers for call distribution through hunt lists. Analytics show usage patterns and help optimize call flow.

Cisco route list

A Cisco route list defines the order of route groups used for outbound call routing and failover within CUCM.

Cisco route pattern

A Cisco route pattern matches dialed numbers and determines routing behavior. Analytics help validate dial-plan effectiveness.

Cisco voice gateway

A Cisco voice gateway bridges IP telephony with the PSTN or legacy analog/digital circuits.

Cloud calling

Cloud calling provides PBX and telephony services through cloud infrastructures rather than on-premises hardware.

Compliance reporting

Compliance reporting documents that calling activity meets regulatory, legal, or policy requirements (HIPAA, PCI, etc.).

Contact center analytics

Contact center analytics measure performance across agents, queues, SLAs, and customer experience.

Control Hub (Webex Control Hub)

Webex Control Hub is the admin portal for Webex Calling and collaboration services, offering analytics for usage and voice quality.

DID (Direct Inward Dial)

DID numbers allow external callers to reach internal extensions directly without going through an operator or auto attendant.

DSCP (Differentiated Services Code Point)

DSCP marks voice packets with priority settings that enable QoS across networks.

Dial plan

A dial plan governs how numbers are interpreted, normalized, and routed within a UC solution.

Echo cancellation

Echo cancellation removes echo artifacts caused by acoustic feedback or latency to improve audio clarity.

Emergency calling

Emergency calling routes 911 or equivalent calls with accurate location information and proper priority.

Endpoint utilization

Endpoint utilization measures how often desk phones, softphones, and conference devices are used across the enterprise.

Enterprise telephony

Enterprise telephony refers to UC platforms, PBXs, cloud calling systems, and gateways supporting business voice communications.

Extension

An extension is the internal phone number assigned to a device or user for routing calls inside the organization.

First call resolution (FCR)

FCR measures whether customer issues are resolved on the first interaction, improving satisfaction and reducing call volume.

Gateway utilization

Gateway utilization measures traffic load on PSTN or SIP gateways and helps optimize capacity.

Global Call ID

A unique identifier used to correlate CDRs and CMRs across CUCM nodes.

HD voice

HD voice uses wideband codecs to deliver clearer, more natural-sounding audio than traditional telephony.

Hunt group

A hunt group distributes calls among multiple lines or agents using patterns like broadcast or sequential ringing.

Hybrid UC deployment

A hybrid deployment mixes on-premises systems like CUCM with cloud calling platforms like Webex or Teams.

IP phone

An IP phone uses VoIP technology to make and receive calls over IP networks.

IP telephony

IP telephony transmits voice over IP networks instead of legacy circuit-switched infrastructure.

Inbound call

An inbound call originates outside the organization and reaches internal users, queues, or auto attendants.

Jabber (Cisco Jabber)

Cisco Jabber provides messaging, presence, and softphone calling integrated with CUCM.

Jitter

Jitter is variation in packet arrival times and can degrade voice clarity if not controlled.

Latency

Latency is delay between sending and receiving audio packets, affecting real-time communication quality.

Mean Opinion Score (MOS)

MOS is a 1–5 score estimating perceived voice quality based on audio performance metrics.

Microsoft Teams

Microsoft Teams provides meetings, chat, and calling capabilities for enterprise collaboration.

Microsoft Teams Phone

Teams Phone provides enterprise calling via PSTN connectivity, auto attendants, and call queues.

Mobile softphone

A mobile softphone enables enterprise calling features on mobile devices via Wi-Fi or cellular data.

Network performance

Network performance describes how well data and voice travel across the infrastructure, affecting call quality.

Off-net call

An off-net call routes outside the internal voice network to the PSTN.

On-net call

An on-net call stays entirely inside the organization's voice network, typically offering better quality and no usage fees.

Operator Connect

Operator Connect integrates PSTN calling into Microsoft Teams via trusted telecom partners.

Outbound call

An outbound call is initiated by a user to external destinations.

PBX (Private Branch Exchange)

A PBX manages internal/external calls within organizations and may exist on-prem or in the cloud.

PSTN (Public Switched Telephone Network)

The PSTN is the traditional global phone network enabling external voice communication.

Packet loss

Packet loss occurs when voice packets are dropped, causing choppy or distorted audio.

Peak concurrency

Peak concurrency is the maximum number of simultaneous calls and informs trunk capacity planning.

Presence

Presence displays real-time user availability (busy, away, in meeting) to improve routing and communication.

QoS (Quality of Service)

QoS prioritizes voice packets on the network to ensure reliable, high-quality calls.

Queue depth

Queue depth counts callers waiting in a queue at a specific time.

Queue time

Queue time measures caller wait duration before an agent answers or the caller abandons.

RTP (Real-time Transport Protocol)

RTP transmits audio/video streams during VoIP calls and is key to understanding media performance.

Real-time reporting

Real-time reporting provides immediate visibility into call status, queue levels, and agent availability.

Remote workforce calling

Remote workforce calling uses UC applications to support distributed employees and requires monitoring of home network quality.

Ring time

Ring time measures the time from when a call begins ringing to when it is answered or abandoned.

SIP (Session Initiation Protocol)

SIP controls call setup, management, and teardown in VoIP communications.

SIP trunk

A SIP trunk connects UC systems to PSTN carriers for inbound/outbound calling.

Webex Calling

Webex Calling is Cisco’s cloud-based enterprise PBX offering calling, routing, and device management.

Zoom

Zoom is a cloud collaboration platform providing meetings, chat, and telephony through Zoom Phone. It supports hybrid and remote work environments.

Zoom Phone

Zoom Phone is Zoom’s enterprise cloud telephony platform with calling, voicemail, auto attendants, and call queues.

Zoom Phone analytics

Zoom Phone analytics provide insights into call quality, capacity, routing behavior, and adoption trends across Zoom Phone environments.