88 terms used in SIP trunking, international voice routing, call-centre operations and softswitch deployment, each defined in plain English. Telecom abbreviations are reused across contexts — ACD means two different things depending on who is speaking — so where a term is ambiguous both meanings are listed.
Use the index to jump to a letter. For questions rather than definitions, see the business VoIP FAQ and the quick answers page.
A
ACD (Automatic Call Distribution)
A call-routing function that places inbound calls into queues and distributes them to agents according to rules such as availability, skill, priority, or longest idle time.
ACD (Average Call Duration)
In carrier reporting, ACD can also mean average call duration: the total connected minutes divided by the number of answered calls. The meaning should be clear from context.
After-Call Work
The short period after a conversation when an agent adds notes, selects a disposition, schedules a follow-up, or updates the CRM before becoming available again.
ANI
Automatic Number Identification is network-delivered information that identifies the originating number. It is related to caller ID but may be handled differently by carriers and billing platforms.
Answer Supervision
The signaling event that tells the billing system a call has actually been answered. Incorrect supervision can create billing disputes or false-answer problems.
ASR
Answer Seizure Ratio is the percentage of attempted calls that receive an answer signal. It is useful for route comparison, but it is also influenced by customer behavior, busy numbers, and unreachable destinations.
ATA
An Analog Telephone Adapter connects a traditional analog handset or fax device to an IP network by converting analog audio into VoIP traffic.
B
B2BUA
A Back-to-Back User Agent terminates one SIP call leg and creates another. PBX and softswitch platforms often use this model to control signaling, routing, media, and billing.
Bandwidth
The network capacity available to carry data. A VoIP call uses only a fraction of a typical broadband connection, but many simultaneous calls require planned upload and download capacity.
Billing Increment
The unit used to round call duration for charging, such as one-second, six-second, or sixty-second billing. A 60/60 route bills the first minute and each later minute in full.
Blind Transfer
A transfer that sends the caller to another user or destination immediately, without the transferring agent first speaking to the recipient.
Busy Hour
The period when a voice network handles its highest normal traffic. Capacity planning should be based on busy-hour demand rather than daily averages.
C
Call Barge
A supervisor feature that allows an authorized manager to join an active agent call, usually for escalation, coaching, or quality control.
Call Detail Record (CDR)
A structured record containing details such as source, destination, start time, answer time, duration, route, charge, and termination status.
Call Forwarding
A rule that redirects incoming calls to another extension, mobile number, queue, voicemail box, or external destination.
Call Queue
A waiting line for inbound callers when all matching agents are busy. Queue strategy, announcements, music, and maximum wait time can be configured.
Call Recording
The capture of call audio for quality review, training, verification, or record-keeping. Recording must follow the consent and privacy rules that apply to the parties involved.
Call Whisper
A private audio prompt heard by the agent, but not the customer, before or during a call. It can identify the campaign, lead source, or support queue.
Caller ID / CLI
Calling Line Identification is the number presented to the receiving network or called party. Display is subject to number ownership, route support, verification, and local regulation.
Channel
A unit of simultaneous call capacity. Ten channels generally allow up to ten active call legs at the same time, subject to platform design and conferencing behavior.
Click-to-Call
A feature that starts a phone call when a user clicks a number inside a CRM, browser, helpdesk, or other business application.
CNAM
Caller Name information associated with a telephone number in supported networks. Name display is not universal and may depend on destination databases and carriers.
Codec
A method used to encode and decode voice. Codec choice affects audio quality, bandwidth, CPU load, compatibility, and whether transcoding is required.
Concurrent Calls
The number of calls active at the same moment. This is a key sizing metric for trunks, servers, licenses, gateways, and internet links.
CPS
Calls Per Second measures how quickly new call attempts are created. High-volume dialers and broadcast systems must stay within the CPS limits of the switch and carrier.
D
DID
A Direct Inward Dialing number is a public telephone number that can deliver inbound calls to a VoIP account, extension, IVR, queue, or PBX.
Dial Plan
A set of rules that interprets dialed digits and decides how a call should be formatted, permitted, transformed, and routed.
DND
Do Not Disturb prevents calls from ringing a selected endpoint. Depending on the system, calls may be sent to voicemail, another agent, or a queue.
DTMF
Dual-Tone Multi-Frequency signals are the keypad tones used for IVR selections, PIN entry, conferencing, and automated services. VoIP systems may send them in-band, as SIP INFO, or as RTP events.
E
E.164 Format
A standard international number format built around the country code and national subscriber number. Systems commonly expect digits such as 14155550100, sometimes with a leading plus sign.
Endpoint
Any device or application that originates or receives a VoIP call, including a softphone, IP phone, ATA, gateway, browser client, or PBX.
Extension
A short internal number assigned to a user, department, room, device, or application inside a PBX.
F
Failover Route
A backup route used when the preferred carrier is unavailable, rejects a call, reaches a capacity limit, or falls below a defined quality rule.
FAS
False Answer Supervision occurs when a network signals that a call was answered before the called party actually answers. This can cause premature billing and misleading statistics.
FXO
Foreign Exchange Office is the interface that receives an analog telephone line from a carrier or PBX. An FXO port behaves like a telephone connected to that line.
FXS
Foreign Exchange Station is the interface that supplies dial tone and ringing voltage to an analog telephone or fax machine.
G
Gateway
A device or service that connects different voice networks or protocols, such as SIP to PSTN, analog to IP, or one codec to another.
H
Hosted PBX
A business phone system operated in a provider or cloud environment. Customers use apps or IP phones while the provider maintains the core switching platform.
I
ICE
Interactive Connectivity Establishment is a framework that tests available network paths and uses technologies such as STUN and TURN to connect media through NAT and firewalls.
IP Authentication
A trunk security method that accepts traffic only from approved public IP addresses. It avoids per-device registration but requires stable source IPs.
IP Phone
A desk or cordless phone designed to connect directly to an IP network and register with a SIP service or PBX.
IP PBX
A software- or appliance-based private branch exchange that manages extensions and calls using IP networks rather than only traditional telephone wiring.
ITSP
An Internet Telephony Service Provider supplies services such as SIP trunks, virtual numbers, origination, termination, hosted PBX, or wholesale voice.
IVR
Interactive Voice Response plays prompts and collects keypad or speech input so callers can reach departments, obtain information, or complete a self-service task.
J
Jitter
Variation in the arrival time of voice packets. Excessive jitter can make speech sound broken, robotic, or uneven.
Jitter Buffer
A temporary buffer that smooths irregular packet arrival. Too little buffering may cause choppy audio; too much can add noticeable delay.
L
Latency
The time taken for voice data to travel from one side of a conversation to the other. High latency makes natural conversation difficult because speakers begin talking over each other.
Least-Cost Routing (LCR)
A routing method that selects an eligible carrier using rate, destination, quality, capacity, or priority rules. The cheapest route is not always the best route.
Local Presence Number
A number from the customer’s target market that can ring a remote team. It can make a business easier to reach without requiring a physical office in that country.
M
Media Gateway
A network component that converts voice media between IP networks and traditional telephony interfaces or protocols.
MOS
Mean Opinion Score is a quality estimate commonly expressed on a one-to-five scale. It summarizes how understandable and natural a call is expected to sound.
N
NAT
Network Address Translation lets many private devices share a public internet address. It can complicate SIP signaling and RTP media unless the system handles address translation correctly.
Number Porting
The process of moving an existing telephone number from one provider to another while keeping the same public number.
O
Origination
The delivery of inbound calls from the public telephone network to a VoIP platform, SIP trunk, PBX, or application.
P
Packet Loss
Voice packets that do not reach the destination. Small losses may be concealed by the codec, while sustained loss causes missing words and broken audio.
PDD
Post-Dial Delay is the time between sending the final dialed digit and hearing ringback, an announcement, or an answer. Long PDD can make a route feel slow.
PoE
Power over Ethernet supplies electrical power and network connectivity to an IP phone through one Ethernet cable when compatible switches or injectors are used.
Power Dialer
A dialer that automatically calls contacts from a list, usually one after another, to reduce manual dialing time.
Predictive Dialer
A dialer that estimates agent availability and places multiple calls ahead of time. It requires careful pacing, abandonment controls, and legal compliance.
Preview Dialer
A mode that displays the contact record before the agent chooses to start the call. It is useful when preparation and personalization matter.
Progressive Dialer
A mode that automatically dials one contact when an agent becomes available. It offers more control than predictive dialing while still reducing idle time.
Proxy Server
A SIP server that receives signaling requests and forwards them toward the correct user, carrier, or destination without necessarily terminating both call legs.
Q
QoS
Quality of Service gives real-time voice traffic priority over less time-sensitive network traffic. Correct router and switch configuration can reduce delay and packet loss during congestion.
R
Rate Deck
A destination-by-destination price list, usually organized by prefix, country, network, billing increment, and effective date.
Registrar
A SIP server that records where a user or endpoint is currently reachable after it successfully registers.
Ring Group
A set of extensions that ring together or in sequence when a call reaches a shared sales, support, or departmental number.
RTP
Real-time Transport Protocol carries the audio or video packets for many VoIP calls. SIP usually controls the session, while RTP carries the media.
RTCP
Real-time Transport Control Protocol reports media statistics such as packet loss, jitter, and timing to help endpoints monitor call quality.
S
SBC
A Session Border Controller protects and controls SIP traffic at the edge of a voice network. It can enforce policy, normalize signaling, hide topology, manage NAT, and relay media.
SDP
Session Description Protocol describes media details such as IP address, port, codec, and direction during call setup.
SIP
Session Initiation Protocol is widely used to register users and establish, modify, and end voice or video sessions over IP networks.
SIP ALG
A router function that attempts to rewrite SIP traffic. Some implementations help, but many cause registration, one-way-audio, or header problems and may need to be disabled.
SIP Registration
A username-and-password method through which an endpoint tells the provider its current network address and proves it is allowed to use the account.
SIP Trunk
A logical IP connection between a PBX and a voice provider that carries one or many simultaneous calls.
Softphone
A telephone application for a desktop, browser, tablet, or smartphone. It uses a microphone, speaker, or headset instead of a dedicated handset.
Softswitch
A software-based switching platform that controls call routing, signaling, billing, users, and carrier connections. Some platforms also relay or transcode media.
SRTP
Secure Real-time Transport Protocol encrypts voice media between compatible endpoints or network elements.
STUN
Session Traversal Utilities for NAT helps an endpoint discover the public address and port seen outside its local network.
T
Termination
The delivery of an outbound VoIP call to a mobile, landline, toll-free, or other destination network.
TLS
Transport Layer Security can protect SIP signaling from interception or modification when both sides support and correctly configure it.
Transcoding
The conversion of live media from one codec to another. It improves compatibility but adds processing load and may slightly reduce quality.
TURN
Traversal Using Relays around NAT sends media through a relay when endpoints cannot establish a direct path.
U
Unified Communications
A combined environment for services such as business calling, messaging, meetings, presence, and collaboration.
V
Virtual Number
A telephone number that is not tied to one physical line. Calls can be routed to apps, PBX extensions, queues, or remote destinations.
Voicemail Transcription
A feature that converts a voicemail recording into text so users can scan the message without listening to the entire audio file.
VOS Server
A carrier or reseller voice-switch environment used to manage SIP customers, rates, routing, billing, balances, gateways, and call records. Exact functions depend on the installed platform and license.
W
Warm Transfer
A transfer in which the first agent speaks to the receiving person before connecting the customer, allowing context to be passed privately.
WebRTC
A browser-based real-time communication technology that can support voice and video without a traditional installed softphone.