We’ve created a glossary of frequently used IP telephony terms below for your quick reference. Please let us know if you feel anything is missing and we can add it for you. Thank you!
“Automatic Call Distributor” often used by call centers. A phone system that manages and routes inbound calls to a specific group of people at first availability based on caller’s preference/selections (such as calling Sales or CSR instead of a specific person).
The “Average Length of Call”
An “Automatic Number Announcement Circuit” – used by technicians to determine the assigned phone number to a particular line.
A telephone that transmits voice or video data as electronic pulses over POTS (aka “Plain Old Telephone Service” ) enabling richer quality but less clarity and functions. Supports standard phone, fax, and modems. Note: Do NOT connect your analog phones directly with digital phone system – get a digital-to-analog adapter first or you risk frying the phone systems.
A free and open source framework for building communication applications by digium, it’s basically a software implementation of a PBX. It turns an ordinary computer into a communications server and powers IP PBX systems, VoIP gateways, conference servers and more.
“Analog Telephone Adapters” – a hardware interface device between a PSTN analog phone system and a digital or VoIP network service. With an ATA, you can use your old PSTN phone system with VoIP service.
Basic Rate Interface
Formally known as “OpenPBX.org” – CallWeaver is a community-driven vendor-independent cross-platform open source PBX software project originally derived from Linux-based Asterisk. Supports major PSTN technologies.
XML based computer telephony language that allows a company to describe a phone-to-website application based on how the call should be handled and/or interact with caller based on their responses.
“Call Control XML” – XML based computer telephony language that controls the call session (placement, answer, transfer, conference etc).
Similar to a PBX but provides switching from service provider site instead of customer premise and can support multiple locations.
Technology enables the compression and decompression of audio/data.
“Call Processing Language” used to describe and control IP telephony services.
A telephone that converts voice/video data into binary format (0′s and 1′s) and transmit over digital phone systems. You’ll get higher clarity and able to include more features/functions, but a less rich quality compared with analog.
An open-source telephony platform designed to facilitate the creation of voice and chat driven products.
“Foreign eXchange Office” interface – the phone port that receives an analog line.
Adapters to connect VoIP system to analog phone line.
“Foreign eXchange Subscriber” interface – the phone port that delivers the analog line to the subscriber.
Adapters to connect analog phone/fax machine to VoIP.
G.711 Codec (a-law & μ-Law)
ITU-T standard for audio companding (compression and expansion) and provide best voice quality for VoIP as it’s the same codec used by PSTN and ISDN. U-Law is used in North America and Japan while A-Law is used in the rest of the world.
Speed codec for wideband (HD) audio that delivers voice calls via VoIP with superior quality than a regular landline or cell phone call.
Audio data compression codec used with VoIP applications with conservative bandwidth while offering toll-quality speech. Also known as CS-ACELP or Conjugate Structure Algebraic Code Excited Linear Prediction. G.729 allows moderate transmission delays (10-ms frames) to provide quality of service, interoperability and increased bandwidth to its users.
Commonly used with VoIP telephony and video conferencing, H.323 is an ITU standard protocol to provide specifications regarding audio-visual communications over IP network.
IP Phones (or VoIP Phone)
A digital telephone specifically designed to utilize VoIP network to transmit voice data communication via web. Instead of a conventional phone jack, IP Phones have Ethernet ports to connect with the internet (typically via a VoIP Server, VoIP Gateway, or another VoIP Phone).
Software application that allows the computer to function as a VoIP phone with use of headset connected directly with the computer instead of physical phone equipment.
“Integrated Service Digital Network” – a type of circuit switched phone network system that allows digital voice/data transmission via conventional copper phone wires Richer voice quality and better speed than analog systems.
An Internet Telephony Service Provider.
Interactive Voice Response – technology that allows automated response and interaction with callers through use of voice and keypad inputs.
Unwanted and sudden disturbance to the signal which can cause noticeable sound distortion.
A type of phone system traditionally used by company with fewer users. Key system telephones will have multiple buttons (keys) that lights up to indicates which line or extension is already in use, so the user can select the idle one (by pushing the unlit button) to call out.
An expression for the time frame it takes for a packet of data to be transmitted.
A computer networking device that create a network and used to connect various devices together such as computers, printers and servers within a building or campus.
“Private Branch Exchange” – A private switchboard phone-system enabling a company to use multiple phone extensions for one phone line. Typically used by larger companies with high number of employees.
PCI interface cards to connect VoIP PBX’s with analog lines, T1/E1 lines, and analog equipments.
“Power over Ethernet” – The technology allowing for electrical power (along with data) to be passed thru the ethernet cable instead of additional a/c supply. *Specific equipment is needed to work with POE technology.
Primary Rate Interface
The “Public Switched Telephone Network” – also known as “POTS Plain Old Telephone Service”, is the traditional circuit-switched telephony system.
QOS stands for “quality of service” and refers to the various aspects of a telephony connection including service response time, loss, signal-to-noise ratio, cross-talk, echo, interrupt, frequency response, loudness levels, etc.
A device that connects multiple networks and forward data packets between those networks.
The “Session Initiation Protocol” is a text-based signaling protocol commonly used for VoIP and other multimedia communication sessions, offering features such as call transfer, conference, or hold.
A service offered by an ITSP (Internet Telephony Service Provider) that allows a business to use VoIP communication to outside the company network, once a PBX has been installed, by utilizing the same internet connection. Provide great cost savings.
Standard XML format for specifying interactive voice dialogues between a human and a computer, such as used with automated telephone services.
“Voice over Internet protocol” or “Voice over IP”, a communications protocol that allows for voice data (telephonic communication) to be transmitted via the Internet. Because the call is made over the internet, VoIP will not incur long-distance phone bills.
A network device that converts telephony data between PSTN and IP.
VoIP GSM Gateways
A device that allows direct routing between IP, digital, analog and GSM networks.
“Virtual Private Network” – a secure network allowing users to remote access a central organization network after authentication.