Automatic Call Distributor & Automated Call Distribution (ACD)
An Automatic Call Distributor is a telephony device that answers incoming calls and distributes them to a group of agents or other termination points. ACD systems often use voice menus and work in conjunction with IVR and CTI systems to route the call to the most appropriate destination. Calls may be distributed based on various criteria including customer menu selections, calling party phone number, and time of day.
ATA – Analog Telephone Adapter
An Analog Telephone Adapter is a hardware device that converts analog audio signals into IP packets for transfer over the Internet. ATAs can be used to enable VoIP calling from a traditional telephone handset or fax machine.
Bandwidth refers to a data transfer rate – i.e. the volume of data that can be transferred over a network connection in a given period of time. It is normally measured in bits per second, kilobits per second, megabits per second, or gigabits per second.
Broadband refers to high-speed always-on Internet connections that can transport multiple signals and traffic types. The transmission medium may be optical fiber, coaxial cable, twisted pair or wireless.
Call Park is a telephony feature that lets users place a call on hold and ‘park’ it in the cloud, thus freeing up the phone and allowing the call to be picked up on a different extension by another employee. Call Park is a useful feature for minimizing caller hold times and improving customer service.
Caller ID, also known as Calling Line Identification or Call Display, is a telephony feature that identifies the originating number of an incoming call and displays it on the telephone of the called party.
Codec is short for compression/decompression. It describes the process of encoding voice and video traffic into a digital format for transmission across a network. Codecs use software-based algorithms to convert voice or video information into a byte sequence for transmission to a destination endpoint, where the sequence is converted back to voice or video.
A conference bridge allows a group of participants to attend the same phone call. Meetings can be scheduled in advance, and each participant uses their own phone to dial into a virtual meeting room at the appropriate time. Conference calling provides a convenient way for employees to collaborate with clients, business partners, and each other.
CTI – Computer Telephony Integration
Computer Telephony Integration refers to technology that supports integration or coordination between telephones and computers. Typical CTI functions include screen pops, CLI display, automatic dialing, call routing and call reporting.
DID – Direct Inward Dialing
Direct Inward Dialing is a service that enables organizations to allocate a unique telephone number to each individual user on its PBX system.
DSL – Digital Subscriber Line
Digital Subscriber Line is a technology that allows a broadband Internet connection to be established over a traditional copper telephone line, while also allowing the line to simultaneously carry normal phone calls. A DSL modem is used to convert digital data signals into analog signals suitable for the phone line.
DTMF – Dual-Tone Multi-Frequency
Dual-Tone Multi-Frequency refers to the signal generated when pressing keys on a telephone. The signals consist of two simultaneous tones that use a specific frequency to ensure they aren’t duplicated by the human voice. These tones are standardized and linked to a unique number (0-9 and #), allowing key presses to be used to navigate interactive telephone menus.
E911 is short for ‘Enhanced 911’ and is used to provide emergency services to users who dial 911 from wireless and IP phones. The E911 service enables the caller’s location to be relayed to the call receiver, so that police, fire or medical services can be dispatched to the appropriate destination.
Echo occurs when a speaker hears his or her own voice reflected back with a slight delay. Acoustic echo usually occurs when sound from a speakerphone is picked up by the mouthpiece and retransmitted back down the line. Electromagnetic interference or faulty wiring may also cause echo.
Echo cancellation and echo suppression are methods to improve audio quality by preventing echo or removing it when it already exists. Echo cancellers analyze an outgoing voice stream while also monitoring the return voice stream. If a copy of the outgoing stream is detected in the return stream then it is removed by subtracting it from the signal.
FoIP – Fax over Internet Protocol
Fax over IP refers to the process of sending and receiving faxes over an Internet connection. Similar to VoIP, the fax information is sent as IP packets instead of analog signals over traditional phone lines.
G.711, also known as Pulse Code Modulation (PCM), is a narrowband codec that delivers uncompressed toll-quality audio with very accurate speech reproduction. Each call consumes 64 kbps, so if call volumes are high then G.711 can require a large amount of bandwidth. This is one of the oldest codecs, having been introduced by the ITU in 1972.
G.729 is a codec that offers powerful compression and excellent bandwidth utilization, only consuming 8 kbps per call. It is a licensed codec, meaning users indirectly pay to use the codec when they buy hardware that supports G.729.
An IP Phone converts analog voice signals into digital IP packets, allowing phone calls to be placed over a network connection. IP Phones include signaling protocols like SIP or H.323 to ensure the call is routed to the correct location. They can be hardware-based devices or software-based softphones. Traditional PSTN phones can also be used with an Analog Telephone Adapter to function as an IP Phone.
IP telephony refers to technologies that transmit voice, video, fax and other forms of communication over an IP network instead of via the traditional circuit-switched public telephone network.
ISP – Internet Service Provider
An Internet Service Provider is a company that offers services for connecting to and using the Internet. Services provided by ISPs include Internet access, Internet transit, domain name registration and web hosting.
IVR – Interactive Voice Response
An Interactive Voice Response system is an application that presents an audio menu to the caller and allows the caller to make selections via key presses or spoken responses. Calls can then be routed to the appropriate destination based on the caller’s input.
Jitter is a measure of the variability in the arrival time of voice packets across a network. Jitter is common on switched networks because each packet may follow a different path to the destination. Jitter buffers collect packets as they arrive and relay them to the receiver at a constant pace and in the correct order to ensure conversations are understandable.
A Key System is a basic multiline business telephone system, typically used by organizations with fewer than 50 employees. Key System handsets have multiple buttons representing individual lines. Users pick up the receiver and press a button to access a line, and the button lights up to show the line is in use. Key Systems offer features such as hold buttons, speakerphone, intercom and paging.
Latency refers to the time taken for packets to travel across a network. On VoIP calls, latency is the delay experienced between the time one party speaks and the other party hears their words. One reason for latency is propagation delay – i.e. the time taken for digital voice packets, transmitted in the form of light, to travel along fiber optic cables from source to destination. Latency may also be caused by the network devices that handle and forward packets.
LEC – Local Exchange Carrier
Local Exchange Carrier refers to a local telephone company, as opposed to a long distance interexchange carrier (IXC). Most urban areas of the USA are served by LECs known as Regional Bell Operating Companies or Baby Bells.
LNP – Local Number Portability
Local Number Portability refers to the ability to reassign a customer’s telephone number to an alternative carrier, location, or type of service. For mobile phones, the equivalent service is known as Full Mobile Number Portability (FMNP}. In most cases there are some transferability limitations in terms of geography and technology.
A modem (short for modulator-demodulator) is a hardware device that encodes (modulates) digital information into electrical signals for transmission across standard telephone lines, and demodulates incoming signals as they arrive from the far end. Modems are classified by their baud rate, or their maximum data transfer rate in bits per second (bps).
Packet Loss occurs when data packets fail to reach their destination, due to being delayed and dropped as a result of network congestion, outages, or misconfiguration. In a VoIP environment, packet loss leads to choppy or unintelligible audio. Packet loss is measured as a percentage of packets lost versus packets transmitted.
Private Branch Exchange & Private Automatic Branch Exchange (PBX, PABX)
A Private Branch Exchange is a business telephone system that switches calls internally between users and also allows users to share a pool of external lines for local, long distance and international calls. A PBX is more scalable than a Key System, and offers more advanced features. PBX systems may be on-premise or hosted.
PoE – Power over Ethernet
Power over Ethernet is a technology that enables electrical current to be transmitted over standard Ethernet network cables rather than using traditional power cords. PoE is a convenient way to supply power to devices like VoIP phones, security cameras and RFID readers, without having to install dedicated electrical wiring.
POP – Point Of Presence
A POP is a physical demarcation point, termination point, or interface between communicating entities such as Internet service providers, networks, and local or long distance telecoms carriers.
POTS – Plain Old Telephone System
The Plain Old Telephone Service is the traditional phone service used in most homes, using analog signals transmitted over copper loops. This is in contrast to modern high-bandwidth digital communications services like ISDN and FDDI.
PSTN – Public Switched Telephone Network
The Public Switched Telephone Network incorporates a wide range of global interconnected voice networks, including traditional telephone lines (POTS), fiber optic cables, cellular networks, microwave transmission links, satellites and subsea cables. The PSTN was originally a network of fixed analog phone lines, but is now predominantly digital.
QoS – Quality of Service
QoS (Quality of Service) describes a concept whereby data bandwidth and error rates can be measured, improved and guaranteed. QoS is especially important to real-time traffic like voice and streaming multimedia, where any data loss or delay would result in unacceptable audio or image quality. Delivering sufficient QoS is an essential aspect of modern IP networks.
A router is a device that transmits data packets between networks. Packets are forwarded from one router to the next across a number of interconnected networks from source to destination. When a packet arrives on a router’s incoming interface, the router checks the packet’s network address information and uses its routing table to determine the appropriate outbound interface on which to retransmit the packet.
RTP – Real-time Transport Protocol
The Real-time Transport Protocol is a protocol used to transmit audio and video across IP networks. RTP is used for applications such as VoIP, video conferencing and streaming television services. RTP works in conjunction with RTCP (RTP Control Protocol), with RTP carrying the media stream while RTCP monitors transmission statistics and QoS.
SIP – Session Initiation Protocol
Session Initiation Protocol is a signaling protocol used to control and manage peer-to-peer multimedia communication sessions for voice calls, video calls, and instant messaging between two or more endpoints connected over an IP network. SIP is a text-based protocol with a similar syntax to HTTP.
SIP Trunking is a service based on the Session Initiation Protocol in which Internet Service Providers offer telephony and unified communications services to customers equipped with an IP-PBX. With SIP Trunking, the traditional telephone trunk is replaced with an Internet connection, allowing organizations to combine voice, video and data on a single line.
A softphone is a computer application that allows users to make and receive VoIP telephone calls over the Internet. Softphone software may be installed on a personal computer, tablet, or even on a cellphone. Softphones are often used with a headset connected to the computer or mobile device. Softphones support a variety of codes including G.729 and G.711.
In networking, a switch is a piece of hardware that allows devices on a computer network to communicate via packet switching, whereby the switch receives, processes, and forwards incoming data packets to the appropriate destination. In telephony, a switch is a telephone exchange or PBX that connects two or more digital voice circuits based on dialed number or other criteria.
VoIP – Voice over Internet Protocol
Voice over Internet Protocol refers to voice communication conducted using hardware or software-based IP Phones over the Internet or private IP networks, as opposed to the traditional circuit-based PSTN.
A VoIP gateway is an interface between the PSTN and a private or public IP network. It compresses, packetizes, routes, signals and decompresses analog voice, video and fax traffic for digital transmission.