| AHT (Average Hold Time) | The average length of time between the moment a caller finishes dialing and the moment the call is answered or terminated. |
| ANI (Automatic Number Identification) | A telephone function which transmits the billing number of the incoming call (Caller ID, for example). |
| ANSI (American National Standards Institute) | The American standardization body known for interface recommendations and standardization of programming languages. ANSI is a non-profit making, government-independent organization. |
| ASR (Answer-Seizure Ratio) | The ratio of successfully connected calls to attempted calls (also called 'Call Completion Rate'). |
| ATA (Analogue Telephone Adapter) | Used to connect a standard telephone to a high-speed modem to facilitate VoIP and/or fax calls over the Internet. |
| Broadband | A descriptive term for evolving digital technology that provides consumers a single switch facility offering integrated access to voice, high-speed data service, video demand services, and interactive delivery services. |
| Call Deflection | Call Deflection allows a called endpoint to redirect the unanswered call to another endpoint. |
| CDR (Call Detail Record) | Information regarding a single call collected from the switch and available as an automatically generated downloadable report for a requested time period. The report contains information on the number of calls, call duration, call origination and destination, and billed amount. |
| CLEC (Competetive Local Exchange Carrier) | This is a company that competes for business with an ILEC. |
| CODEC (Compression-decompression) | In VoIP it is a voice compression-decompression algorithm that defines the rate of speech compression, quality of decompressed speech and processing power requirements. The most popular codecs in VoIP are ITU-T G.723.1 and G.729 (AB). |
| Common Carrier | An entity licensed by the FCC or a state agency to supply local and/or long distance telecommunications services to the general public at established and stated rates. |
| Compression | Compression is used at anywhere from 1:1 to 12:1 ratios in VOIP applications to consume less bandwidth and leave more for data or other voice/fax communications. The voice quality may decrease with increased compression ratios. |
| Congestion | The situation in which the traffic present on the network exceeds available network bandwidth/capacity. |
| DID (Direct Inward Dial) | A specially configured phone line from the telephone company that allows for dialing inside a company directly without having to go through an attendant. A DID line cannot be used for outdial operation since there is no dialtone offered. However, it can be configured so an outside caller can reach an inside extension with a 7-digit number through the phone company's central office. |
| DTMF (Dual-Tone Multi Frequency) | The type of audio signals generated when you press the buttons on a touch-tone telephone. |
| Dynamic Jitter Buffer | Collects voice packets, stores them, and shifts them to the voice processor in evenly spaced intervals to reduce any distortion in the sound. |
| Firewall | A system designed to prevent unauthorized access to or from a private network. Firewalls can be implemented as hardware, software, or a combination of both. All messages entering or leaving the intranet pass through the firewall, which examines each message and blocks those that do not meet the security criteria specified on the firewall. |
| FOIP (Fax Over Internet Protocol) | The term used for the technology that transports facsimiles over the Internet. |
| G.711 | An ITU-T PCM half-duplex codec that uses either A-law or ?-law compression (64 kbps, high quality, minimum processor load). |
| G.723.1 | An ITU-T double rate CELP codec (6.4/5.3 kbps, medium quality, high processor load). |
| G.726 | An ITU-T ADPCM wave form codec (16/24/32/40 kbps, good quality, low processor load). |
| G.728 | An ITU-T low delay CELP codec (16 kbps, medium quality, very high processor load). |
| G.729 | An ITU-T ACELP codec (8 kbps, medium quality, high processor load). |
| Gateway | In IP telephony, a network device that converts voice and fax calls, in real time, between the public switched telephone network (PSTN) and an IP network. The primary functions of an IP gateway include voice and fax compression/ decompression, packetization, call routing, and control signaling. Additional features may include interfaces to external controllers, such as Gatekeepers or Softswitches, billing systems, and network management systems. |
| ILEC (Incumbent Local Exchange Carrier) | Incumbent Local Exchange Carrier. A telephone company in the U.S. that was providing local service when the Telecommunications Act of 1996 was enacted. ILECs include the former Bell operating companies (BOCs), which were grouped into holding companies known collectively as the regional Bell operating companies (RBOCs) when the Bell System was broken up by a 1983 consent decree. ILECs are in contradistinction to CLECs (competitive local exchange carriers). A "local exchange" is the local "central office" of an LEC. Lines from homes and businesses terminate at a local exchange. Local exchanges connect to other local exchanges within a local access and transport area (LATA) or to interexchange carriers (IXC) such as long-distance carriers AT&T, MCI, Qwest and Sprint. |
| Integrated T-1 | Comprised of 24 64Kbps channels, T1 lines can be used for a diverse number of applications. Commonly referred to as an integrated T1 or channelized T1, this highly flexible circuit is designed for businesses that need to run multiple services over the same line. Common applications for integrated T1 service include, Frame Relay/dedicated long distance and Internet/point-to-point. Often confused with a fractional T1, integrated service is made up of multiple fractional T1 services. |
| IP Telephony | The transmission of voice and fax phone calls over data networks that uses the Internet Protocol (IP). IP telephony is the result of the transformation of the circuit-switched telephone network to a packet-based network that deploys voice-compression algorithms and flexible and sophisticated transmission techniques, and delivers richer services using only a fraction of traditional digital telephony's usual bandwidth. |
| ITSP (Internet Telephony Service Provider) | Provider of telephony based services. |
| Jitter | The variation in the amount of Latency among Packets being received. |
| LAN (Local Area Network) | A LAN is a group of computers and associated devices that share a common communications line or wireless link and typically share the resources of a single processor or server within a small geographic area (for example, within an office building). |
| Latency | Also called Delay. The amount of time it takes a Packet to travel from source to destination. Together, Latency and Bandwidth define the speed and capacity of a network. |
| NAT | Also known as Network Address Translation. A networking protocol that allows network of private IP address to be set up using a single real IP address. Using NAT, a local area network (LAN) can be set up with no special configuration of the Internet connection. To the Internet, the network looks like one computer, but on the LAN, each computer has its own internal IP address. |
| Off-Net | Not served, or not able to be served, using a given service provider’s facilities. A call originating (or terminating) off-net has originated (or terminated), not on the network managed directly by the subscriber’s service provider, but on another service provider’s network. The latter provider usually bills the former provider for roaming or for service resale on a wholesale basis; the subscriber’s provider, in turn, usually bills the end user on a retail basis. |
| On-Net | Served, or able to be served, using a given service provider’s facilities. A call originating (or terminating) on-net has originated (or terminated) on the subscriber’s service provider’s managed network. |
| Origination | Refers to calls that originate in the PSTN public switched telephone network and are carried to their destination over the Internet. Telephone communication networks use various theories on multi-paneled circuitry integration, network distance reduction algorithms, switch theory, chaos non-linear dynamics, randomized multi-processed information, game theory, and wave-particle theorem of mass and energy stability over various virtual static networks, to implement origination. The topic is often considered to be the most complex matter in VOIP, as it is vital to the integration and maximization of potential between the broadband connections, and telecommunication wires. |
| Packet | In data communication, the basic unit of information transferred. |
| PBX (Private Branch Exchange) | An in-house telephone switching system that interconnects telephone extensions to each other, as well as to the outside telephone network. |
| PRI (Primary Rate Interface) | An ISDN service that provides 23 64-Kbps B (Bearer) channels and one 64-Kbps D (Data) channel (23 B and D). |
| PSTN (Public Switched Telephone Network) | The worldwide set of interconnected switched voice telephone networks that deliver fixed telephone services to the general public and are usually accessed by telephones, key telephone systems, private branch exchange trunks, and certain data arrangements, transmitting voice, other audio, video, and data signals. Completion of a PSTN circuit between the call originator and the call receiver requires network signalling in the form of either dial pulses or multifrequency tones. The PSTN includes local loops; short-haul trunks; long-haul trunks, including international links; exchanges; and switching technology. |
| QoS (Quality of Service) | Measure of performance for a transmission system that reflects it's transmission quality and service availability. Standards based QOS for VoIP usually involves the implementation of Ethernet standards 802.1p and 802.1q at layer 2 across an Ethernet. |
| Router | Typically a machine, though it can also be software, that acts as a gateway to provide access to network resources, irrespective of the protocols or operating systems the users use. |
| RSVP (Resource Reservation Protocol) | A protocol that supports the reservation of resources across an IP network. Applications running on IP end systems can use RSVP to indicate to other nodes the nature (bandwidth, jitter, maximum burst, and so on) of the packet streams they want to receive. RSVP depends on IPv6. Also known as Resource Reservation Setup Protocol. |
| RTP (Real-Time Transport Protocol) | Commonly used with IP networks. RTP is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services. RTP provides such services as payload type identification, sequence numbering, time stamping, and delivery monitoring to real-time applications. |
| SIP (Session Initiation Protocol) | An application-layer control protocol, a Signaling protocol for Internet Telephony. SIP can establish sessions for features such as audio/videoconferencing, interactive gaming, and call forwarding to be deployed over IP networks thus enabling service providers to integrate basic IP telephony services with Web, e-mail, and chat services. In addition to user authentication, redirect and registration services, SIP Server supports traditional telephony features such as personal mobility, time-of-day routing and call forwarding based on the geographical location of the person being called. |
| Softswitch | Also called a Proxy Gatekeeper, Call Server, Call Agent, Media Gateway Controller, or Switch Controller. Software used to bridge a public switched telephone network and voice over Internet by separating the call control functions of a phone call from the media gateway (transport layer). Softswitch performs call control functions such as protocol conversion, authorization, accounting and administration operations. |
| T1 | 1.544-Mbps point-to-point dedicated digital circuit provided by the telephone companies consisting of 24 channels. |
| TAP (Telephony Application Programming Interface) | A programming interface that allows Windows client applications to access voice services on a server. |
| TCP (Transmission Control Protocol) | Connection-oriented transport layer protocol that provides reliable full-duplex data transmission. TCP is part of the TCP/IP protocol stack. |
| Termination | Placing calls that terminate in the PSTN public switched telephone network. |
| Trunk | A communications channel between two points, typically referring to large-bandwidth telephone channels between switching centers, that handle many simultaneous voice and data signals. |
| Trunking | Trunking means that several connections in a network may be established simultaneously, and that setup of connections proceeds automatically using the channels available at the time in question. In this way many users may share a few connections, and if the number of connections is increased, the capacity of the network is increased more than proportionally. This means that an optimal trunking effect is obtained in very large networks. |
| VoIP (Voice Over Internet Protocol) | Transportation of voice calls across the Internet. |