ACD: Average Call Duration.
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.
AS (Autonomous System): A group of networks under mutual administration that share the same routing methodology.
ASP (Application Service Provider): An independent, third party provider of software-based services delivered to customers across a wide area network (WAN).
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.
ATM (Asynchronous Transfer Mode): A technology for switched, connection-oriented transmission of voice, data and video. It makes high-speed dedicated connections possible between a theoretically unlimited number of network users and also to servers.
Asterisk: An open source that provides all the functionality of high-end business telephone systems. It is the world's most flexible and extensible telephone system, providing many features that are not yet available in even the most advanced proprietary systems. It is also the world's cheapest telephone system. The software is free and runs on inexpensive Linux servers.
Backbone: A high-speed network spanning the world from one major metropolitan area to another.
Bad Frame Interpolation: Interpolates lost/corrupted packets by using the previously received voice frames. It increases voice quality by making the voice transmission more robust.
Bandwidth: The maximum data carrying capacity of a transmission link. For networks, bandwidth is usually expressed in bits per second (bps).
Billing Increment: A call duration measurement unit, usually expressed in seconds.
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.
CALEA (Communications Assistance for Law Enforcement Act): A 1994 act that requires telecommunications services to provide wiretapping access. The act specifically excludes information services, so the question is whether VoIP is a telecommunications service, and thus covered by the act, or an information service, and thus exempted. VoIP providers are receiving pressure to comply with the act.
Call Deflection: Call Deflection allows a called endpoint to redirect the unanswered call to another endpoint.
Call Detail Record (CDR): 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.
Circuit-Switched: Communication system that establishes a dedicated channel for each transmission. The copper-wire telephone system (POTS) uses circuit-switching, as do PBX systems. Dedicated channels mean strong reliability and low latency, but the downside is that only one type of communication can use the channel at any given time.
CLEC (Competitive Local Exchange Carrier): A telephone company that competes with the larger incumbent carriers (ILECs) through reselling the ILEC services and/or creating services that use the ILEC's infrastructure. The Regional Bells are ILECs; local phone companies are frequently CLECs.
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 G.723.1 and G.729.
Compression: VoIP uses various compression ratios, the highest approximately 12:1. Compression varies according to available bandwidth.
Congestion: The situation in which the traffic present on the network exceeds available network bandwidth/capacity.
CSMA/CD (Carrier Sense Multiple Access/Collision Detection): This is the access procedure to the Ethernet in which the participating stations physically monitor the traffic on the line. If no transmission is taking place at the time the particular station can transmit. If two stations attempt to transmit simultaneously this causes a collision that is detected by all participating stations. After a random time interval the stations that collided attempt to transmit again.
Dial-peer (Addressable Call Endpoint): A software structure that binds a dialed digit string to a voice port or IP address of the destination gateway. Several dial peers always exist on each router in the network, and at least two will be involved in making a call across the network, one on the originating end and one on the terminating end. In Voice over IP, there are two kinds of dial peers: POTS and VoIP. VoIP peers point to specific VoIP devices.
Dial-peer hunting: Process when the originating router tries to establish call on different dial peers if the originating router receives a user-busy invalid number or an unassigned-number disconnect cause code from a destination router.
DiffServ (Differentiated Services): A quality of service (QoS) protocol that prioritizes IP voice and data traffic to help preserve voice quality, even when network traffic is heavy.
DNIS (Dialed Number Identification Service): A telephone function which sends the dialed telephone number to the answering service.
DSP (Digital Signal Processors): All digital audio systems use DSP technology in order to differentiate between signal and noise. In telephone communication, too, much noise creates problems in maintaining connections, and in VoIP systems the DSP component provides features such as tone generation, echo cancellation, and buffering.
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.
E911 (Enhanced 911): Technology allowing 911 calls from cellular phones to be routed to the geographically correct emergency station (PSAP: Public Safety Answering Point). VoIP users currently have limited access to 911 services, and with some providers none, because VoIP is not geographically based.
FCC (Federal Communications Commission): The regulator of telephone and telecommunications services in the United States. It's not yet known the full extent to which the FCC will regulate VoIP communications. Part of the complication lies with determining the regulation of communications that begin or end on an FCC-regulated system, such as the standard telephone service.
Firewall: Security software or appliance that sits between the Internet and the individual PC or networked device. Firewalls can intercept traffic before it reaches network routers and switches, or between router/switch and PC, or both. Because the job of firewalls is to prevent access from specific packets over specific network ports, some must be specially configured to allow VoIP traffic to pass through.
FoIP (Fax over Internet Protocol): The fax counterpart to VoIP, available from some providers either free or at additional cost. FoIP is actually more reliable than VoIP because of its tolerance for poor latency.
H.323: The standard call protocol for voice and videoconferencing over LANs, WANs, and the Internet, allowing these activities on a real-time basis as opposed to a packet-switched network. Initially designed to allow multimedia to function over unreliable networks, it's the oldest and most established of the VoIP protocols. See also SIP and MGCP.
Latency: The time it takes for a packet to travel from its point of origin to its point of destination. In telephony, the lower the latency, the better the communication. Latency has always been an issue with telephone communication taking place over exceptionally long distances (the United States to Europe, for example). With VoIP, however, latency takes on a new form because of the splitting of the message into packets (see packet-switched) and network delay in general.
MGCP (Media Gateway Control Protocol): Another protocol competing with H.323 (see also SIP). MGCP handles the traffic between media gateways and their controllers. Especially useful in multimedia applications: the media gateway converts from various formats for the switched-circuit network, and the controller handles conversion for the packet-switched network. Designed to take the workload away from IP telephones themselves and thereby make IP phones less complex and expensive.
Packet-switched: Communication system that chops messages into small packets before sending them. All packets are addressed and coded so they can be recompiled at their destination. Each packet can follow its own path and therefore can work around problematic transmission segments. Packet switching is best when reaching a destination is the primary concern and latency is permissible, such as sending e-mail and loading Web pages.
PBX (Private Branch Exchange): A privately owned system for voice switching and other telephone related services. It routes calls from the public telephone system within an organization and allows direct internal calls.
PDD (Post Dial Delay): When a telecom switch is trying to establish the best possible route for the call.
POTS (Plain Old Telephone Service): Nothing more than a standard telephone line, the kind Ma Bell and then AT&T handled exclusively before the deregulation of the telephone industry. Upgrade your POTS to DSL, and you have broadband; add VoIP, and you have a system that uses POTS, the PSTN and the Internet in one seamless system.
PSTN (Public Switched Telephone Network): The network of wires, signals, and switches that lets one telephone connect to another anywhere in the world. Some VoIP services provide a gateway from the Internet to the PSTN and vice versa.
RTP (Real Time Protocol): Also known as Real Time Transport Protocol. Controls the transmission of packets of data that demands low latency (such as audio and video). Supports real-time transmission over IP networks and streaming as one means of delivery.
QoS (Quality of Service): Refers to the quality of the voice call over a VoIP network. A major issue in VoIP communications, because the high quality of telephone calls has always been taken for granted. Latency, packet loss, network jitter, and many other factors contribute to QOS measurements, and numerous solutions have been offered by vendors of routers and other network components.
SIP (Session Initiation Protocol): Communication protocol that operates similarly to H.323 but is less complex and more Internet- and Web-friendly. Fully modular and designed from the ground up for functioning over IP networks, it can be tailored more easily than H.323 for Internet applications. SIP and H.323 can and do coexist.
SoftPhone: A software app that gives you the ability to make and receive calls over the Internet using your PC and a headset or a microphone and speakers. A softphone's interface can look like a traditional phone dial pad or more like an IM client.
Universal Service: The availability of affordable telecommunications technology for all Americans, part of the 1966 Telecommunications Act, and regulated by the FCC. Current discussions revolve around the applicability of VoIP to universal services and whether or not VoIP providers should be taxed accordingly.
Virtual Phone Number: A feature of VoIP that allows you to attach additional phone numbers with different area codes to your basic VoIP service. This feature allows people to phone you without incurring long-distance charges from the same or adjacent nontoll area codes. All outgoing calls, however, are billed as if coming from your main phone number. Virtual phone numbers typically each cost a few extra dollars per month.
VoIP (Voice over Internet Protocol): The technology behind Internet phones. VoIP works by digitizing voice signals and sending them as packets through the same networking channels as your data.
Voip Escrow was created to provide a safe and secure platform for buyers and sellers of "minutes" to conduct business. This specificly applies to a segment of the industry (primarily non-US) dealing in call origination and termination points for VoIP traffic servicing targeted customer populations (e.g. Midle East, Eastern Europe, SE Asia). The service acts as a middle-man who protects the buyer by assuring the buyer they receive minutes they have ordered, and protecting the seller by ensure the money is available to him/her for minutes they have provided.
Labels: Business VoIP, IP Telephony, VoIP Applications, VoIP Terms
Source: http://broadband-nation.blogspot.com/2012/07/ip-telephony-voip-terms-and-definitions.html
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