These are terms, acronyms, definitions and more in-depth analysis of some of the terms relating to Voice Over IP.

Analog audio signals - Analog audio signals are used to transmit voice data over telephone lines. This is done by varying or modulating the frequency of sound waves to accurately reflect the pitch of the sound. The same technology is used for radio wave transmissions.

ATA - ATA or the analog telephone adaptor is the hardware device that connects the conventional telephone to the Internet through a high speed bandwidth line, provides the interface to convert the analog voice signals into IP packets, delivers dial tone and manages the call setup.

Audio encoding - The ITU has defined multiple audio codecs for use with H.323. All of them are also compatible with SIP, which is codec-agnostic.

G.711 is 3 kHz audio encoded at 64-kbps. G.711 is PCM audio, the format used for voice delivery over traditional telephone networks and exchanges.

G.722 is high-quality 7kHz audio in 48-, 56-, or 64-kbps streams. Two lower-quality, narrow-band revisions exist: G.722.1 encodes the audio at 24- or 32-kbps, and G.722.2 encodes at around 16kbps.

G.723.1 is used for compressing speech at very low bit rates: 5.3- and 6.3-kbps.

G.728 is 3.4kHz audio encoded at 16-kbps, but uses much smaller packet sizes (.625 millisecond, as compared to 37.5ms for G.723.1) to guarantee low delays.

G.729 is a newer voice codec using 8-kbps streams and 15ms packet sizes. There are two variations, G.729 and G.729A, that differ only in their mathematical implementation.

Speex is an open source speech codec. In contrast to the G-series codecs listed above, it is not protected by patents. It encodes at variable bitrates, from 2.15- to 44.2-kbps.

GSM6.10 is another open source codec, encoding at 13.3-kbps. At this time there is an unresolved patent dispute surrounding the codec, but is still supported by multiple software programs.

Audio Menu - A verbal choice provided by a recording over the phone. Audio choice menus are common in automated attendant, IVR and fax-on-demand systems. They are prompts for caller input. Audio menus can instruct you to speak commands or hit touch-tones as commands.

Audio Response Unit (ARU) - A computer telephony system incorporating voice store and forward technology. There are both passive and interactive ARUs. Passive ARUs simply play out messages. Interactive ones play messages based on input from callers.

Audio Teleconferencing - Or Audio Conferencing. The original technology used for audio teleconferencing was based on PBX conferencing circuits. Setting up conference calls through the PBX is cumbersome, voice quality degrades as the number of people on a call increases and there are capacity limitations. As a result, specialized conference bridges were developed to improve capacity and voice quality. Conference bridges, however, require trained operator intervention to schedule and invoke most features. As a result, individual corporations find the cost of ownership prohibitive, and the market for such products has been concentrated on service bureau providers. Today, PC-based systems combine the freedom of conference bridges. By installing a conference server on your voice networks, you can set up, attend, and manage your own conferences over any touch-tone telephone. Additionally, users can schedule meetings using desktop software from their e-mail systems, or from a Web browser. The latest word in this area is having the endpoints themselves being able to provide local mixing, hence eliminating the need for network based conference servers!

Bandwidth - Bandwidth is the volume of data that can be transmitted over a communication line in a fixed amount of time. It is expressed in bits per second (bps) or bytes per second for digital devices and in cycles per second, or Hertz (Hz) for analog devices. Bandwidth can also be defined as the difference between a band of frequencies or wavelengths.

Call duration - The time interval between when the phone is taken off the hook for a test call and when it is put back on the hook.

Circuit switched networks - These networks have been used for making phone calls since 1878. They use a dedicated point-to-point connection for each call. This reduces their utility because no network traffic can move across the switches that are being used to transmit a call.

Client (Softphone client) - The software installed in the user’s computer to make calls over the Internet.

Call hunting - A calling feature for inbound calls that will "roll past" a busy signal or try multiple numbers until the call is answered.

Class 5 (Telephony) switch - A Class 5 switch, in United States telephony jargon refers to a telephone switch or exchange located at the local telephone company's central office, directly serving subscribers. Class 5 switch services include basic dial-tone, calling features, and additional digital and data services to subscribers using the local loop. A key part of SIP/VoIP/IMS networks/systems are IP based class 5 switches (In the IMS environment they are known as class 5 App Servers).

Clipping - The loss of speech-signal components, resulting in the dropping of the initial or end parts of a word or words.

Codec - Codec is a term that arises from the Compressor-Decompressor or enCOder/DECoder process. It is used for software or hardware devices that can convert or transform a data stream. For instance, at the transmitting end codecs can encode a data stream or data signal for easy transmission, storage or encryption. At the receiving end, they can decode the signal in the appropriate form for viewing. They are most suitable for videoconferencing and streaming media solutions.

Compression - This is a term that is used to indicate the squeezing of data in a format that takes less space to store or less bandwidth to transmit. It is very useful in handling large graphics, audio and video files.

Conference Bridge - A device used to connect multiple parties over the phone. A proctor or operator can man conference bridges, or they can be supervised. There are both stand-alone conference bridges and conference bridge functions built in to some PBXs (Private Branch Exchange). These systems have circuitry for summing and balancing the energy (noise) on each channel so everyone can hear each other. More sophisticated conference bridges have the ability to "idle" the transmit side of channels of non- speaking parties. Some conference bridges use "clVoxising" to idle or reject the input of touch tones or other signals. There are VoIP based Conference Bridge servers. They may be controlled via protocols such as SIP or Megaco. they send/receive media by using the RTP protocol.

Data compression - This is the process that is used to compress large data files into mall files so that they use less bandwidth during transmission and less disk space when stored. The compression depends upon the repeatable patterns of binary 0s and 1s. The higher the number of repeatable patters, the higher is the compression. The right compression codes can compress data files to 40% of their original size. The graphics files can be compressed even more – from 20% to 90%.

Dial Plans - Dial Plans are sequences of characters used to translate dialed numbers into outbound dial strings. Dial Plans can be used as filters; to allow, disallow or manipulate dialed numbers. If a dialed number, in the device, matches a set Dial Plan the device will then transmit the dialed numbers outbound.

Dial Plans can be used to prevent calls to certain destinations such as 411 and International Numbers or to add in an area code for 7-digit dialing. Dial Plans look very similar from manufacturer to manufacturer but are not always the same. Most dial plans are based on the (MGCP RFC 3435) Dial Plan but have modifications for various reasons.

Dial-tone delay - The time interval, measured in milliseconds, between when a phone is taken off the hook and when a dial tone sounds.

DNS - A computer program running on a web server, translating domain names into IP addresses. In the last years special types of domain names records were added to the DNS world-wide system, which provide support to SIP/VoIP (SRV/NAPTR, ENUM).

Dual-tone multifrequency (DTMF) - The system used by touch-tone telephones. DTMF assigns a specific frequency (made up of two separate tones) to each key so that it can easily be identified by a microprocessor. This is basically the technology behind touch tone dialing.


Emergency 911 calls - This is an emergency telephone number that handles all calls related to police, fire or medical emergencies. The number, which is allotted under the North American Numbering Plan (NANP), is answered by either a telephone operator or an emergency service dispatcher, who, in turn, alerts the appropriate emergency service.

Find-me/follow-me - A feature that allows calls to find you wherever you are, ringing multiple phones (such as your cell phone, home phone, and work phone) all at once.

Frame Relay - In data communications, a packet switching method that uses available bandwidth only when it is needed. This fast packet switching method is efficient enough to transmit voice communications with the proper network management.

Full Duplex - In telephony and data communications, the ability for both ends of a communication to simultaneously send and receive information without degrading the quality or intelligibility of the content.

Gateway In VoIP systems - A network device that converts voice and fax calls in real time from the public switched telephone network (PSTN) to an IP network.

H.323 - An ITU standard that lays down guidelines for real time voice and videoconferencing utilities on the Internet. The H.323 standard supports voice, video, data, application sharing and whiteboarding and defines media gateways for conversion to packets.

High-availability - Refers to devices or deployment strategies designed to provide access to fully functioning systems at all times. One such strategy is to cluster devices so that the primary device can fail over to the secondary one if necessary.

Interactive Voice Response IVR - In computer telephony, Interactive Voice Response is a horizontal application wherein computer-based information is accessed over the phone - with a telephone versus a computer. An IVR platform uses computer telephony components to translate callers' touch-tones or voice commands into computer queries after the callers hear an audio menu. IVR systems can also be used for callers to change the information in a database instead of just "listen" to the information.

Internet - The current-day public and global computer network or "information super-highway." The Internet is an outgrowth and combination of a variety of university and government sponsored computer networks. Federal and private sector subsidies supported the DARPA-NET. NSFnet (National Sciences Foundation) and thousands of other subnetworks, which were used to do inter-agency research and communication. Today, the Internet is made up of millions upon millions of computers and subnetworks - almost entirely supported by commercial funds except in countries where deregulation has not occurred. The internet is the substrate and chief communications backbone for the World Wide Web (WWW), the "graphical interface" of the Internet.

Internet congestion - Internet congestion occurs when a large volume of data is being routed on low bandwidth lines or across networks that have high latency and cannot handle large volumes. The result is slowing down of packet movement, packet loss and drop in service quality.

IP - IP, which is the acronym for Internet Protocol, defines the way data packets, also called datagrams, should be moved between the destination and the source. More technically, it can be defined as the network layer protocol in the TCP/IP communications protocol suite.

IP address - An IP address, also known as Internet Protocol address, is the machine number used to identify all devices that are connected to the net. Each device has its own unique number which it uses to communicate. This number is fixed in the case of those computing devices that have a fixed IP address. The rest are allotted a dynamic IP address, which is valid for the period they are connected to the net. The numbers range from 0.0.0.0 to 255.255.255.255.

IP mapping - IP mapping is the process of identifying IP addresses on the basis of their geographical locations. The mapping enables web administrators to pinpoint the location of any computing device connected to the Internet.

IP Phone - AKA Internet Phone, SIP Phone or VoIP Phone. An IP phone is one that converts voice into digital packets and vice versa to make phone calls over Internet possible. It has built-in IP signaling protocols such as SIP or H.323 that ensure that the voice is routed to the right destination over the net. On the media side the IP Phone uses audio or/and video codecs such as G.711 or/and H.261 respectively over RTP. The IP phones come with several value added services like voicemail, e-mail, call number blocking etc.

ISP - Internet Service Provider. A business that provides subscriber-based access to the Internet. Subscribers can be individuals or businesses/ ISPs operate at the fourth or lowest level of the Internet. At the third level, regional providers aggregate traffic from lower-order ISPs to the second, backbone level. The highest level in North America is the NAP (Network Access Point), which act as peer-to-peer interconnection points for the largest backbones. There are three "official" NAPs located in San Francisco, Chicago and Pennsauken, New Jersey. ISPs use both Internet Routers, Servers and Rack-Mounted modems to provide a variety of services including Web Site hosting, FTP service, e-mail accounts, unified messaging, audio and video broadcasting and in some cases - Internet Telephony and Fax Gateway service.

ITU - ITU, which is the acronym of International Telecommunication Union, is a telecommunications standards body based in Geneva. It works under the aegis of the United Nations and makes recommendations on standards in telecommunications, information technology, consumer electronics, broadcasting and multimedia communications.

Jitter - It is a term used to indicate a momentary fluctuation in the transmission signal. This happens in computing when a data packet arrives either ahead or behind a standard clock cycle. In telecommunication, it may result from an abrupt variation in signal characteristics, such as the interval between successive pulses.

Kbps - Kbps is the acronym for kilobits per second and is used to indicate the data transfer speed. If the modem speed, for instance, is 1 Kbps then it means that the modem can route data at the speed of one thousand bits per second.

Lag - Lag is the term used to indicate the extra time taken by a packet of data to travel from the source computer to the destination computer and back again. The lag may be caused by poor networking or by inefficient or excessive processing.

Latency - Latency is the time that elapses between the initiation of a request for data and the start of the actual data transfer. This delay may be in nanoseconds but it is still used to judge the efficiency of networks.

Mean opinion score (MOS) - A measurement of the subjective quality of human speech, represented as a rating index. MOS is derived by taking the average of numerical scores given by juries to rate quality and using it as a quantitative indicator of system performance.

Messaging - In computer telephony, any means of message store and forward. This includes fax mail, voice mail and broadcast messaging. This horizontal application is the most popular of all other voice solutions. Messaging systems provide for the store and forward of "non-real time" communication. For example, a recorded voice message can be stored for later play back either locally or remotely, or a fax can be received and stored before it is re-transmitted to the ultimate recipient. Messages, then, can vary in content and media type - the distinction being that they are recorded or stored for pick up in the future.

NANP - Stands for North American Numbering Plan. A telephone numbering system that has evolved the way area codes and numbers are allotted. The system was established in 1947 and covers the United States, Canada and a few neighboring areas. It uses a three-digit area code and seven-digit telephone numbers. Its fiat is, however, limited to the public switched telephone networks only.

Packet - A logically grouped unit of data. Packets contain a payload (the information to be transmitted), originator, destination and synchronizing information. The idea with packets is to transmit them over a network so each individual packet can be sent along the most optimal route to its. Packets are assembled on one end of the communication and re-assembled on the receiving end based on the header addressing information at the front of each packet. Routers in the network will store and forward packets based on network delays, errors and re-transmittal requests from the receiving end.

Packet loss - Packet loss is the term used to indicate the loss of data packets during transmission over a computer network. This may happen on account of high network latency or on account of overloading of switches or routers that are unable to process or route all the incoming data.

Packet Switching - A means of economically sending and receiving data over alternate, multiple network channels. The premise for packet switching is the packet, a small bundle of information containing the payload and routing information. Packet switching takes data, breaks it down into packets, transmits the packets and does the reverse on the other end. Packets can be sent in order and then be received in a different order - only to be put back in the correct order in seconds.

PBX - Private Branch Exchange. Or PABX (Private Automatic Branch Exchange). In telephony, a PBX system behaves as a customer's premises over trunk lines (thus the term "branch"). At first, PBXs mimicked a small telephone company switchboard. Users would use an operator to take and make telephone calls to and from the PSTN (Public Switched Telephone Network). Over time, users were able to dial directly, without the use of an operator. Today, computer telephony platforms such as automated attendants are able to route incoming calls automatically, too.

Peer-to-Peer (P2P) - The term peer-to-peer is used to indicate a form of computing where two or more than two users can share files or CPU power. They can even transmit real time data such as telephony traffic on their highly ad hoc networks. Interestingly, the peer-to-peer network does not work on the traditional client-server model but on equal peer nodes that work both as "clients" and "servers" to other nodes on the network.

POP - Point of Presence, equivalent of a local phone company's central office. The place your long distance carrier terminates your long distance lines just before those lines are connected to your local phone company's lines, or to your own direct hookup.

Alternate Definition: Post Office Protocol. An Internet standard for the storage and retrieval of email messages

POTS (plain old telephone service) - The typical, familiar model of a single phone line and a single phone number.

Protocol - It is a convention or standard that defines the procedures to be adopted regarding the transmission of data between two computing end points. These procedures include the way the sending device should sign off a message or how the receiving device should indicate the receipt of a message. Similarly, the protocols also lay down guidelines for error checking, data compression, and other relevant operational details.

PSTN - Public Switched Telephone Network. The combination of local, long-distance, and international carriers that make up the worldwide telephone network.

QoS (quality of service) - The ability of a network (including applications, hosts, and infrastructure devices) to deliver traffic with minimum delay and maximum availability.

Real Time - A communication wherein any perceptible delay between the sender and receiver are minimal and tolerated. Regular telephone calls are real time. Point-to-point fax transmissions are "close" to real time. Voice messaging is in non-real time.

Router - A router is a network device that that handles message transfer between computers that form part of the Internet. The messages, which are in the form of data packets, are forwarded to their respective IP destinations by the router. A router can also be called the junction box that routes data packets between computer networks.

Sample Rate - This is the number of samples of audio carried per second, measured in Hz or kHz (one kHz being 1 000 Hz). For example, 44 100 samples per second can be expressed as either 44 100 Hz, or 44.1 kHz. Bandwidth is the difference between the highest and lowest frequencies carried in an audio.

Service Provider - An addressable entity providing application and administrative support to the client environment by responding to client requests and maintaining the operational integrity of the server.

SIP (Session Initiation Protocol) - An Internet Engineering Task Force (IETF) standard for initiating, maintaining, and terminating an interactive user session involving video, voice, chat, gaming, virtual reality, and more.

SIP phone (Also see above IP Phone) - A SIP phone is a telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the Internet for signaling (and uses RTP for media). The SIP phones come with several value added services like voicemail, e-mail, call number blocking etc. There are (normally) no charges for making calls from one SIP phone to another, and negligible charges for routing the call from a SIP phone to a PSTN phone.

Soft phone - IP telephony software that lets users send and receive calls from non-dedicated hardware, such as a PC or Pocket PC device. It is typically used with a headset and microphone.

Soft switch - It is a software application that is used to keep track of, monitor or regulate connections at the junction point between circuit and packet networks. This software is loaded in computers and is now replacing hardware switches on most telecom networks.

TCP - Transmission Control Protocol. The transport layer protocol developed for the ARPAnet which comprises layers 4 and 5 of the OSI model. TCP controls sequential data exchange in TCP/IP for remotely hosts in a peer-to-peer network.

Telephony - Taken from Greek root words meaning "far sound", telephony is the discipline of converting or transmitting voice or other signals over a distance, and then re-converting them to an audible sound at the far end.

Video encoding - There are fewer video codecs (than audio codecs) associated with the H.323 and SIP protocol suites.

H.261 is a video codec use for wideband (>= 64Kbps). H.263 is used for narrowband (< 64-kbps). Both are widely supported.

H.264 is a newer narrowband codec that produces higher-quality results than H.263 and is recommended in its place. H.264 is also known as ISO 14496-10 and as MPEG-4 part 10 and as MPEG-4 AVC (Advanced Video Coding)..

VoIP (Voice over IP) - The process of making and receiving voice transmissions over any IP network. IP networks include the Internet, office LANs, and private data networks between corporate offices. The main advantage of VoIP is that users can connect from anywhere and make phone calls without incurring typical analog telephone charges, such as for long-distance calls.

VOIP Gateway - This device provides the conversion interface between the public switched telephone network (PSTN) and an IP network for voice and fax calls. Its primary functions include: voice and fax compression/decompression, packetization, call routing and control signaling. It also provides an interface to Gatekeepers or Softswitches, billing systems, and network management systems.

VOIP PBX - VoIP PBX, which stands for Voice over Internet Protocol Private Branch eXchange, is a telephone switch that converts IP phone calls into traditional circuit-switched TDM connections. It also supports traditional analog and digital telephones.

VOIP Phone - A VoIP phone is one that uses the Internet to route voice calls by converting the voice data into IP packets and vice versa. The phones come with built-in IP signaling protocols such as H.323 or SIP that help in the routing of data to the right destination. A VoIP phone can also be a software application that is installed in the user's PC. In this case it is known as the Softphone. Also, the calls in this case have to be made from the PC, and not through a telephone instrument.

Web Browser - Client software used to view information on Web servers. Can display graphics. Web browsers are also packaged with email clients, newsreaders and in some cases, IP Telephony clients.

Web Server - On the World Wide Web, a server dedicated to storing data (such as Web pages in HTML format) and distributing it to Web Browsing users. Web browsers are able to download video, text, still images and audio from Web Pages. Some servers support Unified Messaging.