Switching to VoIP

An advanced, compressed version of plain PCM digitizing. This algorithm is used by the G.726 codec.

any unwanted signal that distorts a phone call by increasing the strength of the sound signal.

The scale used by PCM digitizing algorithms in many countries outside North America.

A buffered switching system that transports packets called cells across packet-based, often optical, data links.

A telephony application that provides interactive prompts and greetings for incoming calls to a PBX, so that calls can be routed without intervention from a human reception operator.

The signaling using to notify emergency dispatchers of a caller's postal address on the 911 system.

Also called automatic call delivery , a telephony application that routes inbound calls to the appropriate answering party based on a logic-based algorithm, such as the originating area code or the language the caller speaksi.e., Spanish or English, etc.

Nickname for incumbent local exchange carriers (ILECs), the companies that sprung from the AT&T antitrust breakup of the early 1980s.

An ITU signaling standard that allows two simultaneous phone calls on a digital phone line. The slang term for BRI is ISDN line .

A continual linear stream of binary data, such as those in a T1 or a TDM bus.

An area of high concentration of telephone connections and operators, usually using queuing and hunting applications.

Cisco Systems' Windows-based softPBX.

The building where an exchange switch resides, supporting telephone service for a geographically narrow group of PSTN subscribers, usually from a few hundred to tens of thousands.

A type of business phone service available from RBOCs that offers enhanced calling features.

A device that splits T1 circuits into digital channels or analog signals for use by TDM or analog phones, respectively.

A hardware-locating technique that enables Cisco's proprietary E911 ALI solution; also, the method by which Cisco PoE is requested from a Cisco switch.

An algorithm that quantizes and packages an analog signal for transport across a packet network. Common telephony codecs include G.711 and G.729A.

An algorithm used by some voice codecs to reduce the amount of bandwidth required to reliably transmit spoken sounds.

The simulation of white noise by an endpoint so that its listener is less aware of silence suppression during times in the call when nobody is speaking.

Refers to the group of QoS standards that use precedence and buffering to improve network availability for voice applications.

The point at which telephone company-owned facilities end and subscriber-owned facilities begin. Usually a cross-connect panel, DS1 mounting, or a similar wiring terminal.

In Cisco media gateways, a configuration for switching phone calls to an H.323, MGCP, SCCP, or SIP endpoint based on the digits signaled by the caller. Equivalent to Asterisk's Dial command, though much simpler.

A circuit connecting a softPBX to the PSTN that provides inbound and/or outbound calling service.

A QoS mechanism that provides class of service (CoS), precedence-based treatment of packets for voice applications, but no guarantee against capacity overages.

A single channel of a T Carrier circuit. T1s have 24 DS0s.

A T1 circuit.

A device that terminates one end of a T1 circuit, allowing its channels to be assigned for voice and data use. DSU/CSUs provide signaling and error checking for the T1's bitstream, too.

A standard for in- band signaling of dialed digits on legacy telephones.

An ITU convention for phone numbers on the PSTN and in H.323 VoIP networks. E.164 is commonly supported in MGCP and SIP networks, too.

The European equivalent of T1. E1 circuits provide eight additional DS0s per circuit than their North American cousins.

Enhanced 911 service; the signaling and human element protocols for handling of public safety dispatch calls in the United States. Predated by (nonenhanced) 911 service. The most significant enhancement of E911 is automatic location identification signaling that allows dispatchers to know the postal address of an emergency caller.

A unit of measure for voice traffic capacity in a PBX environment. Erlang ratings help system builders decide how many trunks are needed between two voice networks, like the PSTN and a PBX.

A freely available packet analysis software tool for Windows, Linux, and Macintosh.

The most common standard for local area networking connectivity. Modern Ethernet offers transmission speeds from 10 mbps to 10,000 mbps.

An alphanumeric code that has a programmed purpose on a PBX, such as calling a specific phone or voice mailbox. Legacy PBXs may support only numeric E.164 codes.

The national government agency in the United States that is charged with regulating communications systems that cross state lines, such as the Internet and long-distance voice networks.

A telephony application that can attempt to locate a mobile user on her desk phone, cell phone, satellite phone, or Internet phone, when the PBX receives a call for her.

A service level commitment that dictates that, when a caller tries to reach somebody on the PSTN, his call will be reliably connected 99.999% of the time.

A wide area networking technology that permits many subscribers to use the infrastructure of a long-distance carrier's network for transport of packet-based data.

Refers to the characteristic of data traveling simultaneously between two participants in both directions on the same pathway . A full-duplex speakerphone allows a caller to speak while listening to his calling partner. See also half-duplex .

The most common voice codec in telephony applications; it uses PCM to encode sound signals on an 8-bit quantization scale at the PSTN-standard sampling rate of 8 kHz. G.711 uses 64 kbps of bandwidth for a single one-way sound signal.

A device or software program that provides a proxy or intermediary between two systems with incompatible technologies. Example: a media gateway for connecting legacy telephony circuits like POTS lines to a VoIP server.

An ITU recommendation for the use of Ethernet and TCP/IP for voice connectivity.

Refers to the characteristic of data traveling in only one direction at a time between two participants in a conversation. A half-duplex speakerphone allows a caller to hear his partner's voice only while he himself is silent. See also full-duplex .

A group of extensions or phone company trunks that ring simultaneously or in sequence when one of them is called. (Simultaneous hunt groups are sometimes called ring groups.)

The interfaces used to connect legacy phone lines to a TDM bus, for instance to connect a POTS trunk to a PBX.

A TSP that links Asterisk users and IAX-compatible VoIP networks free of charge.

Call control signals that occur within the audio spectrum or electrical frequencies also used to transmit the sounds of the phone call. In the context of a T1, in-band means substitution of bits normally used for audio for bits used to indicate call control signals. See also robbed-bit signaling .

See regional Bell operating company (RBOC) .

A type of desktop computer application that allows users to send text messages, voice, and video back and forth using the Internet.

Pronounced "eye-add," a router with a built-in data link interface, such as a DSL modem or cable modem. An IAD is designed to replace two or more access devices on the customer premise . Some IADs may contain a router, DSL modem, firewall, Ethernet switch, VoIP ATA, and/or SIP proxy.

A family of ITU standards that dictate how transport and signaling work on digital telephone lines, including T1s.

A signaling and media packaging protocol designed to connect Asterisk systems. Unlike SIP, IAX is intended for use in telephony applications only and uses only one socket, so it's compatible with NAT firewalls. IAX is pronounced "eeks." The current version of IAX is 2.

The body that developed most of the standards in use on the PSTN and many cell phone networks. Formerly known as the CCITT, a French abbreviation that, roughly translated, means Consulting Committee for International Telephones and Telegraphs.

The advisory body responsible for the specification of SIP and many other Internet protocols.

The family of resource QoS reservation recommendations recommended by the IETF. Chief among them is RSVP.

A legacy device that allows several business phones to share several telephone company POTS trunks. Also known as a KSU (key system unit) or KTS (key telephony system).

See key system .

In traditional telephony, a two- or four-wire circuit that connects a telephone to a switch. See also trunk .

A communications network that links devices on a geographically narrow basisi.e., computers or phones in a single office. See also wide area network (WAN) .

A regional company that operates a local telephone network. Examples include SBC, Verizon, and ATX.

Nickname for the old AT&T national telephone network. Also, a pet name for regional Bell operating companies (RBOCs) that came from the antitrust breakup of AT&T.

The predominant technique for measuring users' opinions about the quality of their phone service.

A device that allows traditional telephony technologies, like T1 circuits, POTS lines, and analog phones, to be used with a VoIP network.

In SIP terms, a request from a SIP user agent to a SIP server agent.

An adjective that refers to sound latencyi.e., one-way latency from speaker to listener.

The scale used by PCM digitizing algorithms in North America and a few other parts of the world.

A stackable switching standard intended to augment and replace ATM.

An H.323 softphone with video and white-board capabilities that comes with Microsoft Windows operating systems.

The state a phone is said to be in when it is being dialed, when it has a call in progress, or when it hasn't been hung up.

An H.323 softphone available for Windows, Linux, and Mac OS X.

Call control signaling that occurs outside the audio spectrum and electrical frequency range used to carry the sound of the call. Out-of-band signaling may occur on a separate network altogether, like SS7, or just in a separate logical connection, like SIP. See also in-band signaling .

The noise that results when a sound signal is distorted because it was too strong for the transducer that played it back or the microphone that recorded it. See also additive noise .

A privately developed group of standards for enabling cable television network operators to deliver voice services using IP using the cable network as the carrier.

The amount of time that passes between the transmission of each frame in a VoIP sound stream. Inversely expressed as the packet rate.

A Windows software utility that, among other things, allows the user to see which routers have QoS enabled.

In Asterisk terms, a defined channel that can be used to route calls. SIP and IAX trunks are called peers. See also dial-peer .

In frame-relay networks, the point-to-point logical pathway established for use by the subscriber through the provider's network cloud.

The simplest kind of telephone service available from the phone company

A way of centrally powering IP phones over an Ethernet Cat5 cable plant using DC voltage. The common POE standard is 802.3af. Also known as inline power.

A CTI application that makes outbound calls from a database of phone numbers on behalf of an operator. Frequently used by telemarketers .

The notification of a user's status, often including her availability or desire to receive calls. Presence is common in instant messaging apps.

Sometimes called Prime in marketing materials, a signaling standard that permits 23 simultaneous voice calls on a T1 circuit using PCM encoding.

A device that provides centralized call-routing and telephony applications for a group of business telephones. PBXs use a TDM bus to support analog or digital endpoints and often have a built-in autoattendant and voice mail capability.

An office with a dispatcher who can receive calls from the PSTN on behalf of a local public safety jurisdiction. See also automated location identifier (ALI) .

The privately operated, publicly regulated global network that facilitates telecommunications.

The state-run agencies that regulate local telephone system operators' services.

The simplest digitizing algorithm for analog signals of audio or video, PCM is employed by the G.711 voice codecs in their North American and non-North American varieties, m law and Alaw.

A ubiquitous family of PSTN signaling recommendations that includes ISDN.

The general group of techniques and standards for improving performance in a VoIP network.

The specific group of standards that provide end-to-end bandwidth reservation for media channels on a converged network. One such standard is RSVP.

An American local telephone company that was born from the breakup of AT&T in 1983, also called ILEC.

An IETF QoS standard for bandwidth control on wide area routed networks.

In SIP terms, a message from a SIP server agent to a SIP user agent, usually in response to a method. See also method .

The nickname for in-band signaling on T1 circuits, which is rarely used for new voice applications today.

A trunk group that can receive many simultaneous calls destined for the same phone number. See also trunk group , hunt group .

The process wherein sound waveform amplitudes are quantified by digital or analog means; for example, graphing points on a sound wave. Sampling is the first step in pulse code modulation (PCM).

A protocol for encrypting RTP media payloads.

A simple mechanism used by SIP endpoints to communicate their capabilities during call setup.

An IETF protocol for enabling telephony, media delivery, and instant messaging applications on IP networks.

The sound of your own voice that you hear in your analog telephone.

The network that provides billing and calling signaling for the PSTN. Also known as Common Signaling System 7.

A method of conserving bandwidth by not transmitting any data during periods of silence in a VoIP phone call. Often accompanied by comfort noise generation.

A desktop softphone application for Windows and Macintosh that uses an Internet peer-to-peer network to facilitate calls between its subscribers.

An open -source security software designed to assist system administrators with intrusion prevention and detection on IP networks.

A server that performs call-routing functions, replacing the traditional legacy PBX or key system.

See softPBX.

Any distortion of a sound signal that causes loss of the intended signal's amplitude or strength.

The amount of cost savings reaped through a switch from legacy telephony to new VoIP-based systems. (Also, the cost savings reaped through any technology adoption project.)

A device that directs voice or data traffic at OSI layer 2. Ethernet can be switched technology, as can ATM. Also, a device that performs telephony call routing based on a dial-plan, such as a PBX.

The physical layer specification for optical carrier circuits that are used to connect CO switches and in very high-density data applications.

An ITU specification for carrying multiplexed data on a digital, four-wire circuit over medium distances (under about 5 miles). Repeaters can be used to extend the range of the circuit. See also E1 .

The hierarchy that describes digital circuits with DS0 channels, including T1s and T3s.

An IP network operator that provides voice calling services to its subscribers. Examples include Vonage and BroadVox Direct.

The method by which multiple voice signals are combined into a single bit stream for transport over a digital bus, such as the one inside a PBX or a T1 circuit.

A simple protocol commonly used by IP phones to obtain their operating config and firmware updates, which are files stored on a TFTP server.

Any physical or logical pathway between two VoIP networks or switches. An analog two-wire POTS line between a PBX and a CO switch is considered a trunk.

A group of trunks that reach a common logical destination, such as an incoming rollover group.

A spec for the format of SIP user aliases. Example: sip:ted@sip.oreilly.com.

A broad-purpose physical/data link interface technology that allows PC softphones to be used with special handsets to re-create the look and feel of a hard phone. USB is also commonly used for connecting printers, mice, keyboards, etc.

A way of securely transporting IP and non-IP packets between two private networks across the Internet.

Also known as Virtual LAN, a specification for partitioning broadcast domains on Ethernet segments and across WANs.

A telephony application that allows callers to record messages for recipients who aren't available to answer the call at the moment.

Technique for transporting encoded voice data on an ATM network without the overhead of TCP/IP encapsulated in the ATM cell.

A switch-to-switch trunk comprised of VoIP rather than a direct copper analog loop or other physical/data link layer trunk.

A network that uses routers or switches to connect a geographically diverse campus. The Internet is the world's largest WAN, of course.

A feature of Zaptel FXO interfaces that allows signaling of automatic removal instructions to telemarketers during incoming call attempts.

The TDM channel type Asterisk uses to interface with POTS lines and T1 circuits. In order to use these legacy channel types, Asterisk requires Zaptel driver software and PCI interface cards made by Digium.