VoIP Glossary

This VoIP glossary provides an overview including a short definition of the most common terms used in VoIP.

Select letter: A | B | C | D | E | F | G | H | I | L | M | N | P | Q | R | S | T | U | V | W |


  • ACDAutomatic Call Distribution – is the automatic routing of calls based on specific characteristics or selection criteria chosen by the caller. ACD is used in conjunction with IVR (Interactive Voice Response) as well as fully automated voice applications.
  • ALG – Application Layer Gateway – is a software component for managing special application protocols, such as SIP (Session Initiation Protocol) and FTP (File Transfer Protocol).
  • API – Application Program Interface – is an interface that allows two computer applications to communicate with each other.
  • Appliance describes all hardware on which telephony software can be installed and operated.
  • ATA Analogue Telephone Adapter – allows the connection of analogue devices (telephones, fax machines etc.) to the IP telephone system.


  • BLF – Busy Lamp Field – shows the line condition for other users on the same PBX. These buttons are often used as speed dial on desk phones.


  • Call Center is a group of agents who handle a large volume of requests via phone. Inbound Call Centers mainly deal with incoming product or service information requests and customer support. Outbound Call Centers mainly deal with telemarketing, market research or outbound sales calling campaigns. A Contact Center also handles individual communication using live chat, email, web meetings and live support. The main tools of call & contact centers are call queues and website chat and talk plugins.
  • Call Flows are voice applications which automatically forward calls according to a defined business logic, e.g. assignment of a call to the responsible administrator using the Caller ID -> Database query. See also “ACD”.
  • Codec refers to an encoder / decoder that digitally encodes and decodes data or signals to send over a data network. In VoIP telephony, these play a particularly important role, as they determine how and in what quality audio data and video data are transported within a network. Supported codecs include opus, speex, G.711 (PCMU/A) G.722, G.729, GSM, iLBC, iSAC, L16 for audio and VP8, OpenH264 for audio and video.
  • CTI – Computer Telephony Integration – Softphones installed on a computer typically use CTI to control existing desk phones, such as entering the phone number via the computer and dialling automatically via the desk phone. In this context, CTI is also used in the integration of third-party application software, such as CRM and other management systems. By clicking on the number of a contact, the call is initiated via the desk phone. For incoming calls, the CRM contact is displayed. This feature is also often referred to as Click2Call / Click to Call.


  • DECTEnhanced Cordless Telecommunication – is a standard for wireless telephony that is commonly used in combination with VoIP.
  • DID – Direct Inward Dialling – is also known as DDI (Direct Dial-In) and allows direct dialling to a VoIP user from a PSTN line. Telecoms providers provide these phone numbers or number blocks as part of their SIP Trunk. KCM Telecom provides DIDs / phone numbers for more than 60 countries.
  • DNS – Domain Name Server – assigns each name / domain a unique IP address.
  • DTMF – Dual Tone Multi Frequency – also frequency dialling, is used for signalling between telephony devices and switching centers in the voice signalling band. Every digit is represented by a dual tone sent in the voice band. In times of VoIP and using compressed codecs (which can cause problems in the transfer of dual tones) in 2000 an RFC was introduced (now RFC4733, original RFC2833) by the IETF regarding DTMF RTP Payload. The suggested solution is that in VoIP dual tones get transferred in the RTP band, not in the voice band anymore. So, the transfer of dual tones is not influenced by the used voice codec anymore. DTMF is also referred to as dial tones and are mandatory for using voice menus / IVRs.
  • DTLS –  Datagram Transport Layer Security – is a protocol for privacy protection in electronic communications based on TLS (Transport Layer Security).


  • Echo cancellation is a hardware and software-usable algorithm with an echo suppression / reduction concept – a voice processing system analyses the incoming voice stream and controls the outgoing voice stream prior to its transmission. Should the system encounter identical voice streams, they will be “subtracted” from the outgoing voice stream and thus removed before transmission.
  • ENUM – E.164 Number Mapping – refers to a translation of traditional telephone numbers in the Internet addresses of VoIP subscribers (VoIP URI).
  • Extension is a user account in the PBX / Phone System. Each user receives an internal extension number under which he can be reached within the company. This extension can be connected to an external number, so that the extension can also be reached from the outside (DID / DDI). User extensions must be configured for use with at least one phone. These phones can be desk phones, softphones, web clients or smartphone clients. Door intercoms and other systems are also connected to an extension. In addition, there are also system extensions for IVR menus, call queues, ring groups and more.


  • Failover refers to a network of at least two computers (clusters) in which a second computer takes over its tasks if one computer fails. This significantly increases availability in telecommunications (high availability). A step further is to mirror the servers in physically different data centers (disaster recovery). Failover is part of KCM Telecom’s dedicated hosting in our Vienna Data Centers, so your data is secure with us! 
  • Fast Path is the name for a transmission type that shortens the latency of contacting servers. It shortens the times required for the transfer of a data packet between your own provider and your own modem / router.
  • FoIP – Fax over IP – describes the use of fax technology over IP networks / Internet. Transmission protocols are T.38 or fax via G.711.
  • FQDNFully Qualified Domain Name – is the full name used to identify a computer or host on the Internet. It specifies all domain levels, at least one host name and one domain name. An FQDN could be mypbx.3cx.com. “mypbx” is the host, “3cx.com” is the domain in which the host is located.
  • FXO – Foreign Exchange Office – refers to an analogue interface for telephones or telephone systems. It is the receiver interface of an analogue signal.
  • FXS – Foreign Exchange Station – describes an analogue interface for end points such as phones and fax. Among other things, this is the telephone socket in the wall. FXS interfaces can also be integrated in VoIP gateways. FXS sends the analogue signals.


  • Gateway is a device that provides interfaces between traditional telephony (analogue, ISDN) and VoIP. FXS and FXO ports connect traditional telephone lines and devices to the IP network. The gateway translates the signals in both directions.


  • Hosted PBX is a data center / cloud hosted PBX. The PBX is not installed on premise. On premise desk and soft phones are usually connected to the remote PBX via a session border controller so that they can be used and managed locally. KCM Telecom offers physical VPN gateways connected to our data center that act as Session Border Controller, so you can connect Desk Phones on premise to the PBX in the Cloud.
  • Hot-desking refers to a model of office space management that allows employees to share workspaces. Employees log on to the desk phone with their data, the phone is then automatically configured to the extension of the employee until logout.


  • IaaS – Infrastructure as a Service – is one of three service models of cloud computing. It provides the user with all the components of a data center infrastructure such as hardware, computing power, storage space or network resources from the cloud. KCM Telecom’s hosting is an example for IaaS. The other two service models are PaaS (Platform as a Service) and SaaS (Software as a Service).
  • IP address is used for unique addressing for routing data in IP networks. IPv4 addresses consist of four bytes (numbers from 0 to 255). Due to the limited number of available public IP addresses based on this IPv4 notation, further IPv6 addresses are introduced which allow a much larger number of IP addresses.
  • IP phones, SIP phones, VoIP phones or Desk Phones are SIP-enabled phones, which are used as soft phones or hard phones in the context of IP telephony. Hard or Desk Phones require a VPN Gateway / Session Border Controller to be connected to our Cloud PBX. KCM Telecom is offering this hardware.
  • ISP – Internet Service Provider – is a provider that allows users and businesses to access the Internet. Frequently, other services such as telephony, TV and mobile services are offered.
  • IVR / Digital Receptionist – Interactive Voice Response – is a computer-controlled voice dialogue system used for interactive announcement services. Callers receive navigation options via a digital voice menu in order to call up services by voice or input on the telephone keypad (DTMF).


  • LCR Least Cost Routing is a concept which routes outgoing calls in such a way as to incur minimal costs. KCM Telecom partners with leading providers for call termination globally in order to minimise costs for airtime.


  • MAC address is a specific address for identifying a physical network-enabled device. Each network hardware has a unique MAC address. If a system sends a data packet to another in the same local network, the corresponding MAC address is first searched for the destination IP address.
  • MTU / RWIN – IP Maximum Transmission Unit – The MTU of a network interface indicates how large the largest packet that can still be transmitted without fragmentation. The RWIN parameter is connected to the MTU. RWIN determines the maximum amount of data that a system sends over a TCP connection without the need for receipt acknowledgment by the remote station.


  • NAT – Network Address Translation – means grouping individual devices into local networks behind a common public IP address. Each internal device is assigned an IP of the format 192.168.xxx.xxx. This counteracts the IPv4 address shortage. Another solution to the shortage is IPv6.
  • NTP – Network Time Service – NTP is a UDP-based protocol on port 123 that is used to synchronize the clock on different systems on the Internet.


  • PaaS – Platform as a Service – is one of three service models in cloud computing. A development environment from the cloud that helps users develop or test their own software applications. Examples are MS Azure or Google Cloud. The other two service models are IaaS (Infrastructure as a Service) and SaaS (Software as a Service).
  • PBX / PABX – Private Branch Exchange / Private Automatic Branch Exchange – is an expression for a telephone system that can be used to connect end points such as phones, fax and answering machines with each other as well as with the public telephone network.
  • Provisioning allows easy and centralized configuration of IP phones and clients. With auto-provisioning phones receive their information directly from the PBX. Data such as manufacturer, extension, caller ID and MAC address are stored in the PBX. This saves the manual setup of each individual phone. KCM Telecom offers even more methods to configure your phones and clients like QR Code Provisioning or provisioning via email attachment.
  • PSTN – public switched telephone network – is the circuit-switched, global public telephone network (landline) for handling proprietary telephony. Similarly, IP-based communication is handled by data packets via the Internet.


  • QoS – Quality of Service – is a method that enables a stable VoIP connection by prioritizing IP packets. SIP packets are prioritized over other data packets in order to guarantee the best possible voice quality.
  • Queue / Call Queue is a feature that groups extensions that act as agents. Calls to this queue are forwarded to these agents according to various ring strategies. When all agents are busy, additional calls are held on hold until the next agent is available again. Queues are the main features of Call Centers.


  • RESTful API is a new programming interface for integrating various systems, such as IP PBX with a web-based CRM. This interface uses simple HTTP requests such as GET, PUT, POST and DELETE. This interface is especially used in the development of web services.
  • RFC – Request for Comments – is a series of consecutively numbered technical and organizational documents that formulate Internet standards. These requests are non-binding but are established through general acceptance and use in commercial software and freeware on the Internet and in UNIX communities.
  • RTP Real-Time Transport Protocol – is a protocol for the transmission of audio and video data, which is used in IP telephony and web-based video conferencing. RTP provides for the continuous transmission of multimedia data, whereas RTCP (Real-time Transport Control Protocol) transmits control messages between sender and receiver. SRTP (Secure Real-Time Transport Protocol) has the advantage of additional authentication and secure transmission. ZRTP is the key exchange protocol for VoIP end points to agree keys for encrypting voice or video streams using SRTP. 
  • Ring Group is a group of extensions that shares incoming calls. Based on defined ring strategies, e.g. “Simultaneous”, a call is routed to all group members, all phones start ringing until a group member answers the call. Unlike call queues, incoming calls are not held in a queue.


  • SaaS – Software as a Service – is one of three service models of cloud computing. It’s software that starts directly from the cloud without necessarily requiring a local installation. An example is Office 365 or KCM Call & Meet / KCM Cloud PBX. The other two service models are IaaS (Infrastructure as a Service) and PaaS (Platform as a Service).
  • SBC – Session Border Controller is a piece of hardware or software, that securely connects two separate networks to communicate with each other. The connection is usually established via VPN. KCM Telecom offers VPN Gateways to connect local desk phones with KCM Cloud PBX in our data center. The Session Border Controller manages traffic between these two networks. If one desk phone calls another one (colleague to colleague), then the voice traffic stays local, however on external calls, the voice traffic is passed on to KCM Cloud PBX. This helps reducing external traffic significantly and keeps admin firewall configuration to a minimum.
  • SDP – Session Description Protocol – is a format for describing multimedia sessions, which contributes to the negotiation of streaming media parameters (such as codecs and transport protocols).
  • SIP – Session Initiation Protocol – is a protocol developed by the IETF MMUSIC Working Group that is used to set up, hold and terminate communication sessions. These sessions may include, but are not limited to, telephone conversations including multimedia streaming, instant messaging, virtual reality scenarios, and playing online computer games.
  • SIP Client / Softphone is a software phone that can be used as replacement for hardware phones. Calls can be made and answered, conferences can be started, or presence state can be set for an extension.
  • SIP Forking guarantees accessibility on different IP phones with the same extension number.
  • SIP responses, also called SIP status codes, indicate possible answers to a SIP request. E.g. “200 OK” indicates the request was successful.
  • SIP server, also known as SIP proxy, is the most important component of an IP PBX. It is responsible for initiating, controlling and ending a conversation. The actual reception and transmission of audio data is handled by a media server via RTP.
  • SIP trunk is generally the telephone connection of your VoIP provider to connect your PBX to the public network via the internet. KCM Telecom provides this connection and DIDs out of the box with KCM Call & Meet and optional with KCM Cloud PBX.
  • SIP URIs are used to address subscribers to SIP-based calls. The SIP protocol uses URIs (Uniform Resource Identifiers) that contain a user ID and a domain. It is thus the SIP telephone number of a call partner. SIP URIs use notation familiar from e-mail addresses.
  • Skill-based routing describes a signalling method for call queues in which calls are routed to the right agents based on defined capabilities or knowledge. This method is very popular with call centers and support hotlines.
  • Smartphone Client is a software for Android and iOS, which ensures the communication between smartphone and IP PBX via an Internet connection. The user can then be reached on his smartphone at the company number or can initiate calls via the corporate telephone system from anywhere (One Number Concept).
  • Softphone is software that can be installed on any IP-enabled device, providing the functionality of a telephone without the need for a physical hard phone.
  • Speech-to-text is a service that uses speech recognition to transcript speech into text. This service is utilised for the transcription of voicemail messages.
  • SSL – Secure Sockets Layer – is the standard protocol for establishing encrypted connections between web server and browser. The use is recognizable in websites with “https:” instead of “http:”.
  • STUN server – Simple Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs] – allows NAT clients (such as computers behind a firewall) to communicate with a VoIP provider outside the local network.


  • T.38 is a protocol that allows you to send faxes over IP networks by converting fax tones to digital data. Another method is fax over G.711, in which the fax data is transmitted as audio data via RTP.
  • TAPI – Telephony Application Programming Interface – is a programming interface for telephony applications. Applications for TAPI include, for example, software telephony, video conferencing or call centers. Third-party software can be connected to these systems via TAPI, such as CRM programs (see also “CTI”). It is an obsolete interface technology, which is gradually replaced by scripting interfaces and new cloud-based interfaces such as RESTful API.
  • TCP – Transmission Control Protocol – is an extension of the Internet Protocol (IP) for transmitting data from host to host. TCP provides error control and correction and can break up and transfer large volumes of data into smaller data packets. The UDP protocol can’t do that. Therefore, TCP is preferred in Internet traffic.
  • TLS – Transport Layer Security – is a protocol that ensures security and privacy between connected systems. It is the most widely used protocol and is used in browsers and other applications that require the secure exchange of data over a network, such as VPNs, instant messaging or VoIP. An advanced form of this protocol is DTLS (Datagram Transport Layer Security).
  • TTS – Text-to-Speech – refers to the computer-generated conversion of texts into speech and is used in the context of interactive voice menus (IVR).


  • UC / UCC – Unified Communications / UC & Collaboration – refers to the integration of all communication channels, devices and media with the goal that everyone can interact with everyone in real time regardless of location. This connection of different communication technologies into a single environment includes, for example, e-mail, telephony (both landline and softphone), voicemail, conferencing and presence indication. UCC adds more components to UC for collaboration, such as document sharing & editing, video conferencing, whiteboards, etc. KCM offers all UCC features with their services – UCaaS.
  • UDP – User Datagram Protocol – is an alternative transmission protocol to TCP. Both rely on the Internet Protocol (IP). UDP provides port numbers to differentiate different user requests and control mechanisms to ensure that the data has arrived. UDP is mainly used for the transmission of voice and audio data. While TCP sends lost data packets again, the UDP does not. When transmitting voice, the brain takes over filling in gaps.


  • Voicemail is an extension of the answering machine that allows voice messages to be received, monitored and managed. These voice messages can also be delivered as e-mail and transcribed from Speech to Text.
  • VoIP – Voice over Internet Protocol – is also known as IP Telephony, Internet Telephony or Digital Telephony. VoIP allows you to make calls over the Internet or other computer networks based on the Internet Protocol.
  • VP8 – Video Compression Format or Video Compression Specification – is a free video codec used to encode or decode HD video for video telephony or video conferences.
  • VPN Virtual Private Network – is a connection between two network participants, which is secured by a certificate and enables encrypted data traffic. KCM Telecom offers VPN Gateways to connect local desk phones with KCM Cloud PBX. This is also often referred to as Session Border Controller (SBC).


  • WebRTC – Web Real-Time Communication – refers to an open standard for audio, video and data communication directly through the web browser. The browsers Chrome and Firefox already support this standard. Examples for applications are web and video conferences or video calls like KCM Call & Meet or KCM Cloud PBX.

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