Here we give a brief overview of the the most important VoIP or internet telephony related definitions, terms and acronyms. If you feel we should add further information please contact us.
Analogue Telephone Adapter. These devices allow you to use standard telephones for VoIP. It is basically a translater between what comes out of your analogue telephone and the digital voice over IP network.
Comfort Noise Generation. Many VoIP devices offer silence suppression - where rather than using bandwidth to transmit silence, a message is transmitted saying there is no sound. The device at the other end then uses comfort noise generation to create background noise (since when one talks on the phone, and it is silent at the other end, it is generally not absolutely quiet, there is some background noice). Thus, CNG is an attempt to make the silence seem more natural.
Direct Inward Dialling (also referred to as a 'virtual number'). This is a normal phone number on the PSTN (which can be anywhere in the world) that can be called by any phone worldwide, and forwards the call to your VoIP phone. Many VoIP service providers offer direct inward dialling as part of their service, or as an add-on. There are also third party companies that offer direct inward dialling, and will forward the call to your VoIP account no matter which service provider you are with.
Dual Tone Multi-Frequency. These are the tones produced by your touch dial telephone when dialling numbers (as opposed to the sequences of clicks that were used in the days of pulse dialling, and is still found on some very old telephones). Interactive telephone menus are able to work because of the fact that DTMF frequencies are standardised - a pair of frequencies is uniquely linked to a number (or # or *) on the telephone keypad. The 'dual tone' part of DTMF refers to the fact that each key-press produces two frequencies depending on the row and column of the key.
e164 offers a method of using your current telephone number as a means of being contacted in the IP world. Your telephone number gets mapped into a DNS zone, and the zone can contain your contact information (VoIP, instant messenger, email, anything). See e164.org for further details.
Free World Dialup. This is one of the most popular free SIP based VoIP services. It has been around for a few years, and is based on a community effort. See Free World Dialup more information about the service.
Foreign Exchange Office. The FXO port is what the cord in the back of your telephone comes out of (on the telephone side). Your phone provides an FXO interface to the telephone network.
Foreign Exchange Subscriber. Subscriber equipment (ie., telephones) plug into FXS ports. For example, the telephone port in the wall, is an FXS port.
A VoIP protocol that preceeded SIP. Unlike SIP, the H.323 standard specifies the complete voice over IP protocol, and not just the signalling methods.
Integrated Access Device. All-in-one VoIP adapters that also include a router and possibly an ADSL modem are known as IADs.
Network Address Translator. Defined in RFC 1631, network address translation allows one public IP address to be shared between a number of devices. This is done by the device with the public address, acting as a gateway between an Internal network (running on private IP addresses - RFC 1918) and making all requests coming from the Internal network appear as though they are coming from the gateway device itself. It is commonly used in home networks.
Private Automated Branch Exchange. A PABX is as sophisticated piece of equipment which connects an office to the telecommunications network and allows many workers to share only a few telephone lines. Advanced features such as voicemail, hold, transfer, least cost routing and more are also provided by PABXs. A PABX without the sopihisticated features is known as a PBX - a private branch exchange.
Plain Old Telephone System. The name says it all.
Public Switched Telephone Network. The standard telephone network used today.
Quality of Service. Exactly like it says. It is a way to mark some Internet traffic as being more urgent. For example, one might mark voice traffic as being "important to get there in real time, but if there is a huge delay, just drop the packets", where as downloading emails could have a QoS saying "just get the data when there is unused space on the line to my house". thus if you use both applications, whenever you use VoIP, it will be guaranteed a higher priority than emails. QoS is not implemented on an Internet-wide basis, though there is a lot of hardware that can deal with QoS parameters.
Ringer Equivalence Number. This is a number which defines how much power can be put out be an FXS interface. Every phone/answering machine/modem/fax machine/etc also has a REN which defines how much power it draws through the phone line (eg., to make the phone ring). For example, the REN of the phone socket in the wall, might be 5, and if you have 3 telephones attached, each with a REN of 1, then everything will work. However, if each of those phones have a REN of 2 (making the total REN of the phones 6), this will exceed the REN of your phone line and your phones will no longer work properly due to lack of power.
Session Initiation Protocol. A protocol allowing a user to 'call' another user on their Internet phone. Think of SIP as a way to call someone, rather than 'dialing' their IP address, you call a phone number which the SIP proxy (the service provider) uses to identify them, and forward them an 'invite' message to the conversation.
A sip line is a protocol used for establishing a communication pathway on an IP network. A SIP line supports communication as simple as a two-way telephone call or a multi-media rich exchange including web video conference, voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists or an online video game. SIP has become the protocol of choice for signaling communication via VoIP notably when VoIP is used on a mobile phone. SIP’s simple protocol and method of establishing and terminating communication over an IP creates scalability, extendibility, and reliability in multiple coding languages.
SIP is a request and response protocol that is similar to the other main Internet protocols, HTTP and SMTP (the protocols that power the World Wide Web and email); consequently, SIP plays well with other Internet applications. SIP is a simple method that service providers can use to build converged voice and multimedia services.
Simple Traversal of UDP through NATs. STUN allows SIP devices behind NATs to discover their public IP addresses (that of the gateway), and the type of NAT in use. This is one step in creating a system whereby the VoIP device can be contacted by other Internet telephones.
Voice activity detection. This feature allows VoIP clients to detect when the person is speaking versus background noise. If the VAD algorithm is not sophisticated enough, and silence suppression is enabled, then some of your voice might get cut off during a conversation.
Voice over Internet Protocol. Used to refer to all types of Internet Telephony. It encompasses just about any system that carries voice traffic over the Internet. For home users, it provides the ability to talk to other VoIP users without having to pay standard telephone usage charges (although you still have to pay for your Internet connection!).
SIP and H.323 are the two main standards for VoIP, although SIP is becoming the most popular due to its generic nature, and the fact that unlike H.323, it was designed specifically for wide area Internet Telephony. There are also some custom protocols such as that used by Skype.