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.
ATA
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. See our VoIP hardware section for more information.
CNG
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.
DID
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.
DTMF
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
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.
FWD
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.
FXO
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.
FXS
Foreign Exchange Subscriber. Subscriber equipment (ie., telephones)
plug into FXS ports. For example, the telephone port in the wall, is an
FXS port.
H.323
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.
IAD
Integrated Access Device. All-in-one VoIP adapters that also
include a router and possibly an ADSL modem are known as IADs.
NAT
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.
PABX
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.
POTS
Plain Old Telephone System.
The name says it all.
PSTN
Public Switched Telephone Network. The standard telephone network
used today.
QoS
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.
REN
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.
SIP
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. You can
find out more about SIP from sipcenter or http://www.voip-info.org/wiki-SIP.
STUN
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.
VAD
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.
VoIP
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.
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