VoIP Telephony : a Compact Glossary

This is a basic glossary of VoIP terms.

ACD

Automatic Call Distributor - A system for automatically routing an incoming call to the next available agent's call position.

Asterisk

Named after the * character, Asterisk is a richly featured PC-based telephone exchange system which can support internet telephones and through the use of additional hardware will connect to traditional analogue and ISDN telephone lines. The system is Open Source and is available under the GPL license.

ATA

Analog Telephone Adapter - A small device that can be used to connect a traditional analogue phone to an internet connection, typically through an RJ45 socket. More sophisticated ATA's like the Linksys 2102 will connect to a traditional phone line and will also allow 2 analogue phones to have simultaneous connections to SIP accounts over a broadband connection

CentOS

CentOS Linux is an enterprise-class linux server distribution derived from sources freely provided by Red Hat. See here for more information. CentOS (Community ENTerprise Operating System) changes some packages to remove branding and artwork. It is Free Software. CentOS is used by the Trixbox package.

channel

In the context of Voice over IP, a channel is the virtual pathway by which a voice transmission takes place across a network. Note that the term channel may be used in other ways, for example to refer to a signalling pathway, or to include video content.

codec

Compressor/Decompressor - A piece of software that compresses speech data. Examples of commonly encountered codecs are G.711 alaw, which is used by European PTT's, G.711 ulaw used in the US, G.723.1, G.726, G.729, GSM, iLBC, Speex.

dial plan

The Asterisk dial plan controls how all incoming and outgoing calls are handled and routed.

FreePBX

FreePBX, formerly known as AMP, the Asterisk Management Portal, is a graphical user tool which can be used to configure Asterisk as an alternative to directly editing text configuration files.

IAX

Inter-Asterisk eXchange protocol - Now in its second version, and sometimes referred to as IAX2, this protocol is used to set up and maintain VoIP connections between Asterisk servers. Digium produces a small adapter referred to as an IAXy which can interface an analogue phone or fax to an Asterisk server.

H323

H323 protocol - A signalling protocol intended for telephony and video. Unlike SIP it supports interfacing to traditional telephony systems using a gateway.

Information Provider

Typically a VOIP information provider will provide a server which will enable subscribers using an internet connection to register their SIP phones and then call other subscribers. This service is usually free. For an additional charge providers will supply ordinary PSTN telephone numbers to subscribers and allow them to connect to POTS (Plain Old Telephony System) customers throughout the world. The better VOIP information providers also allow VOIP subscribers to talk to others on different networks. This may be through forwarding using the fully qualified domain name, referring to the enum directory or using a SIP Broker.

NAT

NAT - Network Address Translation. Usually an individual user or organisation has one or a few IP addresses made available to them. In order to have a larger number of devices connected to the internet, a separate private addressing scheme is used for local addresses and a conversion process is carried out by a device called a router. This process is known as network address translation.

NAT can give rise to problems with VOIP channels using the SIP protocol, and there are a number of solutions involving phone settings or the use of a STUN server.

PBX

PBX - Private Branch Exchange. A privately owned system for voice switching and other telephone services. The system routes calls from the public telephone system to phones within an organisation and of course allows internal calls to be routed as well.

POE

POE - Power over Ethernet. An arrangement by which spare wires in Ethernet cables are used to provide a voltage to a device. This is an excellent arrangement for IP telephones which support POE as the power adapter is eliminated. A hub, switch or router with built-in POE is needed. Note that Cisco devices use non-standard wiring!!

POTS

POTS - Plain Old Telephone Service. The traditional telephone service which uses analogue voice signalling from the phone to the local exchange.

PTT

PTT - Post, Telephone and Telegraph authority. Historically British Telecom was the monopoly provider in the UK.

QOS

QOS - Quality of Service The quality of a voice call over a VoIP network. Latency, packet loss, network jitter, echo and other factors may contribute to the quality of a call. There are quality of service solutions offered by some providers that reserve bandwidth across the network.

rtp

RTP - Real Time Protocol. An IP protocol for the transmission of data packets of that require low latency like voice and video data. Asterisk uses ports 10000 to 20000 for its rtp traffic.

SIP

SIP stands for Sessions Initiation Protocol, which is a jolly good way to do Voice over IP.

SIP is a set of open standards managed by the IETF. Because of the open standards approach SIP devices are generally interoperable between different vendors. In a SIP-based VoIP (voice over IP) system, most of the intelligence is in the phones rather than centrally in the PABX.

This results in a much more scalable system. Installing additional phones only requires there to be a suitable network connection and, apart from licensing issues, the system cannot become "full" unlike a PABX.

SIP addresses have the same format as email addresses and in future they may become the same thing - just one address for email, mobile, home and work phones. Your SIP address could provide one telephony identity where ever you are.

Softphone

Softphone - A software application which you can use as an IP phone. An excellent tool for testing and for Road Warriors as well. They work best when used with a USB headset. Our favourite is xten-lite which you can download from: http://www.counterpath.com/index.php?menu=download

SugarCRM

SugarCRM is a customer relationship management system which can be integrated with Asterisk. The Trixbox distribution includes SugarCRM. For more information see here

Trixbox

Trixbox, in earlier versions known as Asterisk@Home, provides an automated installation of Asterisk, SugarCRM, FreePBX and the Hudlite operator panel. Although there is additional configuration work to do, Trixbox is an excellent starting point for a VOIP PBX installation.

Trunk

A trunk is a pathway between two switches. To connect your VOIP PBX to the outside world you might use an analogue line, an ADSL connection, an ISDN BRI or broadband ISDN or ATM line.