The Internet telephony system consists of call terminating devices, gateways, gatekeepers or proxies and multi-point conference units. Call terminating devices can be traditional telephones or audio-equipped computers. Gateways are used for transmission conversion of calls into a format transportable over the Internet. Gatekeepers provide centralized call management functions such as call admission, bandwidth management, address translation, and call authentication and user location. Multi-point conference units manage multi-party conferences. The functions of the above components may be met through software or hardware and may be integrated into single units. These components communicate with each other using voice-transporting protocols.

Gateways consist of two functional parts, media gateway that converts audio data, and a media gateway controller that communicates using gateway control protocol (GCP). IP Telephony uses either VoIP or SIP technologies for voice communication. VoIP is a combination of hardware and software that enables users to make calls via the Internet. Voice signals are converted to packets of data and are transmitted on shared, public lines, by avoiding tolls of a public-switched telephone network (PSTN).

VoIP is available with a simple microphone and computer speakers. IP Telephones or VoIP boxes can also be used. SIP (Session Initiation Protocol) technologies are ideal for existing PBX systems. A SIP phone system allows the user to take the phone and a number anywhere giving full mobility and interconnection worldwide. SIP requires a SIP telephone and a broadband connection. SIP telephones communicate over the Internet, and provide the capability for users to transfer calls between SIP phones and regular phones on the PBX.