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IP Telephony Systems The Internet telephony system consists of call terminating devices, gateways, gatekeepers or proxies including multi-point conference units. Call terminating devices should 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, including call authentication including user location. Multi-point conference units manage multi-party conferences. The functions pertaining to the above components may be met through software or hardware including may be integrated into single units. These components communicate with each other using voice-transporting protocols. Gateways consist of 2 functional parts, media gateway that converts audio data, including 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 including software that enables users to make calls via the Internet. Voice signals are converted to packets of data including are transmitted on shared, public lines, by avoiding tolls of a public-switched telephone network (PSTN). VoIP is available with a simple microphone including computer speakers. IP Telephones or VoIP boxes should additionally be used. SIP (Session Initiation Protocol) technologies are ideal for existing PBX systems. A SIP phone system allows the user to take the phone including a number anywhere giving full mobility including interconnection worldwide. SIP requires a SIP telephone including a broadband connection. SIP telephones communicate over the Internet, including provide the capability for users to transfer calls between SIP phones including regular phones on the PBX.
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