In early 1995 VocalTec* introduced the first IP telephony software product.
IP telephony allows data, voice, and video to be transmitted over a single network infrastructure, through LANs, WANs and the Internet.
The Internet and IP-based networks are increasingly being used as alternatives to the public switched telephone network. By bridging the traditional circuit-switched telephony world with the Internet, gateways offer the advantages of IP telephony to the standard telephone.
IP telephone gateways make the connection possible: the gateway connects to the telephone world on one side and to the IP data world on the other side. A gateway can communicate with any standard telephone in the world to 'translate' the audio signals. The gateway takes the standard telephone signal, digitizes it (if it is not already digital), compresses it, packetizes it using the Internet Protocol (IP) and routes it to any remote destination over the LAN, WAN or the Internet. The gateway also handles the addressing of the users. Addressing a remote computer, its IP address must be known, addressing a remote phone connected to a gateway, only the regular telephone number must be known.
The conversion from and to the network take place in the same time, allowing a full-duplex conversation. Phone-to-PC or PC-to-phone operation can take place with one gateway. Phone-to-phone operation needs the communication of two gateways.
The downside? Delay of the human voice affects the conversation. Humans can tolerate about 250 msec of latency before it has a noticeable effect. But today's IP telephony products connected over the Internet mostly sound like traditional calls routed over a satellite circuit. The IP telephony gateways must supply echo cancellation.
H.323, the standard for videoconferencing over IP/Ethernet, is clearly emerging as the standard 'call control' protocol for IP telephony. It defines packet standards for terminal, equipment and services for multimedia communications over large area networks (LANs), communicating to systems connected to telephony networks such as ISDN.
In real-time transmission, up to 30% of the packets in a transaction might be lost or delayed, which is the same as lost in real time. IP telephony applications need to recover from lost packets by effectively reconstructing the lost data. G.723 is a new algorithm standard for compressed digital audio over telephone lines.
The typical telephone signal, the DTMF tone (dialing tone) cannot successfully be transmitted over the IP telephony. These tones are detected on the gateway, transmitted encoded and newly generated on the remote side.