The two-way transmission of voice over a packet-switched IP network, which is part of the TCP/IP protocol suite. The terms "IP telephony" and "voice over IP" (VoIP) are synonymous. However, the term VoIP is widely used for the actual services offered (see VoIP for more details), while IP telephony often refers to the technology behind it. In addition, IP telephony is an umbrella term for all real-time applications over IP, including voice over instant messaging (IM) and videoconferencing.|
Starting in the late 1990s, the Internet and its TCP/IP protocol suite began to turn the data communications and telephone industry upside down. IP has become the universal transport for almost all data and video communications worldwide. It is increasingly becoming the infrastructure for voice traffic as well. Today, every communications carrier has built or is using an IP backbone for some or all of its voice services. In addition, large enterprises are either already using IP for some amount of internal voice traffic or have plans to implement it or create test beds.
Data Over Voice Became Voice Over Data
Data was first transmitted over telephone networks, starting in the 1960s, and by the late 1980s, data routinely traveled over digital voice circuits. By the 1990s, the majority of worldwide communications traffic had changed from voice to data, and as IP networks began to flourish, the economics of using IP for voice began to emerge.
Although the backbone of the global telephone network had been converted to digital for some time, the circuit-switched nature of the public switched telephone network (PSTN) is wasteful. Even though one person talks and the other listens, both "to" and "from" channels are always dedicated. In addition, newer voice codecs cut the digital requirement from the traditional 64 Kbps (PCM) down to 8 Kbps with respectable quality. Thus, the bandwidth requirement for voice on an IP network is 1/16th that of the PSTN's dedicated, digital circuits.
Starting in the mid-1990s, advertiser-supported, free telephone service from PC to PC or between phones and PCs using the public Internet became popular, especially for international calls. Call quality over the Internet can be erratic because the Internet provides no guarantee of quality of service (QoS). However, when an organization has control over its network, quality can be excellent. Private enterprises with their own IP networks, as well as major telcos and IP telephony carriers that have developed IP backbones, can provide voice quality that competes with the traditional PSTN.
Transport and Signaling
IP telephony uses two protocols: one for transport and another for signaling. Transport is provided by UDP over IP for voice packets and either UDP or TCP over IP for signals. Signaling commands that establish and terminate the call as well as provide special features such as call forwarding, call waiting and conference calling are defined in a signaling protocol such as H.323, SIP, MGCP or MEGACO (see IP telephony signaling protocol).
The integration of packet-switched IP with the traditional SS7-based telephone system is a complex undertaking with numerous protocols competing for attention. See ITXC and IP on Everything.
This illustration shows the interaction between the traditional telco system and IP carriers, which are often one and the same. In order to understand the flow between these devices, it is important to note the difference between voice packets (blue lines) and signaling (red lines). (Illustration assistance courtesy of GNP Computers and Pulver.com.)
We can be certain that they didn't have the IP protocol in mind when they set up this telephone switchboard in 1882. It was used to switch phone calls between all the lawyers in Richmond, Virginia. (Image courtesy of AT&T.)