A CDE Definition
(Skinny Station Protocol) Cisco's proprietary implementation of the H.323 IP telephony model. Skinny phones can also be configured for the SIP protocol. See IP telephony.
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 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. See TCP/IP.
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 became the universal transport for data and video communications worldwide, and 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 do so.
Data Over Voice Became Voice Over Data
Starting in the 1960s, data was transmitted over analog telephone networks, 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 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 SIP, H.323, MGCP or MEGACO (see IP telephony signaling protocol).
The integration of packet-switched IP with the traditional SS7-based telephone system was a complex undertaking with numerous protocols competing for attention. See ITXC and IP on Everything.
Integrating IP Telephony with the PSTN
Not Quite IP Telephony
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Terms By Topic
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