Voiceover Internet Protocal (VoIP): Evaluation of Technologies for Deployment

Abstract

Traditionally, real-time voice communications–both within and outside of corporations (enterprises)—are achieved using public domain circuit-switched telephone networks (PSTNs). These networks use technologies that have been maturing over the last 150 years. However, the recent advances in and proliferation of packet-based switching technologies, World Wide Web (WWW) and PC-based user interfaces, and innovative digital signal processing (DSP) techniques are making real-time voice transmission using packet switching–particularly IP-based techniques—more feasible, at least within the logical boundaries of enterprise networks. Although the IP-based Internet is only about 25 years old, its compatibility with Ethernet2 , and its flexibility, openness and low-cost availability have enabled it to gain more than 260 million users worldwide3. In addition, proliferation of the WWW, multimedia PCs, and innovations in DSP during the last five years has made voice transmission over packet networks, particularly over IP, very attractive economically. In this chapter, we briefly review the technologies and standards—as recommended by IETF and ITU-T—that are making voice over IP (VoIP) a reality in both public and enterprise networks. Our focus is on low bit rate speech compression and silence suppression, voice packetization and encapsulation, sources of packet-voice impairments and methods to mitigate them, and packet-voice transmission engineering.