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Speech Signal Processing
Published in Richard C. Dorf, Circuits, Signals, and Speech and Image Processing, 2018
Jerry D. Gibson, Bo Wei, Hui Dong, Yariv Ephraim, Israel Cohen, Jesse W. Fussell, Lynn D. Wilcox, Marcia A. Bush
The most familiar form of network for voice communications is the PSTN, which consists of a wired, time division multiplexed (TDM), circuit-switched backbone network with (often) copper wire pair local loops [6]. The PSTN was designed and evolved with voice communications as a primary application. The voice codec most often used in the PSTN is 64 kbits/sec logarithmic pulse code modulation (log-PCM) designated by the ITU-T as G.711, and which is taken as the standard for toll-quality voice transmission. The TDM links in the PSTN are very reliable with bit error rates (BERs) of 10−6 to 10−9. As a result, bit error concealment is not an issue for G.711 transmission over TDM PSTN links, even though bit errors in G.711 encoded voice generate very irritating “pops” in reconstructed speech. Furthermore, G.711 is designed with several asynchronous tandems in mind, since it was possible to encounter several analog switches during a long-distance telephone call prior to the mid-1980s. Even eight asynchronous tandems of G.711 with itself has been shown to still maintain a MOS greater than 4.0 when a single encoding is 4.4 to 4.5.
Legacy to Current-Day Telephone Networks
Published in Stephan S. Jones, Ronald J. Kovac, Frank M. Groom, Introduction to COMMUNICATIONS TECHNOLOGIES, 2015
Stephan S. Jones, Ronald J. Kovac, Frank M. Groom
The PSTN has been a part of our daily existence for over 100 years. The historical significance of Alexander Graham Bell’s invention, or possibly his ability to patent his idea only moments before his competitor Elisha Gray, has been the cornerstone of communications technology from the late 1800s through the 1990s. The PSTN is based on the creation of a direct connection from the sender to the receiver over an extremely sophisticated delivery network comprised of copper, fiber optics, satellites, and fixed wireless and mobile wireless circuits. The network is comprised of five basic components (with numerous subsets): the telephone, network access, COs, trunks and special circuits, and CPE. The purpose of this section is to provide a firm understanding of the basic components of the PSTN.
Voice Transmission
Published in Goff Hill, The Cable and Telecommunications Professionals' Reference, 2012
Stuart D. Walker, Rouzbeh Razavi
This section describes the relevant details of current standardized voice codecs in the context of the network for which they were designed. Most speech coding systems in use today are based on telephone bandwidth narrowband speech, nominally limited to about 300 to 400 Hz and sampled at a rate of 8 kHz. Public Switched Telephone Network (PSTN) is the most familiar network for conversational, circuit-switched, Time Division Multiplexed (TDM) voice communications. The primary voice codec used for PSTN is a 64-kbit/s, log-PCM codec standardized by ITU-T as G.711 in 1988. Because of the very low bit error rate performance of PSTN circuits, no error concealment scheme is provided by G.711 when it is used for PSTN. Examples of other voice codecs standardized over the years by ITU-T for PSTN include G.726 and G.727, both of which were standardized in 1990 and use ADPCM technique. G.726 covers the transmission of voice at rates of 16, 24, 32, and 40 kbit/s, but the most commonly used mode is 32 kbit/s on international trunks in phone networks.
Design and implementation of a VoIP PBX integrated Vietnamese virtual assistant: a case study
Published in Journal of Information and Telecommunication, 2023
Hai Son Hoang, Anh Khoa Tran, Thanh Phong Doan, Huu Khoa Tran, Ngoc Minh Duc Dang, Hoang Nam Nguyen
Voice-over-Internet Protocol (VoIP) comprises a set of software and hardware technologies for making voice calls that use a data network instead of a traditional Public Switched Telephone Network (PSTN) system. VoIP is widely used in corporate environments, and the adoption of this technology by businesses is expected to continue growing in the coming years (Packer & Reuschel, 2018). The main reason for the popularity of this model is cost saving. Both large and small companies acknowledge that deploying and managing separate data and voice networks is expensive. In contrast, converged voice and data networks enable unified communications services while reducing costs. Furthermore, the costs associated with traditional phone calls are usually higher than those associated with VoIP calls (Karapantazis & Pavlidou, 2009).