International Journal of Networks and Communications 2012, 2(2): 7-12 DOI: 10.5923/j.ijnc.20120202.02 Voice Quality Evaluation of a Call Using Fuzzy Logic Oyetade Durojaiye * , Elizabeth. N. Onwuka Telecommunications Engineering Department, Federal University of Technology (FUT) Minna, Niger State, Nigeria Abstract Voice will remain a fundamental communication media that cut across people of all walks of life. It is there- fore important to make it very affordable. VoIP has been increasingly popular in recent times due to its affordability, how- ever, poor reliability and voice quality remain important factors that limit the widespread adoption of VoIP systems. Good voice quality is a key factor for users transiting from the Public Switched Telephone Network (PSTN) to VoIP networks. It has been shown that line echo is one of the key factors that deteriorate voice quality in VoIP systems. Several non-real-time algorithms have been developed in literature to estimate various aspects of voice quality in VoIP systems. But there is no real-time algorithm that estimates the echo content of a VoIP conversation, which could enable the operator take some cor- rective actions to improve the quality while the call is in progress. In this paper, the authors propose a real-time fuzzy algo- rithm to estimate the strength of the line echo component of the voice quality in VoIP networks. The results obtained shows that the algorithm is able to track and estimate echo content of a live VoIP traffic in real-time. This algorithm could be em- bedded in VoIP systems to enable operators monitors calls in real-time. Keywords VoIP, Quality of Service (QoS), Mean Opinion Score (MOS), Perceptual Evaluation of Speech Quality (PESQ), Perceptual Analysis/ Measurement System (PAMS), Perceptual Speech Quality Monitor (PSQM), ETSI- Compu- tation Model (E-model) 1. Introduction A modern real-life telephony network very often consists of local Public Switched Telephone Network (PSTN) that collects voice traffic from users; and is connected to a long-distance network optimized for data transport by means of the Internet Protocol (IP). The interconnection of the PSTN to the IP network is implemented through a Me- dia Gateway (MG), which can convert the PSTN signals into the format required by the IP, and vice versa[1][9]. In order to ensure good voice quality, several non-real-time algorithms that estimate the voice quality such as the Per- ceptual Analysis Measurement System (PAMS), Perceptual Evaluation of Speech Quality (PESQ), Perceptual Speech Quality Monitor (PSQM) and the European Telecommuni- cations Standards Institute (ETSI) Computation Model (E-model) have been developed. A major hindrance in carrying voice traffic over data networks is the increased echo content, which is as a result of longer delays encountered in these networks. Echo detec- tion and cancellation is therefore critical in achieving good quality voice signal in packet switched networks, which face longer delays due to the bursty nature of the IP net- work[10]. * Corresponding author: tade@sainttade.com (Oyetade Durojaiye) Published online at http://journal.sapub.org/ijnc Copyright © 2012 Scientific & Academic Publishing. All Rights Reserved 1.1. Theoretical Background In most cases our everyday conversations take place in the presence of echoes. We hear echoes of our speech waves as they are reflected, for instance, from the floor and the walls. However, if the reflected waves arrive shortly after we spoke them, we do not perceive them as echo but as some reverberation. On the other hand if the reflected wave takes 20 or 30 milliseconds (ms) to come back to us, we will identify it as an annoying echo[5]. Echo is mainly dependent on the amount of delay present in the circuit or network. Most callers will hear echo of their own voice if the circuit contains as little as 30 milliseconds of round-trip delay. If the round-trip delay approaches 50 ms, virtually all callers will complain of echo if it is left uncontrolled[3][6]. Echo is closely related to other factors such as delay, jitter and packet loss that affect the voice quality of a VoIP call. Delay is introduced into the telecommunications network primarily by transmission facilities and transmission equipment. The delay could be negligible or significant[12]. Depending on the network topology, and the type of trans- mission equipment used in the network, 30 ms of roundtrip delay can occur in connections that are across country or just across town[7]. The International Telecommunication Union (ITU) has a guide for the amount of delay introduced by specific trans- mission medium as shown in Table 1.