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Chapter 16
DOI: 10.4018/978-1-4666-7377-9.ch016
Quality Analysis of VoIP in
Real-Time Interactive Systems
over Lossy Networks
ABSTRACT
Voice over Internet Protocol (VoIP) systems have been spreading massively during the recent years.
However, many challenges are still facing this technology among which is the lossy behavior and the
uncontrolled network impairments of the Internet. In this chapter, the authors design and implement a
VoIP test-bed utilizing the Adobe Real-Time Media Flow Protocol (RTMFP) that can be used for many
voice interactive applications. The test-bed was used to study the efect of changing some voice param-
eters, mainly the encoding rate and the number of frames per packet as function of the network packet
loss. Several experiments were conducted on several voice fles over diferent packet losses, concluding
in the best combination of parameters in low, moderate, and high packet loss conditions to improve the
performance of voice packets measured by the Perceptual Evaluation of Speech Quality (PESQ) values.
INTRODUCTION
The market penetration of VoIP has massively
influenced traditional telephony as witnessed by
the popularity of many VoIP services and applica-
tions (e.g. SKYPE, Cisco TelePresence, Ekiga and
Oovoo (Vaughan, 2012)). It has become one of
the most popular Internet Protocol (IP) based real-
time communication services in recent years, it is
of no doubt that VoIP networks are cost-effective
over conventional telephone networks with Time
Division Multiplexing (TDM) and leased lines.
In real-time interactive systems, VoIP has been
entitled an important role in enabling voice based
real-time services, such as voice conferencing, that
is applied in various sectors and fields every day.
According to Zhao & Yagi & Nakajima & Juzoji,
(2002). VoIP is taking a firm hold in the telecom-
munication market for a wide range of interactive
applications including telemedicine and e-health.
Maha Z. Mouasher
The University of Jordan, Jordan
Ala’ F. Khalifeh
German-Jordanian University, Jordan