Enhancing the quality of Internet voice communication for
Internet telephony systems
K.V. Chin, S.C. Hui and S. Foo
School of Applied Science, Nanyang Technological University
Nanyang Avenue, Singapore 639798
Summary
Internet real-time voice communication involves transmitting digitised voice signals which
can be lost in the packet switched network environment of TCP/IP, thereby causing
intermittent voice losses and quality degradations. Various methods such as silence
substitution, waveform substitution, sample interpolation, Xor mechanism, embedded speech
coding, and combined rate and control mechanism have been proposed to enhance voice
delivery and to minimise the quality impact caused by these losses. This paper proposes a
quality-based dynamic voice recovery mechanism that combines network transmission
control and voice recovery to deliver voice signals with optimal intelligibility and quality.
This is accomplished by considering the subjective rating of different codecs that are used in
the coding and transmission of digital audio and network packet loss conditions. The dynamic
mechanism results in voice delivery that, at minimum, satisfies voice intelligibility while
tolerating moderate packet loss caused by network congestion. This mechanism has been
successfully incorporated into the Internet Telephone Software System developed at the
School of Applied Science, Nanyang Technological University.
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1. Introduction
Internet telephony systems are basically synchronous distributed systems whereby two users
who are physically separated are able to carry out real-time voice communication over the
Internet. Currently, a wide range of academic and commercial Internet telephony systems
such as IPhone [1], NetMeeting [2] and NeVoT [3] has been developed. This growing interest
is mainly motivated from huge potential cost savings by making it possible to make
transcontinental telephone calls at the prices of local telephone calls plus nominal standard
Internet connectivity charges. However, the quality of communica