Abstract
Voice communications over the Internet are growing rapidly thanks to faster Internet and mobile networks. Although multimedia communication techniques such as voice over IP (VoIP) have been using UDP, TCP is gaining gradual popularity since it enables passage through network address translators (NATs) and firewalls. Many researches have been done on real-time communications using TCP, which indicates the possibility of achieving low delay communications over TCP. However, little mention has been made of user-perceived voice quality over TCP. This paper presents a VoIP system using multiple TCP connections. We conducted thorough voice quality evaluations over these connections by changing network and VoIP application parameters. Our evaluations show that voice communications can be achieved in 10% packet loss rate and 100 ms round-trip time environments if the number of TCP connections, the sending unit and the decoding buffer size are optimized. We also provide optimal parameter guidelines for achieving high quality voice communications over multiple TCP connections.
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