Abstract: Voice over Internet Protocol (VoIP) is a cutting-edge technology that transports audio data across a packet-switched network. The benefits of VoIP have convinced even some skeptics. However, the quality of service (QoS) of VoIP solutions is still a major concern among current users and non-adopters. Therefore, achieving superior QoS is one key challenge that designers must meet before VoIP technology receives better acceptance. Numerous areas in a VoIP system can affect QoS, and improvements in one area may impact other areas. Therefore, this research proposes complementary improvements in multiple components of VoIP solutions. This research includes a survey of existing infrastructure QoS tools and architectures. The survey leads to an in-depth study of various congestion-avoidance techniques, finding that no existing method can provide superior QoS for VoIP. This research proposes a new adaptive transmission-control mechanism that is effective at curbing congestion for VoIP while reducing the amount of packet loss. This research also studies various well-known sender-based loss-recovery techniques. The experimental results show that a simple loss-recovery algorithm can improve QoS significantly in the face of packet loss. This research additionally examines various approaches for handling delay jitter and proposes a practical, low-complexity, and efficient mechanism to manage delay jitter. A commercial VoIP solution has now proven this delay-jitter mechanism to be effective in providing superior QoS. Bandwidth scarcity is a common root cause of various problems that affect QoS. Therefore, this research explores innovative solutions to improve resource utilization, and proposes a new teleconference technique that reduces bandwidth consumption by almost 50% compared to existing approaches. VoIP systems use audio codecs (coder/decoders) to encode and decode audio data to reduce bandwidth consumption. Consequently, correct implementation of codec software is crucial for providing superior QoS. This research introduces a creative and effective system for testing and validating real-time codec software. This innovative system can test and validate a codec implementation in a live VoIP session, and the system reduces software debugging time from weeks or months down to mere hours. Furthermore, this system can evaluate the QoS of a live VoIP session in real time.