الفهرس 
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Abstract
Voice over Internet Protocol (VoIP) has the potential to be integrated with other Internet applications to provide interactive multimedia communication services that are difficult to deploy over the traditional telephony circuit-switched networks. To fully exploit the benefit of service integration, it is necessary that VoIP services can be seamlessly provided with a good Quality-of Service (QoS) over several different network technologies. Voice over Internet Protocol (VoIP) enables voice traffic to be carried over an IP network such as the global Internet. The goal of this thesis is to improve the quality of VoIP considering current network problems such as packet loss. We propose a scheme that uses redundancy at the sender side. The sender transmits a redundant copy of the stream based on lower-rate encoding technique on another path. This redundancy is supposed to increase the likelihood of a receiver to replace lost or delayed packets from the original speech stream using the redundant copy. The proposed solution is to be implemented mainly at the end hosts and can be supported with minimal changes to the infrastructure of the Internet. This thesis passed on collaboration between the research communities in Egypt and the USA. Simulation is performed using ns-2 simulator. Practical implementation is performed to show that voice quality can be significantly increased for long distance links, where packet loss is high, by using routing diversity than just sending all the packets through one path.