Research And Implementation Of The QoS Technologies In VoIP Terminal System

Since mid 90's, the fast development of Internet has taken great effect on the commercial telecommunication service, and multimedia communication system based on packet switching network becomes the focus of people's concern. Audio frequency and video frequency compression technology, real-time data transmission technology and network technology make it possible to achieve multimedia communication on IP network. As a kind of new voice communication technologies, VoIP has been more and more popular in modern life. VoIP means transmitting voice on IP network. As one kind of voice and facsimile communication technologies based on IP network, VoIP features high transmission efficiency and low cost. It has been applied extensively in many fields. Being combined with phone network and multimedia technology, the VoIP technology can create numerous new businesses. Since VoIP develops so quickly in current telecommunication service, it deserves to be researched to a deep extent.Firstly, this dissertation discusses the basic theory of VoIP and several different models for constructing VoIP systems. Next, we focus on the key technologies to implement the QoS (Quality of Service) in VoIP terminal systems, which is the main topic of this discussion. The limitation of network bandwidth can lead to the loss of packets at the receiving end. This kind of packet loss degrades the speech quality, and causes impairments present in IP network, such as namely jitter and delay. To deal with the phenomenon that the network bandwidth changes dynamically and frequently, this dissertation presents two adaptive QoS technologies, namely Dynamic Speech Codec and Adaptive Packet Loss Compensation Strategy. During the real-time changing procedure of the bandwidth, these two technologies can dynamically adjust the loss compensation so as to guarantee QoS of the VoIP system. In addition, acoustic echo is one of the most prominent problems which decrease the quality of speech in VoIP if loudspeakers are used. To deal with that problem, an echo suppression method is proposed based on remote voice activity detection. The method is effective in a variety of application scenarios, and is easy to implement. The corresponding echo suppressor is successfully applied in a practical VoIP system. Testing results show that this method can effectively suppress the echo, and can achieve practical utility.Finally, a VoIP software system called PPTalk is discussed. PPTalk is developed based on the technologies presented in this dissertation. The software design and implementation of the system has been presented in detail. Testing shows that, in the case of a broadband network, the performance and QoS of PPTalk is as good as those of the most popular VoIP system Skype; however, in the case of a narrow bandwidth, the QoS of PPTalk can achieve better MOS scores than Skype does, especially in terms of the continuity of voice.