Research To The Quality Of Voice Over IP

Due to the rapid development of computer and broadband network technology, the multimedia communication-based information network has been the world focus. The transferred information by Internet is converted from simple text information to rich and colorful multimedia information, such as voice,images,graphics and real time video. IP Telephony as the provision of telephone-like services over the Internet is considered as one of the goals of the next generation network (NGN). Despite the advantages of IP Telephony, however, there are a number of barriers to keep it from more widespread use. The most prominent among them is the poor quality of voice connections on the Internet. Designed to handle non-real time data traffic, Internet can only provide "best effort" service which cannot satisfy the Quality of Service requirement for carrying voice and video. How to improve its real-time capability so that QoS of communication can be guaranteed is a critical technique issue and one of the technique difficulties of IP Telephone.Firstly, this dissertation discusses the basic theory of VoIP, the several models of its construction, and sets forth its main protocols such as H.323, SIP, RTP/RTCP, at the same time points out these protocols' characteristic and the differences of each other. During the following dissertation, I simply introduce the several key technologies about VoIP besides presenting the factors of influencing the Quality of Voice and the technologies of improving the Quality of Voice. The discussion of QoS(Quality of Service) in VoIP is our working center. Impairments present on IP networks, namely jitter and delay can lead to the loss of packets at the receiving end. This packet loss degrades the speech quality. Then I explore and analyze the techniques of jitter buffer and packet recovery used to improving the quality of voice in detail and has carried on a experiment based on static jitter buffer.