In recent years, with the development of information technology, the networkconstruction including the wireless and the fixed phone system is developed rapidly.The IP version 6 will be applied in IP network and all the traffic will be integratedtogether including the data network, the CATV network and the fixed phone network.It can be said that voice over IP will be used widely, and related technologies likespeech coding, network transmission and integration between wireless and fixedphone network have become the focus. At present, most of the VoIP systems are based on the speech codec with a fixedrate, which makes packet-loss serious, especially for continuous packet-loss.Packet-loss will cause degradation of the speech quality. This is a bottleneck, whichmakes the application of VoIP limited. In this thesis, an efficient and adaptive network speech communication systemwas proposed based on above problems through analyzing all kinds of existed VoIPspeech transmission system. First, the AMR codec was investigated carefully and anew main-sub codec was proposed according to the character of AMR codec. Second,a network speech communication system for preventing packet-loss was given basedon AMR. Finally, the system is simulated by NS and C language. The computer simulation and subject listening test show that the proposedsystem outperforms traditional IP system over the average packet-loss, average delayand subjective listening quality.
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