In this thesis, we proposed a control model which can be used to address the voice quality of VoIP with packet loss and loss recovery. Based on this model we presented a control mechanism for VoIP to recover the voice quality under lossy network. In this control mechanism, the error correction is performed at packet level with Reed Solomon codes and a control table is intro-duced. The control table is used to select the RS coding for error correction, and the use of this table can reduce the computing complexity at runtime. We implemnted the proposed mechanism on a home gateway with VoIP capability and carried out some experiments. The results show that by using the proposed mechanism the voice quality can be maintained above the designated level under the packet loss rate up to 30%, and the bandwidth overhead is more efficient than the traditional AFEC mechanisms. As a next step, his mechanism can be extended to support more than one codec at the same time. This is a future work which can make the control mechanism more completed.
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