Quality codec interface for VoIP on differentiated services networks

As converged services are becoming the standard for corporate networks, concerns have been arising on the best way to ensure efficient use of expensive WAN bandwidth. Achieving excellent quality of multimedia and voice communication requires the allocation of large bandwidth for these services. In most cases, lossy compression codecs are used to reduce bandwidth requirements resulting in lower quality of service. The codec selection is maintained even during low bandwidth utilization. In this paper, we present a codec interface that combines high-fidelity high-bitrate and low-fidelity low-bitrate codecs. Audio packets from the low-bitrate codec are transmitted on the highest priority class of service of differentiated services networks. This ensures that these packets always arrive at destination. Audio packets from the high-bitrate codec are transmitted on the lowest priority class of service. At low bandwidth utilization, the high-fidelity high-bitrate packets will be used and during congestions, these packets would be dropped. This ensures achieving the best possible voice quality during varying network loads. Simulation results measuring the mean opinion score of voice quality show that superior voice quality can be achieved with this method over using standalone codecs while maintaining excellent quality of service for data traffic as well.