A source and channel rate adaptation algorithm for AMR in VoIP using the Emodel

We present a dynamic joint source and channel coding adaptation algorithm for the AMR speech codec based on the ITU-T Emodel. This model takes both delay and packet loss into consideration. We address the problem of finding the optimal choice of source and channel bit rates given QoS information about the wired and wireless IP network and subject to constraints on maximum packet loss, maximum delay and maximum allowed transmission rate. Our results show that an adaptation is necessary to preserve acceptable levels of quality while making optimal use of the allowed bandwidth. Our technique requires a small number of computations that allows real time operation in parallel to voice streams.

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