aacPlus is a highly efficient audio codec that is being used in a growing number of applications where the compressed audio data is encapsulated in a real-time protocol and transmitted over error prone channels. In this paper the implication of packet losses during transmission and techniques to mitigate the impact on the resulting audio quality are discussed. Example transmission channel characteristics are used to show how typical protocol configuration parameters are derived. The benefits of the described techniques are evaluated and verified by setting up a complete simulation chain and performing listening tests.
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