Speech playout buffering based on a simplified version of the ITU-T E-model

In Internet-protocol (IP) telephony, problems of transmission delay variations are frequently addressed with adaptive dejitter buffering techniques. These are aimed at setting the buffer dimension so as to limit the packet end-to-end delay, the total packet loss, or both together. The selection of delay and loss limits is of key importance for the resulting conversational quality. This problem is addressed in this letter, whose main contribution is the introduction of a perceptually motivated optimality criterion that allows the receiver to automatically balance packet delay versus packet loss. In the proposed approach, the dejitter buffer size is adaptively set, and the adopted criterion relies on the use of a simplified version proposed by Cole and Rosenbluth of the conversational-quality International Telecommunication Union (ITU) Telecommunication Standardization Sector (ITU-T) E-Model.