Robust audio transmission over internet with self-adjusted buffer control

Abstract

This paper presents some main problems found in audio transmission over Internet, and proposes an approach to overcome side effects of packet loss and transmission delay variation. The new approach is realized by adopting the concept of robust transmission at the sender site and implementing a self-adjusted buffer (SAB) scheme at the receiver site. Comparing with the traditional approach, in which a normal packet containing a single audio frame, a robust packet containing some auxiliary audio frames is transmitted periodically. When a normal packet is lost, the corresponding frame can be recovered using the auxiliary frame contained in the robust packet. In this way, more robust audio transmission can be achieved in Internet. Additionally, it is difficult to synchronize the communication at the receiver side, because there is no way to control the network jitter in Internet. A fully buffering functionality scheme called self-adjusted buffer (SAB) control scheme and a shorter playback timer (SPT) implementation approach are proposed in this paper. SAB is adopted to achieve a self-adjusted synchronization at the receiver side. SAB adjusts the receiver process automatically to accommodate itself to the current network traffic. A robust Internet telephone (R-Iphone) system is developed based on the robust packet concept and SAB synchronization technique.