Adaptive streaming of MPEG-based audio/video content over wireless networks

This paper describes a method for robust streaming of combined MPEG audio/video content (encoded either with MPEG-2 or MPEG-4/H.264) over in-home wireless networks. We make use of currently used content distribution format (MPEG Transport Stream) and network protocols (either RTP or TCP/HTTP). The transmitted bit-rate is constantly adapted to the available network bandwidth, such that audio and video artifacts caused by packet loss are avoided. Bit-rate adaptation is achieved by using a packet scheduling technique called I-Frame Delay (IFD), which performs priority-based frame dropping upon insufficient bandwidth. We show an implementation using RTP and an implementation using TCP. Measurements on a real-life demonstrator set-up demonstrate the effectiveness of our approach.