Enhancing the quality of low bit‐rate real‐time Internet communication services

Abstract

The Internet environment, with its packet‐switched network and lack of resource reservation mechanisms, has made the delivery of low bit‐rate real‐time communication services particularly difficult and challenging. The high potential transmission delay and data packet loss under varying network conditions will lead to unpleasant and unintelligible audio and jerky video play‐out. The Internet TCP/IP protocol suite can be extended with new mechanisms in an attempt to tackle such problems. In this research, an integrated transmission mechanism that incorporates a number of existing techniques to enhance the quality and deliver “acceptable” real‐time services is proposed. These techniques include the use of data compression, data buffering, dynamic rate control, packet lost replacement, silence deletion and virtual video play‐out mechanism. The proposed transmission mechanism is designed as a generic communication system so that it can be used in different systems and conditions. This approach has been successfully implemented and demonstrated using three separate systems that include the Internet Phone, WebVideo and video‐conferencing tool.