Issues in Mpeg - 2 Transport over Atm

he explosi ve growth of t he Intern et has created demand for new applications traditionally carried over circuit-switched networks. Such applications include audio telephony, video conferencing, and video-on-demand (VoD) services. New standards [1–3] are emerging to support these applications in the context of both connectionless and connection-oriented packet-switched networks. Asynchronous Transfer Mode (ATM) is a standard for broadband networks that allows a wide range of traffic types – ranging from real-time video to best-effort data – to be multi-plexed in a single physical connection-oriented network. A key benefit of ATM technology is its ability to provide quality-of-service (QoS) guarantees to applications. These QoS guarantees are in the form of bounds on end-to-end delay, packet delay variation (jitter), and packet loss rate for a given data rate. Several classes of service have been defined in the context of ATM networks [3] to satisfy the QoS needs of various applications. The constant bit-rate (CBR) and real-time variable bit-rate (RT-VBR) service classes are intended for real-time applications with stringent requirements on delay, jitter, and loss rate, such as video-on-demand (VoD) services. The non real-time variable bit-rate (NRT-VBR) service class is intended for applications where no jitter control is needed, but a delay guarantee is still required. The available bit-rate (ABR) service class is intended for delay-tolerant best-effort applications and uses a rate-based feedback approach to control potential congestion. Regular TCP applications fall under this service class. Finally, the unspecified bit-rate service (UBR) does not offer any service guarantees, has the lowest priority among all the classes, and implements the best-effort service found in traditional IP (Internet Protocol) networks. Within a service class, the feasibility of supporting a specified set of QoS requirements is determined by admission-control algorithms [4]. Because of its ability to support a pre-specified set of requirements, ATM technology is inherently well suited for the design of video dial tone (VDT) networks [5]. Besides the underlying network infrastructure, the coding method used for digital video and audio for VDT services has a significant influence on the viability and performance of such services. MPEG-2 is currently the most popular standard for audio and video compression in VDT networks [2, 1]. Being capable of exploiting both spatial and temporal redundancies , it achieves compression ratios up to 200:1 and can encode a video or audio source to almost any level of quality. The MPEG-2 standard offers two ways of multiplexing elementary audio, …