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Multimedia streaming applications are traditionally delivered over UDP. Recent measurements show that more and more multimedia streaming data are over TCP as web-based TV, P2P streaming, video sharing websites are getting increasingly popular. To improve the quality of experience (QoE) for users and to cope with variability in TCP throughput, streaming applications typically implement buffers. Yet, for improving the QoE and the streaming quality, e.g., playback continuity and timeliness, it is critical to dimension buffers and the initial buffering delay appropriately. [ABSTRACT FROM PUBLISHER]

Constant bit-rate (CBR) encoding methods used for the real-time digital video applications such as terrestrial high definition television (HDTV) broadcasting often produce large picture quality fluctuation due to constraints on buffer size and buffering delay. In this paper, we propose a simple framelayer CBR encoding method which reduces large picture quality fluctuation of an encoded video under constant bit-rate and limited buffer size. We first define an intermediate encoder buffer level to be lower than the half of the maximum encoder buffer level. Then, the encoder buffer level is kept from being close to the maximum (or minimum) buffer level during encoding by slightly decreasing (or increasing) the number of bits estimated to encode each frame if the resulting buffer level becomes higher (or lower) than the intermediate buffer level. The experimental results show that the proposed CBR encoding method reduces the average local PSNR variation by 20.6% compared with previous CBR encoding method on average under constant bit-rate and limited buffer size. [ABSTRACT FROM PUBLISHER]

A critical component in the delivery of VBR streaming video over packet-based networks, is the playout buffer at the receiver side. As packets will incur a variable delay within the network, a mechanism is needed to dejitter the stream and assure that packets are played out at exactly the same rate at which they were generated by the sender. Packet loss at the playout buffer can be induced by either underflow or overflow, and we first identify the general conditions under which these events occur. These general rules require some knowledge on the network delay characteristics of an arbitrary packet, and are then further developed and refined, by taking a closer look at the case where a single delay bottleneck exists within the network. This analysis allows us to derive useful design rules, where we consider two cases, depending on whether or not the receiver has knowledge about the delay the first packet of the stream has incurred. We conclude by demonstrating that the traditional playout paradigm can be enhanced to guard against large initial delays. [ABSTRACT FROM AUTHOR]