%0 Journal Article
%T Distortion-Based Slice Level Prioritization for Real-Time Video over QoS-Enabled Wireless Networks
%A Ismail A. Ali
%A Martin Fleury
%A Mohammed Ghanbari
%J Advances in Multimedia
%D 2012
%I Hindawi Publishing Corporation
%R 10.1155/2012/319785
%X This paper presents a prioritization scheme based on an analysis of the impact on objective video quality when dropping individual slices from coded video streams. It is shown that giving higher-priority classified packets preference in accessing the wireless media results in considerable quality gain (up to 3£¿dB in tests) over the case when no prioritization is applied. The proposed scheme is demonstrated for an IEEE 802.11e quality-of-service- (QoS-) enabled wireless LAN. Though more complex prioritization systems are possible, the proposed scheme is crafted for mobile interactive or user-to-user video services and is simply implemented within the Main or the Baseline profiles of an H.264 codec. 1. Introduction There have recently emerged two forms of video streaming to mobile devices. The first, HTTP adaptive streaming [1], employing reliable TCP transport, has no need to protect the video stream against channel errors but is subject to delays. These delays mainly arise from the repeated transmissions that TCP imposes whenever packets are lost. Additionally, delay may occur due to the pull-based nature of the service. Therefore, though suitable for some forms of one-way commercial streaming, HTTP adaptive streaming is unsuitable for interactive services such as video conferencing. It is also unsuitable for mobile user-to-user streaming, because of the need to create multiple copies of the same video at different resolutions and set up a complex management structure to allow client access to an appropriate stream. Therefore, a second native form of streaming is necessary for delay- or storage-intolerant video streaming, and it is this form of streaming that is the subject of this paper. In this form of streaming [2], video is pushed from the server without the need for a feedback channel to make continual client requests. The Real-time Transport Protocol (RTP) with underlying Internet Protocol (IP)/User Datagram Protocol (UDP) for network routing and transport updates the client-side decoder with synchronization information. If MPEG-2 Transport Stream (TS) packets are multiplexed within each RTP packet, then audio can accompany video in a single packet stream. Adaptive bitrate adjustments (through scalable coding or transcoding) can occur, based on performance metrics carried by Real-time Transport Control Protocol (RTCP) packets, and pseudo-VCR functionality, if needed, is available through the Real-time Streaming Protocol (RTSP). When mobile video streaming in native mode with IP/UDP/RTP packetization, there is a need to avoid periodic increased
%U http://www.hindawi.com/journals/am/2012/319785/