Cross layer techniques for flexible transport protocol using UDP-Lite over a satellite network

Traditional real-time multimedia and streaming services have utilised UDP over RTP. Wireless transmission, by its nature, may introduce a variable, sometimes high bit error ratio. Current transport layer protocols drop all corrupted packets, in contrast, protocols such as UDP-Lite allow error-resilient applications to be supported in the networking stack. This paper presents experimental quantitative performance metrics using H.264 and UDP Lite for the next generation transport of IP multimedia, and discusses the architectural implications for enhancing performance of a wireless and/or satellite environment

Building self-optimized communication systems based on applicative cross-layer information

This article proposes the Implicit Packet Meta Header(IPMH) as a standard method to compute and represent common QoS properties of the Application Data Units (ADU) of multimedia streams using legacy and proprietary streams’ headers (e.g. Real-time Transport Protocol headers). The use of IPMH by mechanisms located at different layers of the communication architecture will allow implementing fine per-packet selfoptimization of communication services regarding the actual application requirements. A case study showing how IPMH is used by error control mechanisms in the context of wireless networks is presented in order to demonstrate the feasibility and advantages of this approach

SoftCast: Clean-slate Scalable Wireless Video

Video broadcast and mobile video challenge the conventional wireless design. In broadcast and mobile scenarios the bit rate supported by the channel differs across receivers and varies quickly over time. The conventional design however forces the source to pick a single bit rate and degrades sharply when the channel cannot not support the chosen bit rate. This paper presents SoftCast, a clean-slate design for wireless video where the source transmits one video stream that each receiver decodes to a video quality commensurate with its specific instantaneous channel quality. To do so, SoftCast ensures the samples of the digital video signal transmitted on the channel are linearly related to the pixels' luminance. Thus, when channel noise perturbs the transmitted signal samples, the perturbation naturally translates into approximation in the original video pixels. Hence, a receiver with a good channel (low noise) obtains a high fidelity video, and a receiver with a bad channel (high noise) obtains a low fidelity video. We implement SoftCast using the GNURadio software and the USRP platform. Results from a 20-node testbed show that SoftCast improves the average video quality (i.e., PSNR) across broadcast receivers in our testbed by up to 5.5dB. Even for a single receiver, it eliminates video glitches caused by mobility and increases robustness to packet loss by an order of magnitude

Generative Compression

Traditional image and video compression algorithms rely on hand-crafted encoder/decoder pairs (codecs) that lack adaptability and are agnostic to the data being compressed. Here we describe the concept of generative compression, the compression of data using generative models, and suggest that it is a direction worth pursuing to produce more accurate and visually pleasing reconstructions at much deeper compression levels for both image and video data. We also demonstrate that generative compression is orders-of-magnitude more resilient to bit error rates (e.g. from noisy wireless channels) than traditional variable-length coding schemes

Video streaming across wide area networks. Research on technologies for content distribution

This research aims to focus on several aspects of video streaming: compression algorithms, broadcasting daemons, encoding technologies, patenting and licensing issues, secure authentication measures and archiving. A range of possibilities is to be examined to individuate a viable technology which will be the underlying framework for Wide Area Net- work (WAN) access to the Video Art Collection of Montevideo. Main requirements are quality of the video playback, standardization and predictable longevity of the compression algorithm adopted. Other parameters we‘ll take into account in our exploration of digital video streaming technologies are: efficiency, costs, compatibility, integration with adopted technologies, independence and freedom to adapt the tools to our needs. Being the field of research a rapidly evolving ground, this research has to be referred as the current status of technology in year 2006. At the time of writing some promising technologies are still in development and, once they achieved maturity, they might slightly affect our results

Scalable Video Coding

International audienceWith the evolution of Internet to heterogeneous networks both in terms of processing power and network bandwidth, different users demand the different versions of the same content. This has given birth to the scalable era of video content where a single bitstream contains multiple versions of the same video content which can be different in terms of resolutions, frame rates or quality. Several early standards, like MPEG2 video, H.263, and MPEG4 part II already include tools to provide different modalities of scalability. However, the scalable profiles of these standards are seldom used. This is because the scalability comes with significant loss in coding efficiency and the Internet was at its early stage. Scalable extension of H.264/AVC is named scalable video coding and is published in July 2007. It has several new coding techniques developed and it reduces the gap of coding efficiency with state-of-the-art non-scalable codec while keeping a reasonable complexity increase. After an introduction to scalable video coding, we present a proposition regarding the scalable functionality of H.264/AVC, which is the improvement of the compression ratio in enhancement layers (ELs) of subband/wavelet based scalable bitstream. A new adaptive scanning methodology for intra frame scalable coding framework based on subband/wavelet coding approach is presented for H.264/AVC scalable video coding. It takes advantage of the prior knowledge of the frequencies which are present in different higher frequency subbands. Thus, by just modification of the scan order of the intra frame scalable coding framework of H.264/AVC, we can get better compression, without any compromise on PSNR

SoftCast

The focus of this demonstration is the performance of streaming video over the mobile wireless channel. We compare two schemes: the standard approach to video which transmits H.264/AVC-encoded stream over 802.11-like PHY, and SoftCast -- a clean-slate design for wireless video where the source transmits one video stream that each receiver decodes to a video quality commensurate with its specific instantaneous channel quality

Performance evaluation of TCP, UDP and DCCP for video traffics over 4G network

Fourth Generation (4G) system has been used more widely than the older generations 3G and 2G. Among the reasons are that the 4G’s transfer rate is higher and it supports all multimedia functions. Besides, its’ supports for wide geographical locus makes wireless technology gets more advanced. The essential goal of 4G is to enable voice-based communication being implemented endlessly. To achieve the goal, this study tries to answer the following research questions: (1), are the old protocols suit with this new technology; (2), which one has the best performance and, (3) which one has the greatest effect on throughput, delay, packet delivery ratio and packet loss. The aforementioned questions are crucial in the performance evaluation of the most famous protocols (particularly User Datagram Protocol (UDP), Transmission Control Protocol (TCP), and Datagram Congestion Control Protocol (DCCP)) within the 4G environment. Through the Network Simulator-3 (NS-3), the performance of transporting MPEG-4 video stream including throughput, delay, packet loss, and packet delivery ratio are analyzed at the base station through UDP, TCP, and DCCP protocols over 4G’s Long Term Evolution (LTE) technology. The results show that DCCP has better throughput, and lesser delay, but at the same time it has more packet loss than UDP and TCP. Based on the results, DCCP is recommended as a transport protocol for real time vide