Real-time interactive video streaming over lossy networks: high performance low delay error resilient algorithms

According to Cisco's latest forecast, two-thirds of the world's mobile data traffic and 62 percent of the consumer Internet traffic will be video data by the end of 2016. However, the wireless networks and Internet are unreliable, where the video traffic may undergo packet loss and delay. Thus robust video streaming over unreliable networks, i.e., Internet, wireless networks, is of great importance in facing this challenge. Specifically, for the real-time interactive video streaming applications, such as video conference and video telephony, the allowed end-to-end delay is limited, which makes the robust video streaming an even more difficult task. In this thesis, we are going to investigate robust video streaming for real-time interactive applications, where the tolerated end-to-end delay is limited. Intra macroblock refreshment is an effective tool to stop error propagations in the prediction loop of video decoder, whereas redundant coding is a commonly used method to prevent error from happening for video transmission over lossy networks. In this thesis two schemes that jointly use intra macroblock refreshment and redundant coding are proposed. In these schemes, in addition to intra coding, we proposed to add two redundant coding methods to enhance the transmission robustness of the coded bitstreams. The selection of error resilient coding tools, i.e., intra coding and/or redundant coding, and the parameters for redundant coding are determined using the end-to-end rate-distortion optimization. Another category of methods to provide error resilient capacity is using forward error correction (FEC) codes. FEC is widely studied to protect streamed video over unreliable networks, with Reed-Solomon (RS) erasure codes as its commonly used implementation method. As a block-based error correcting code, on the one hand, enlarging the block size can enhance the performance of the RS codes; on the other hand, large block size leads to long delay which is not tolerable for real-time video applications. In this thesis two sub-GOP (Group of Pictures, formed by I-frame and all the following P/B-frames) based FEC schemes are proposed to improve the performance of Reed-Solomon codes for real-time interactive video applications. The first one, named DSGF (Dynamic sub-GOP FEC Coding), is designed for the ideal case, where no transmission network delay is taken into consideration. The second one, named RVS-LE (Real-time Video Streaming scheme exploiting the Late- and Early-arrival packets), is more practical, where the video transmission network delay is considered, and the late- and early-arrival packets are fully exploited. Of the two approaches, the sub-GOP, which contains more than one video frame, is dynamically tuned and used as the RS coding block to get the optimal performance. For the proposed DSGF approach, although the overall error resilient performance is higher than the conventional FEC schemes, that protect the streamed video frame by frame, its video quality fluctuates within the Sub-GOP. To mitigate this problem, in this thesis, another real-time video streaming scheme using randomized expanding Reed-Solomon code is proposed. In this scheme, the Reed-Solomon coding block includes not only the video packets of the current frame, but also all the video packets of previous frames in the current group of pictures (GOP). At the decoding side, the parity-check equations of the current frameare jointly solved with all the parity-check equations of the previous frames. Since video packets of the following frames are not encompassed in the RS coding block, no delay will be caused for waiting for the video or parity packets of the following frames both at encoding and decoding sides. The main contribution of this thesis is investigating the trade-off between the video transmission delay caused by FEC encoding/decoding dependency, the FEC error-resilient performance, and the computational complexity. By leveraging the methods proposed in this thesis, proper error-resilient tools and system parameters could be selected based on the video sequence characteristics, the application requirements, and the available channel bandwidth and computational resources. For example, for the applications that can tolerate relatively long delay, sub-GOP based approach is a suitable solution. For the applications where the end-to-end delay is stringent and the computational resource is sufficient (e.g. CPU is fast), it could be a wise choice to use the randomized expanding Reed-Solomon code

The QUIC Fix for Optimal Video Streaming

Within a few years of its introduction, QUIC has gained traction: a significant chunk of traffic is now delivered over QUIC. The networking community is actively engaged in debating the fairness, performance, and applicability of QUIC for various use cases, but these debates are centered around a narrow, common theme: how does the new reliable transport built on top of UDP fare in different scenarios? Support for unreliable delivery in QUIC remains largely unexplored. The option for delivering content unreliably, as in a best-effort model, deserves the QUIC designers' and community's attention. We propose extending QUIC to support unreliable streams and present a simple approach for implementation. We discuss a simple use case of video streaming---an application that dominates the overall Internet traffic---that can leverage the unreliable streams and potentially bring immense benefits to network operators and content providers. To this end, we present a prototype implementation that, by using both the reliable and unreliable streams in QUIC, outperforms both TCP and QUIC in our evaluations.Comment: Published to ACM CoNEXT Workshop on the Evolution, Performance, and Interoperability of QUIC (EPIQ

Improving The Efficiency Of Video Transmission In Computer Networks

In-depth examination of current techniques for enhancing the efficiency of video transmission over digital networks is provided in this study. Due to the growing need for high-quality video content, optimizing video transmission is an important area of research. This review categorizes and in-depth examines a range of methods proposed in the literature to enhance video transmission effectiveness. ABR, DNN architecture, adaptive streaming, Quality of Service (QoS), error resilience, congestion control, video compression, and hardware acceleration for video provisioning are just a few of the cutting-edge techniques that are covered in the discussion, which ranges from the more traditional to the cutting-edge. This essay provides a methodical evaluation of the numerous tactics that are available, along with an analysis of their guiding principles, advantages, and disadvantages. The paper also offers a comparative analysis of various approaches, highlighting trends, gaps, and potential future research directions in this crucial domain, all of which help to create more efficient video compression and transmission paradigms in computer networks

An Adaptive Fuzzy based FEC Algorithm for Robust Video Transmission over Wireless Networks

Forward Error Correction (FEC) is a commonly adopted mechanism to mitigate packet loss/bit error during real-time communication. An adaptive, Fuzzy based FEC algorithm to provide a robust video quality metric for multimedia transmission over wireless networks has been proposed to optimize the redundancy of the generated code words from a Reed-Solomon encoder and to save the bandwidth of the network channel. The scheme is based on probability estimations derived from the data loss rates related to the recovery mechanism at the client end. By applying the adaptive FEC, the server uses the reports to predict the next network loss rate using a curve-fitting technique to generate the optimized number of redundant packets to meet specific residual error rates at the client end. Simulation results in the cellular system show that the video quality is massively adapted to the optimized FEC codes based on the probability of packet loss and packet correlation in a wireless environment

Adaptive end-to-end optimization of mobile video streaming using QoS negotiation

Video streaming over wireless links is a non-trivial problem due to the large and frequent changes in the quality of the underlying radio channel combined with latency constraints. We believe that every layer in a mobile system must be prepared to adapt its behavior to its environment. Thus layers must be capable of operating in multiple modes; each mode will show a different quality and resource usage. Selecting the right mode of operation requires exchange of information between interacting layers. For example, selecting the best channel coding requires information about the quality of the channel (capacity, bit-error-rate) as well as the requirements (latency, reliability) of the compressed video stream generated by the source encoder. In this paper we study the application of our generic QoS negotiation scheme to a specific configuration for mobile video transmission. We describe the results of experiments studying the overall effectiveness, stability, and dynamics of adaptation of our distributed optimization approach