Abstract-An important issue of supporting multi-user video streaming over wireless networks is how to optimize the systematic scheduling by intelligently utilizing the available network resources while, at the same time, to meet each video's Quality of Service (QoS) requirement. In this work, we study the problem of video streaming over multi-channel multi-radio multihop wireless networks, and develop fully distributed scheduling schemes with the goals of minimizing the video distortion and achieving certain fairness. We first construct a general distortion model according to the network's transmission mechanism, as well as the rate distortion characteristics of the video. Then, we formulate the scheduling as a convex optimization problem, and propose a distributed solution by jointly considering channel assignment, rate allocation, and routing. Specifically, each stream strikes a balance between the selfish motivation of minimizing video distortion and the global performance of minimizing network congestions. Furthermore, we extend the proposed scheduling scheme by addressing the fairness problem. Unlike prior works that target at users' bandwidth or demand fairness, we propose a media-aware distortion-fairness strategy which is aware of the characteristics of video frames and ensures maxmin distortion-fairness sharing among multiple video streams. We provide extensive simulation results which demonstrate the effectiveness of our proposed schemes.
Delivering streaming video over wired and wireless networks poses many challenges, primarily due to the throughput variations caused by time-varying network conditions. Scalable video coding gives an elegant way to adapt the video streams to the available transmission resource. In this paper, we consider a multi-user scenario wherein multiple scalable video streams compete for a shared transmission link with limited forwarding capacity. We propose a rate-distortion (RD) optimized rate shaping approach to improve the overall video quality. For this, compact RD side information is sent along with the sequences. Our simulation results show that significant improvements are achieved by the proposed RD-optimized rate shaping approach compared to conventional priority-based rate shaping.
Error trucking (ET) is an error resilience techniquefor real-time video transmission over error-prone communication channels. In this paper we propose proq-based ET for communication scenarios where the sender is in the wired Internet and the receiver is connected via a wireless link. Our main assumptions in /his work are that there is a strong imbalance behveen the tronsmission rules available in the wired and the wireless Internet and that the round-trip delay is mainly caused by the wiredpart ofthe connection. We also assume that the mojority of the packt loss is caused by the wireless link. In order to allow the proxy server to perform ewor tracking an additional update stream is sent thmugh the wired nehvork. This additional information is used by the pm.y to improve the performance on the wireless link. We show that under these assumptions pmxy-based e m r tracking leads to significantly impmved performance for H.264 based red-time video communication in comparison to traditional end-to-end error tracking. IntroductionIn 3G networks, video services are expected to be the most popular ones and maybe the key factor for success. Wireless video applications without real-time constraints (e.g. Multimedia Message Service) have been introduced to the market. However, real-time video communication over wireless networks remains challenging. Decoding of erroneous or incomplete video bit-streams leads to severe quality degradations. Because of motion compensated prediction, these impairments also propagate in space and time and therefore stay visible for a significant amount of time. Hence, an error resilient transmission scheme is essential to achieve good quality in a wireless multimedia communication system. A recent overview of approaches for error resilient video transmission can be found in Error Tracking (ET) [2],[5],[7]is an error resilience technique taking advantage of a back-channel to report corrupted image areas. The encoder reacts to this feedback by tracking the spatio-temporal error propagation. Those frame areas that have been identified to be corrupted are then updated using INTRA coding. Because the update happens in future frames to be encoded, ET does not introduce additional delay.ET is suitable for real-time applications, hut the performance is closely related to round-trip delay. In the Internet, a video sender may be located far away from the receiver and the long trip delay leads to serious error propagation in case of packet loss. Larger image areas are affected and need to be refreshed, which is critical when using a low bit-rate wireless channel, as INTRA coding leads to a bit-rate increase.If we assume that the video sender is located in the wired lntemet and the receiver is a wireless client, the round-trip time for ET is determined by the end-to-end delay between sender and receiver. One solution to cope with long delays between sender and receiver would be to use the base station that serves the mobile client as a gateway and to separate the video transmission into two separate parts:...
In this paper, we derive an analytical model for the evaluation of the performance of a Video on Demand (VoD) system. The model estimates the mean waiting time achievable by the Popularity-Aware Partial cAching (PAPA) algorithm from our previous work. Two approximation strategies are proposed for low computational complexity. Furthermore, we also consider the influence of a starting point shift on the quality of experience and combine the two factors into a universal user satisfaction metric. In order to find the relation between the two impairments, waiting time and starting point shift, sophisticated subjective tests are performed. With the final score model, a more comprehensive evaluation of the system can be obtained with very low computational complexity.
The wireless streaming media communications are fragile to the delay jitter because the conditions and requirements vary frequently with the users' mobility. Buffering is a typical way to reduce the delay jitter of media packets before the playback, however, it will incur a longer end-to-end delay. Our motivation in this paper is to improve the balance between the elimination of delay jitter and the decrease of end-to-end delay. We propose a novel adaptive playback buffer (APB) based on the probing scheme. By utilizing the probing scheme, the instantaneous network situations are collected, and then the delay margin and the delay jitter margin are employed to calculate the step length (sl) which is used to adjust the playback buffer in each time. The adaptive adjustment to the playback buffer in APB enables the continuous and real-time representation of streaming media at the receiver. Unlike the previous studies, the novelty and contributions of the paper are: a) Accuracy: by employing the instantaneous network information, the adjustment to the playback buffer correctly reflects the current network situations and therefore achieves the improved balance between the elimination of delay jitter and the decrease of end-to-end delay; Hence, APB adjustment is accurate in terms of improving such balance; b) Efficiency: by utilizing the simple probing scheme, APB achieves the current network situations without the complex mathematic predictions, which enables the adjustment to be more timely and efficient. Performance data obtained through extensive simulations show that our APB is effective to reduce both delay jitter and playback buffer delay.
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