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1.
Mobile phones are among the most popular consumer devices, and the recent developments of 3G networks and smart phones enable users to watch video programs by subscribing data plans from service providers. Due to the ubiquity of mobile phones and phone-to-phone communication technologies, data-plan subscribers can redistribute the video content to nonsubscribers. Such a redistribution mechanism is a potential competitor for the mobile service provider and is very difficult to trace given users' high mobility. The service provider has to set a reasonable price for the data plan to prevent such unauthorized redistribution behavior to protect or maximize his/her own profit. In this paper, we analyze the optimal price setting for the service provider by investigating the equilibrium between the subscribers and the secondary buyers in the content-redistribution network. We model the behavior between the subscribers and the secondary buyers as a noncooperative game and find the optimal price and quantity for both groups of users. Based on the behavior of users in the redistribution network, we investigate the evolutionarily stable ratio of mobile users who decide to subscribe to the data plan. Such an analysis can help the service provider preserve his/her profit under the threat of the redistribution networks and can improve the quality of service for end users.  相似文献   

2.
Emerging noninfrastructure-based network types like mobile ad-hoc networks (MANETs) are becoming suitable platforms for exchanging/sharing real-time video streams, because of recent progress in routing algorithms, throughput and transmission bit-rate. MANETs are characterized by highly dynamic behavior of the transmission routes and path outage probabilities. In this article a multisource streaming approach is presented to increase the robustness of real-time video transmission in MANETs. For that, video coding as well as channel coding techniques on the application layer are introduced, exploiting the multisource representation of the transferred media. Source coding is based on the scalable video coding (SVC) extension of H.264/MPEG4-AVC with different layers for assigning importance for transmission. Channel coding is based on a novel unequal packet loss protection (UPLP) scheme, which is based on Raptor forward error correction (FEC) codes. While in the presented approach, the reception of a single stream guarantees base quality only, the combined reception enables playback of video at full quality and/or lower error rates. Furthermore, an application layer protocol is introduced for supporting peer-to-peer based multisource streaming in MANETs  相似文献   

3.
Currently, VR video delivery over 5G systems is still a very complicated endeavor. One of the major challenges for VR video streaming is the expectations for low latency that current mobile networks can hardly meet. Network caching can reduce the content delivery latency efficiently. However, current caching schemes cannot obtain ideal results for VR video since it requests the viewport interactively. In this paper, we propose a tiled scalable VR video caching scheme over 5G networks. VR chunks are first encoded into multi-granularity quality layers, and are then partitioned into tiles to facilitate viewport data access. By accommodating the 5G network infrastructure, the tiles are cooperatively cached in a three-level hierarchal system to reduce delivery latency. Furthermore, a quality-adaptive request routing algorithm is designed to cater for the 5G bandwidth dynamics. Experimental results show that the proposed scheme can reduce the transmission latency over conventional constant bitrate video caching schemes.  相似文献   

4.
In a peer-to-peer (P2P) video streaming system, peers not only consume video, but also route it to other peers in the system, where ordinary peers are assumed to have sufficient downlink speed and media capability. This assumption often fails when the P2P system consists of peers that are heterogeneous in their computing power, hardware, and media capability.In this paper, we address a problem of streaming video to mobile devices, which are less capable than ordinary peers. In order to stream video to mobile devices, transcoding is often required to render video suitable for their small display, limited downlink speed, and limited video decoding capability. However, performing transcoding at a single peer is vulnerable to peer churn, which leads to video disruption. We propose interleaved distributed transcoding (IDT), a robust video encoding scheme that allows peers more capable than mobile devices to perform transcoding in a collaborative fashion. IDT is designed in such a way that transcoded substreams are assembled into a single video stream, which can be decoded by any H.264/AVC baseline profile compliant decoder. Extensive simulations and its implementation in a real P2P system demonstrate that the proposed scheme not only reduces computational load at a peer, but also achieves robust streaming in the case of peer failure or packet loss due to adverse wireless channel conditions. We confirm this finding by analyzing the effect of distributed transcoding under peer failure.  相似文献   

5.
6.
Because video streaming over mobile handheld devices has been of great interest, the necessity of introducing new methods with low implementation cost and scalable infrastructures is a strong demand of the service. In particular, these requirements are present in popular wireless networks such as wireless mesh networks (WMN). Peer‐to‐peer (P2P) networks promise an efficient scalable network infrastructure for video streaming over wired and wireless networks. Limited resources of the peers in P2P networks and high error rate in wireless channels make it more challenging to run P2P streaming applications over WMNs. Therefore, it is necessary to design efficient and improved error protection methods in P2P video streaming applications over WMNs. In this paper, we propose a new adaptive unequal video protection method specially intended for large scale P2P video streaming over mobile WMNs. Using this method, different frames have different priorities in receivers along the recovery process. Moreover, we precisely and completely evaluate different aspects related to frame protection in these networks using five important performance metrics including video distortion, late arrival distortion, end‐to‐end delay, overhead and initial start‐up delay. The results obtained from a precise simulation in OMNeT++ show that the proposed adaptive method significantly outperforms other solutions by providing better video quality on mobile wireless nodes. Copyright © 2015 John Wiley & Sons, Ltd.  相似文献   

7.
We design a transport protocol that uses artificial neural networks (ANNs) to adapt the audio transmission rate to changing conditions in a mobile ad hoc network. The response variables of throughput, end-to-end delay, and jitter are examined. For each, statistically significant factors and interactions are identified and used in the ANN design. The efficacy of different ANN topologies are evaluated for their predictive accuracy. The Audio Rate Cognition (ARC) protocol incorporates the ANN topology that appears to be the most effective into the end-points of a (multi-hop) flow, using it to adapt its transmission rate. Compared to competing protocols for media streaming, ARC achieves a significant reduction in packet loss and increased goodput while satisfying the requirements of end-to-end delay and jitter. While the average throughput of ARC is less than that of TFRC, its average goodput is much higher. As a result, ARC transmits higher quality audio, minimizing root mean square and Itakura–Saito spectral distances, as well as several parametric distance measures. In particular, ARC minimizes linear predictive coding cepstral (sic) distance, which closely correlates to subjective audio measures.  相似文献   

8.
Video streaming over wireless links is a challenging issue due to the stringent Quality-of-Service (QoS) requirements of video traffic, the limited wireless channel bandwidth and the broadcast nature of wireless medium. As contention-based or reservation-based (i.e., contention-free) medium access control (MAC) protocols in existing wireless link-layer standards cannot efficiently support multimedia applications such as video streaming, a hybrid approach has been proposed, which uses both contention and reservation-based channel access mechanisms to transmit packets for each video source. Using this content-aware resource management approach, each video source reserves well below its peak data rate, and uses contention-based media access to transmit the remainder of the packets. In this paper, we first propose two conflict avoidance strategies and two buffering architectures for video streaming over ad hoc networks. Considering the interactions of reservation and contention, we develop the analytical model for the saturated traffic case and then extend it to derive tight performance bounds for the unsaturated case. Using the MAC protocols specified in the WiMedia ECMA-368 standard as an example, extensive simulations have been conducted to validate the analysis. Real video traces have been used to examine the video streaming performance. The analytical and simulation results demonstrate the effectiveness and efficiency of the hybrid resource management approach, and also reveal the impact of the conflict avoidance strategy and buffering design on the video performance.  相似文献   

9.
10.
2.5G and 3G cellular networks are becoming more widespread and the need for value added services increases rapidly. One of the key services that operators seek to provide is streaming of rich multimedia content. However, network characteristics make the use of streaming applications very difficult with an unacceptable quality of service (QoS). The 3GPP standardization body has standardized streaming services that will benefit operators and users. There is a need for a mechanism that will enable a good quality multimedia streaming that uses the 3GPP standard. This paper describes an adaptive streaming algorithm that uses the 3GPP standard. It improves significantly the QoS in varying network conditions while monitoring its performance using queueing methodologies. The algorithm utilizes the available buffers on the route of the streaming data in a unique way that guarantees high QoS. The system is analytically modeled: the streaming server, the cellular network and the cellular client are modeled as cascaded buffers and the data is sequentially streamed between them. The proposed Adaptive streaming algorithm (ASA) controls these buffers’ occupancy levels by controlling the transmission and the encoding rates of the streaming server to achieve high QoS for the streaming. It overcomes the inherent fluctuations of the network bandwidth. The algorithm was tested on General Packet Radio Service (GPRS), Enhanced Data rates for GSM Evolution (EDGE) and Universal Mobile Telecommunication System (UMTS) networks. The results showed substantial improvements over other standard streaming methods used today.  相似文献   

11.
In this paper, we present an entire network load‐aware cooperative routing algorithm based on IEEE 802.11 multi‐rate for video streaming over mobile ad hoc networks. The proposed routing algorithm is designed to minimize the consumed time slots while guaranteeing the required time slots at all the pairs of adjacent nodes over the route and the contention neighbors of these nodes to support the route. Furthermore, the proposed routing algorithm can distribute the network loads well over the entire network. This technology is essential because video streaming applications require stringent quality of service and even larger network resources compared with traditional data services, and these demands may dramatically increase the entire network load and/or cause network congestion. Finally, experimental results are provided to show a performance of the proposed routing algorithm. Copyright © 2011 John Wiley & Sons, Ltd.  相似文献   

12.
In this article, an end-to-end quality of service framework for streaming services in 3G mobile networks is considered. Under this scenario, the interaction between UMTS and IETF's protocols and mechanisms for a streaming session is analyzed. By signaling flowcharts, it is shown that both groups of protocols and mechanisms can co-operate to provide seamless end-to-end real-time services. Specifically, the article proposes to make the IP multimedia subsystem aware of the real time streaming protocol, in order to extend its control from SIP to RTSP-based services, such as multimedia streaming services. Supported by this proposed framework, provisioning of audio streaming services over 3G mobile networks is also outlined.  相似文献   

13.
With the development of wireless technologies, video streaming services over heterogeneous wireless networks have become more popular in recent years. Video streaming schemes for heterogeneous networks should consider vertical handover in which the link capacity is varied significantly, because the quality experienced for a video streaming service is affected by the network status. When a vertical handover occurs, an abrupt bandwidth change and substantial handover latency lead to bursty packet loss and discontinuity of the video playback. In this paper, we propose a handover-aware video streaming scheme to provide seamless video streaming services over heterogeneous wireless networks. The proposed scheme adjusts its sending rate and the quality level of the transmitted video streams according to the significant bandwidth variation that occurs in a vertical handover. To expedite the response to the bandwidth variation due to a handover, our scheme uses an explicit notification message that informs the streaming server of a client's handover occurrence. In order to evaluate the performance, we use a simulation environment for a vertical handover between wireless local area networks and cellular networks. Through the simulation results, we prove that our scheme improves the experienced quality of video streaming in vertical handovers.  相似文献   

14.
Dynamic Adaptive Streaming over HTTP (DASH) is the state-of-the-art technology for video streaming and has been widely deployed in both wired and wireless environments. However, mobile DASH users often suffer from video quality oscillation and even video freeze in wireless environments, which results in poor user experience. This is mainly because most quality adaptation algorithms in DASH rely highly on bandwidth estimation to adjust the video quality while wireless network bandwidth is unstable in nature and changes frequently according to wireless channel contention and condition. To provide stable performance, even during severe bandwidth fluctuation, this paper proposes the Wireless Quality Adaptation (WQUAD) algorithm, which eliminates bandwidth estimation from quality adaptation. Thanks to the Scalable Video Codec (SVC), the proposed scheme always prioritizes to lower layers over higher ones as long as the play-out buffer is not completely filled by the lower layers. As a result, the client always fills the buffer with the base layers first and then the upper enhancement layers sequentially. This horizontal adaptation is straightforward and does not require any bandwidth estimation. Through NS-2 simulations, we show that WQUAD achieves (i) stable performance, keeping the video quality level with respect to the long-term network bandwidth, (ii) effective video freeze prevention, and (iii) high video quality on average.  相似文献   

15.
Existing transport layer protocols such as TCP and UDP are designed specifically for point-to-point communication. The increased popularity of peer-to-peer networking has brought changes in the Internet that provided users with potentially multiple replicated sources for content retrieval. However, applications that leverage such parallelism have thus far been limited to non-real-time file downloads. In this article we consider the problem of multipoint-to-point video streaming over peer-to-peer networks. We present a transport layer protocol called R/sup 2/CP that effectively enables real-time multipoint-to-point video streaming. R/sup 2/CP is a receiver-driven multistate transport protocol. It requires no coordination between multiple sources, accommodates flexible application layer reliability semantics, uses TCP-friendly congestion control, and delivers to the video stream the aggregate of the bandwidths available on the individual paths. Simulation results show great performance benefits using R/sup 2/CP in peer-to-peer networks.  相似文献   

16.
Mobile multimedia applications require networks that optimally allocate resources and adapt to dynamically changing environments. Cross-layer design (CLD) is a new paradigm that addresses this challenge by optimizing communication network architectures across traditional layer boundaries. In this article we discuss the relevant technical challenges of CLD and focus on application-driven CLD for video streaming over wireless networks. We propose a cross-layer optimization strategy that jointly optimizes the application layer, data link layer, and physical layer of the protocol stack using an application-oriented objective function in order to maximize user satisfaction. In our experiments we demonstrate the performance gain achievable with this approach. We also explore the trade-off between performance gain and additional computation and communication cost introduced by cross-layer optimization. Finally, we outline future research challenges in CLD.  相似文献   

17.
Due to the ever growing interest of 3-dimensional (3D) technology, together with the the widespread use of new generation smartphones, netbooks, and tablets, the idea to provide 3D multimedia services also to mobile users is attracting the attention of researchers worldwide. The emerging long term evolution-advanced (LTE-A) technology seems to have all the potential to offer such kind of services but no accurate studies have been yet conducted to demonstrate this assumption. To bridge this gap, we investigate herein how the LTE-A network configuration, the traffic load, and 3D video settings may influence to the quality of all services of all active flows. We consider a heterogeneous LTE-A network composed by macro and pico cells and a number of users uniformly distributed into the scenario, that receive at the same time 3D video streams, voice flows and best-effort applications. Moreover, several objective metrics, such as the ratio of lost packets, the peak signal to noise ratio, the Structural SIMilarity, the application goodput, and packet delays, have been evaluated to show the LTE-A effectiveness in delivering 3D videos in a realistic last generation mobile environment.  相似文献   

18.
Video streaming, one of the most popular technologies for online video playback, has already been applied to 4G LTE networks. Previous work has been devoted to understanding the power consumption in general 4G LTE networks, while it is still unclear how the online video streaming makes impact on the power performance of mobile devices. Inspired by this, this paper investigates the relationship between the mobile device’s power performance characteristics and the behaviours of video streaming in 4G LTE networks. There are many natural issues/questions that are clearly interesting and important, while it is non-trivial to answer these issues/questions exactly (e.g., where is the energy saving room? how much is it?). To address a series of issues like the above, we formulate our energy models together with an algorithm that can assist our analysis. Particularly, we design a systematic platform, and conduct a comprehensive and also deep analysis on the power consumption of video streaming in 4G LTE networks. Our experiments reveal us a series of valuable findings—the saving room in the network part is large (from 41.86 to 69.62%), the number of RRC tails and the transmission pattern could be promising for optimizing the power consumption, for example.  相似文献   

19.
Cross-layer design of ad hoc networks for real-time video streaming   总被引:6,自引:0,他引:6  
Cross-layer design breaks away from traditional network design where each layer of the protocol stack operates independently. We explore the potential synergies of exchanging information between different layers to support real-time video streaming. In this new approach information is exchanged between different layers of the protocol stack, and end-to-end performance is optimized by adapting to this information at each protocol layer. We discuss key parameters used in the cross-layer information exchange along with the associated cross-layer adaptation. Substantial performance gains through this cross-layer design are demonstrated for video streaming.  相似文献   

20.
In this paper, we propose a novel solution for the adaptive streaming of 3D representations in the form of multi-view video by utilizing P2P overlay networks to assist the media delivery and minimize the bandwidth requirement at the server side. Adaptation to diverse network conditions is performed regarding the features of human perception to maximize the perceived 3D. We have performed subjective tests to characterize these features and determined the best adaptation method to achieve the highest possible perceived quality. Moreover, we provide a novel method for mapping from scalable video elementary stream to torrent-like data chunks for adaptive video streaming and provide an optimized windowing mechanism that ensures timely delivery of the content over yanl?? gibi. The paper also describes techniques generating scalable video chunks and methods for determining system parameters such as chunk size and window length.  相似文献   

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