A distributed utility‐based scheduling for peer‐to‐peer video streaming over wireless networks

Interactive multimedia applications such as peer‐to‐peer (P2P) video services over the Internet have gained increasing popularity during the past few years. However, the adopted Internet‐based P2P overlay network architecture hides the underlying network topology, assuming that channel quality is always in perfect condition. Because of the time‐varying nature of wireless channels, this hardly meets the user‐perceived video quality requirement when used in wireless environments. Considering the tightly coupled relationship between P2P overlay networks and the underlying networks, we propose a distributed utility‐based scheduling algorithm on the basis of a quality‐driven cross‐layer design framework to jointly optimize the parameters of different network layers to achieve highly improved video quality for P2P video streaming services in wireless networks. In this paper, the quality‐driven P2P scheduling algorithm is formulated into a distributed utility‐based distortion‐delay optimization problem, where the expected video distortion is minimized under the constraint of a given packet playback deadline to select the optimal combination of system parameters residing in different network layers. Specifically, encoding behaviors, network congestion, Automatic Repeat Request/Query (ARQ), and modulation and coding are jointly considered. Then, we provide the algorithmic solution to the formulated problem. The distributed optimization running on each peer node adopted in the proposed scheduling algorithm greatly reduces the computational intensity. Extensive experimental results also demonstrate 4–14 dB quality enhancement in terms of peak signal‐to‐noise ratio by using the proposed scheduling algorithm. Copyright © 2015 John Wiley & Sons, Ltd.     

Extending P2PMesh: topology‐aware schemes for efficient peer‐to‐peer data sharing in wireless mesh networks

Wireless mesh networks (WMNs) have emerged as a promising technology that provides low‐cost broadband access to the Internet for fixed and mobile wireless end users. An orthogonal evolution in computer networking has been the rise of peer‐to‐peer (P2P) applications such as P2P data sharing. It is of interest to enable effective P2P data sharing in this type of networks. Conventional P2P data sharing systems are not cognizant of the underlying network topology and therefore suffer from inefficiency. We argue for dual‐layer mesh network architecture with support from wireless mesh routers for P2P applications. The main contribution of this paper is P2PMesh: a topology‐aware system that provides combined architecture and efficient schemes for enabling efficient P2P data sharing in WMNs. The P2PMesh architecture utilizes three schemes: (i) an efficient content lookup that mitigates traffic load imbalance at mesh routers; (ii) an efficient establishment of download paths; and (iii) a data transfer protocol for multi‐hop wireless networks with limited capacity. We note here that the path establishment and data transfer schemes are specific to P2P traffic and that other traffic would use routes determined by the default routing protocol in the WMN. Simulation results suggest that P2PMesh has the potential to improve the performance of P2P applications in a wireless multi‐hop setting; specifically, we focused on data sharing, but other P2P applications can also be supported by this approach. Copyright © 2011 John Wiley & Sons, Ltd.     

Slack time–based scheduling scheme for live video streaming in P2P network

Nowadays, peer‐to‐peer network plays a significant role in data transfer and communication. The past few years have witnessed considerable growth in this area because of its inherent advantages. Peer‐to‐peer live streaming has a significant impact on video transmission over the Internet. Major factors that influence the performance of P2P live streaming are overlay construction and scheduling strategies. Although, a large number of scheduling schemes are developed but none of them is comprehensive enough to provide solutions to live streaming issues. These suffer from substantial delay and low video quality at the receiver side. In this paper, a new start‐up–based selection procedure and slack time–based scheduling scheme is proposed. The start‐up selection procedure defines the start‐up buffer location for new peer, and the scheduling scheme selects both the chunk and peers. The proposed scheduling scheme uses both push and pull priority–based strategies. The simulation results of the proposed approach demonstrates significant improvement in both the network performance and video quality at the receiver side. It is observed that playback delay, startup delay, and end‐to‐end delay in the network are reduced and quality of the video at receiver side is improved as the distortion and frame loss ratio is decreased.     

Design and Implementation of a Network‐Adaptive Mechanism for HTTP Video Streaming

This paper proposes a network‐adaptive mechanism for HTTP‐based video streaming over wireless/mobile networks. To provide adaptive video streaming over wireless/mobile networks, the proposed mechanism consists of a throughput estimation scheme in the time‐variant wireless network environment and a video rate selection algorithm used to increase the streaming quality. The adaptive video streaming system with proposed modules is implemented using an open source multimedia framework and is validated over emulated wireless/mobile networks. The emulator helps to model and emulate network conditions based on data collected from actual experiments. The experiment results show that the proposed mechanism provides higher video quality than the existing system provides and a rate of video streaming almost void of freezing.     

Performance modeling of FEC‐based unequal error protection for H.264/AVC video streaming over burst‐loss channels

Unequal error protection systems are a popular technique for video streaming. Forward error correction (FEC) is one of error control techniques to improve the quality of video streaming over lossy channels. Moreover, frame‐level FEC techniques have been proposed for video streaming because of different priority video frames within the transmission rate constraint on a Bernoulli channel. However, various communication and storage systems are likely corrupted by bursts of noise in the current wireless behavior. If the burst losses go beyond the protection capacity of FEC, the efficacy of FEC can be degraded. Therefore, our proposed model allows an assessment of the perceived quality of H.264/AVC video streaming over bursty channels, and is validated by simulation experiments on the NS‐2 network simulator at a given estimate of the packet loss ratio and average burst length. The results suggest a useful reference in designing the FEC scheme for video applications, and as the video coding and channel parameters are given, the proposed model can provide a more accurate evaluation tool for video streaming over bursty channels and help to evaluate the impact of FEC performance on different burst‐loss parameters. Copyright © 2014 John Wiley & Sons, Ltd.     

Adaptive cross-layer protection strategies for robust scalable video transmission over 802.11 WLANs

Robust streaming of video over 802.11 wireless local area networks poses many challenges, including coping with bandwidth variations, data losses, and heterogeneity of the receivers. Currently, each network layer (including physical layer, media access control (MAC), transport, and application layers) provides a separate solution to these challenges by providing its own optimized adaptation and protection mechanisms. However, this layered strategy does not always result in an optimal overall performance for the transmission of video. Moreover, certain protection strategies can be implemented simultaneously in several layers and, hence, the optimal choices from the application and complexity perspective need to be identified. In this paper, we evaluate different error control and adaptation mechanisms available in the different layers for robust transmission of video, namely MAC retransmission strategy, application-layer forward error correction, bandwidth-adaptive compression using scalable coding, and adaptive packetization strategies. Subsequently, we propose a novel adaptive cross-layer protection strategy for enhancing the robustness and efficiency of scalable video transmission by performing tradeoffs between throughput, reliability, and delay depending on the channel conditions and application requirements. The results obtained using the proposed adaptive cross-layer protection strategies show a significantly improved visual performance for the transmitted video over a variety of channel conditions.     

End-to-end TCP-friendly streaming protocol and bit allocation for scalable video over wireless Internet

With the convergence of wired-line Internet and mobile wireless networks, as well as the tremendous demand on video applications in mobile wireless Internet, it is essential to an design effective video streaming protocol and resource allocation scheme for video delivery over wireless Internet. Taking both network conditions in the Internet and wireless networks into account, in this paper, we first propose an end-to-end transmission control protocol (TCP)-friendly multimedia streaming protocol for wireless Internet, namely WMSTFP, where only the last hop is wireless. WMSTFP can effectively differentiate erroneous packet losses from congestive losses and filter out the abnormal round-trip time values caused by the highly varying wireless environment. As a result, WMSTFP can achieve higher throughput in wireless Internet and can perform rate adjustment in a smooth and TCP-friendly manner. Based upon WMSTFP, we then propose a novel loss pattern differentiated bit allocation scheme, while applying unequal loss protection for scalable video streaming over wireless Internet. Specifically, a rate-distortion-based bit allocation scheme which considers both the wired and the wireless network status is proposed to minimize the expected end-to-end distortion. The global optimal solution for the bit allocation scheme is obtained by a local search algorithm taking the characteristics of the progressive fine granularity scalable video into account. Analytical and simulation results demonstrate the effectiveness of our proposed schemes.     

Sliding‐window forward error correction using Reed‐Solomon code and unequal error protection for real‐time streaming video

Forward error correction (FEC) techniques are widely used to recover packet losses over unreliable networks in real‐time video streaming applications. Traditional frame‐level FEC encodes 1 video frame in each FEC coding window. By contrast, in the expanding‐window FEC scheme, high‐priority frames are included in the FEC processing of the following frames, so as to construct a larger coding window. In general, expanding‐window FEC improves the recovery performance of FEC, because the high‐priority frame can be protected by multiple windows and the use of a larger coding window increases the efficiency. However, the larger window size also increases the complexity of the coding and the memory space requirements. Consequently, expanding‐window FEC is limited in terms of practical applications. Sliding‐window FEC adopts a fixed window size in order to approximate the performance of the expanding‐window FEC method, but with a reduced complexity. Previous studies on sliding‐window FEC have generally adopted an equal error protection (EEP) mechanism to simplify the analysis. This paper considers the more practical case of an unequal error protection (UEP) strategy. An analytical model is derived for estimating the playable frame rate (PFR) of the proposed sliding‐window FEC scheme with a Reed‐Solomon erasure code for real‐time non‐scalable streaming applications. The analytical model is used to determine the optimal FEC configuration which maximizes the PFR value under given transmission rate constraints. The simulation results show that the proposed sliding‐window scheme achieves almost the same performance as the expanding‐window scheme, but with a significantly lower computational complexity.     

Downlink media streaming with wireless fountain coding in wireline‐cum‐WiFi networks

This paper addresses the problem of streaming packetized media data in a combined wireline/802.11 network. Since the wireless channel is normally the bottleneck for media streaming in such a network, we propose that wireless fountain coding (WFC) be used over the wireless downlink in order to efficiently utilize the wireless bandwidth and exploit the broadcast nature of the channel. Forward error correction (FEC) is also used to combat errors at the application‐layer. We analytically obtain the moment generating function (MGF) for the wireless link‐layer delay incurred by WFC. With the MGF, the expected value of this wireless link‐layer delay is found and used by the access point (AP), who has no knowledge of the buffer contents of wireless receivers, to make a coding‐based decision. We then derive the end‐to‐end packet loss/late probability based on the MGF. We develop an integrated ns‐3/EvalVid simulator to evaluate our proposed system and compare it with the traditional 802.11e scheme which is without WFC capability but equipped with application‐ and link‐layer retransmission mechanisms. Through extensive simulations of video streaming, we show that streaming with WFC is able to support more concurrent video flows compared to the traditional scheme. When the deadlines imposed on video packets are relatively stringent, streaming with WFC also shows superior performance in terms of packet loss/late probability, video distortion, and video frame delay, over the traditional scheme. Copyright © 2011 John Wiley & Sons, Ltd.     

Efficient scalable video multicast based on network-coded communication

In order to increase the efficiency of mobile video transmission in a 5G network, this paper investigates a cooperative multicast of scalable video using network coding with adaptive modulation and coding over dedicated relay-based cellular networks. Different scalable video layers prefer different protection degrees, and user equipments (UEs) in different locations experience different packet loss rates in wireless networks. Guaranteeing that all UEs experience a certain level of video quality is one of the biggest challenges in scalable video multicast. Using the number of satisfied UEs as a metric, the proposed efficient scalable video multicast based on network-coded cooperation (SVM-NC) scheme, combined with adaptive modulation and coding, enhances the attainable system performance under strict time and bandwidth resource constraints for guaranteed smooth playback. Various simulations were performed for performance evaluation. The proposed scheme ensures that the expected percentage of satisfied UEs approximately achieves the maximum number of UEs in a multicast group by using network-coded cooperation over dedicated relay-based cellular networks. In addition, the peak signal-to-noise ratio metric is asymptotic to the maximum performance of high-resolution video quality offered by service providers.     

Improved video streaming using MSVC and nonoverlapping zone routing multipath propagation over MANETs

Providing real‐time video streaming in mobile ad hoc networks is difficult because of the time‐dependent channel status and stringent service requirements. The currently existing route request‐reply–based multihop overlay networks cause considerable control overheads in video transmission resulting in loss of data and communication breakdown. Such networks are more suitable to nonstreaming video applications rather than to time‐sensitive video streaming applications. Therefore, a powerful mechanism needs to be adopted to handle the channel failures amicably and reduce latency effectively in time critical video streaming applications over mobile ad hoc networks. In order to be resilient to the channel failures and reduce latency in such applications, 2 strategies, namely, multistate video coding and 2‐tier–based nonoverlapping zone routing multipath propagation through directional antennas have respectively been incorporated. The performance of the proposed nonoverlapping zone routing multipath propagation system is compared with those of the existing multicast zone routing and zone‐based hierarchical link state routing protocols with parameters average end‐to‐end delay, routing overhead and packet delivery ratio using NS 2.34. The simulation results show that latency and resilience get considerably improved. Finally, the video quality of the proposed work has been verified by subjective and objective video testing methods.     

一种信道自适应的无线视频流差错控制机制

提出了一种自适应的无线视频流差错保护方法。该方法充分考虑MPEG-4FGS码流细粒度可扩展的特性,根据码流的重要性程度采用不等的保护措施,同时能够自适应地根据无线信道的具体状况,将信道带宽在MPEG-4FGS信源编码速率和信道编码速率之间进行最优化分配,使得接收端能获得最佳的重建视频质量。实验结果表明,在各种不同信道状况下,与均等错误保护和固定的不等错误保护方法相比,该方法均可获得更好的性能。     

Rate-distortion based real-time wireless video streaming

This paper presents wireless video streaming techniques that exploit the characteristics of video content, transmission history, and physical layer channels to enable real-time efficient video streaming over wireless networks to a wireless client. The key contribution of the proposed video streaming techniques is the use of rate-distortion based, but simplified, low complexity packet scheduling as well as forward error correction (FEC) rate selection. To this end, we develop an optimization framework that jointly schedules the packets and selects the FEC rates. The rate-distortion optimized packet scheduling and FEC rate selection provides the optimum quality video on the receiver side albeit at a high computational cost. By some intelligent approximations, rate distortion optimized packet scheduling and FEC rate selection technique is transformed into two sub-optimal but low complexity video streaming techniques that can provide high video quality. We perform extensive simulations to understand the performance of our proposed techniques under different scenarios. Results show that, the proposed techniques improve video quality on the average by 4 dB. We conclude that significant benefits to end-user experience can be obtained by using such video streaming methods.     

Managing QoS requirements for video streaming: from intra‐node to inter‐node

Streaming video over IP networks has become increasingly popular; however, compared to traditional data traffic, video streaming places different demands on quality of service (QoS) in a network, particularly in terms of delay, delay variation, and data loss. In response to the QoS demands of video applications, network techniques have been proposed to provide QoS within a network. Unfortunately, while efficient from a network perspective, most existing solutions have not provided end‐to‐end QoS that is satisfactory to users. In this paper, packet scheduling and end‐to‐end QoS distribution schemes are proposed to address this issue. The design and implementation of the two schemes are based on the active networking paradigm. In active networks, routers can perform user‐driven computation when forwarding packets, rather than just simple storing and forwarding packets, as in traditional networks. Both schemes thus take advantage of the capability of active networks enabling routers to adapt to the content of transmitted data and the QoS requirements of video users. In other words, packet scheduling at routers considers the correlation between video characteristics, available local resources and the resulting visual quality. The proposed QoS distribution scheme performs inter‐node adaptation, dynamically adjusting local loss constraints in response to network conditions in order to satisfy the end‐to‐end loss requirements. An active network‐based simulation shows that using QoS distribution and packet scheduling together increases the probability of meeting end‐to‐end QoS requirements of networked video. Copyright © 2005 John Wiley & Sons, Ltd.     

A retransmission‐based scheme for video streaming over wireless channels

Efficient resource allocation is a key factor to improve the efficiency of video transmission over wireless channels. To increase the number of correctly received video frames at the decoder, it is desirable to reduce the video source rate while increasing error protection when the wireless channel is anticipated to be bad or when the receiver buffer is approaching starvation. In this study, we introduce a retransmission‐based adaptive source‐channel rate control scheme for video transmission over wireless packet networks. In this scheme, the level of adaptiveness is optimized to reduce the bandwidth requirement while guaranteeing delay and loss bounds. The proposed scheme has the advantage of providing closed‐form expressions of the near‐optimum parameters of the proposed model, which are then fed back to the transmitter to scale both the source and channel rates adaptively. Simulation and numerical investigations are carried out to verify the adequacy of the analysis and study the impact of the adaptive process on the continuity of the video playback process. Copyright © 2009 John Wiley & Sons, Ltd.     

Cross-layer architecture for scalable video transmission in wireless network

Multimedia applications such as video conference, digital video broadcasting (DVB), and streaming video and audio have been gaining popularity during last years and the trend has been to allocate these services more and more also on mobile users. The demand of quality of service (QoS) for multimedia raises huge challenges on the network design, not only concerning the physical bandwidth but also the protocol design and services. One of the goals for system design is to provide efficient solutions for adaptive multimedia transmission over different access networks in all-IP environment. The joint source and channel coding (JSCC/D) approach has already given promising results in optimizing multimedia transmission. However, in practice, arranging the required control mechanism and delivering the required side information through network and protocol stack have caused problems and quite often the impact of network has been neglected in studies. In this paper we propose efficient cross-layer communication methods and protocol architecture in order to transmit the control information and to optimize the multimedia transmission over wireless and wired IP networks. We also apply this architecture to the more specific case of streaming of scalable video streams. Scalable video coding has been an active research topic recently and it offers simple and flexible solutions for video transmission over heterogeneous networks to heterogeneous terminals. In addition it provides easy adaptation to varying transmission conditions. In this paper we illustrate how scalable video transmission can be improved with efficient use of the proposed cross-layer design, adaptation mechanisms and control information.     

Scalable 3D video streaming over P2P networks with playback length changeable chunk segmentation

3D video distribution over P2P networks has been thought as a promising way for 3D video entering home. The convergence of scalable 3D video coding and P2P streaming can provide diverse 3D experiences for heterogeneous clients with high distribution efficiencies. However, the conventional chunk segmentation and scheduling algorithms originally aiming at the non-scalable 2D video streaming are not very efficient for scalable 3D video streaming over P2P networks due to the particular data characteristics of scalable 3D video. Based on this motivation, this paper first presents a playback length changeable 3D video chunk segmentation (PLC3DCS) algorithm to provide different error resilience strengths to video and depth as well as layers with different importance levels in the 3D video transmission. Then, a hybrid-priority based chunk scheduling (HPS) algorithm is proposed to be tied in with the proposed chunk segmentation algorithm to further promote the overall 3D video P2P streaming performance. The simulation results show that the proposed PLC3DCS algorithm with the corresponding HPS can increase the success delivery rates of chunks with more important levels, and further improve the user’s quality of 3D experience.     

Scalable video coding and transport over broadband wirelessnetworks

With the emergence of broadband wireless networks and increasing demand of multimedia information on the Internet, wireless multimedia services are foreseen to become widely deployed in the next decade. Real-time video transmission typically has requirements on quality of service (QoS). However, wireless channels are unreliable and the channel bandwidth varies with time, which may cause severe degradation in video quality. In addition, for video multicast, the heterogeneity of receivers makes it difficult to achieve efficiency and flexibility. To address these issues, three techniques, namely, scalable video coding, network-aware adaptation of end systems, and adaptive QoS support from networks, have been developed. This paper unifies the three techniques and presents an adaptive framework, which specifically addresses video transport over wireless networks. The adaptive framework consists of three basic components: (1) scalable video representations; (2) network-aware end systems; and (3) adaptive services. Under this framework, as wireless channel conditions change, mobile terminals and network elements can scale the video streams and transport the scaled video streams to receivers with a smooth change of perceptual quality. The key advantages of the adaptive framework are: (1) perceptual quality is changed gracefully during periods of QoS fluctuations and hand-offs; and (2) the resources are shared in a fair manner     

A reliable overlay video transport protocol for multicast agents in wireless mesh networks

A new video transport protocol for multicast agents in wireless mesh networks (WMNs) is proposed in this paper. The proposed protocol enables a significant reduction in the transmission overhead, while providing reliable communication for its use in multicast applications. This proposed reliable protocol provides a practical approach for an overlay peer‐to‐peer multicast facility supported within the application layer. This obviates the need to give upgraded routers capable of handling multicast broadcasting or modify the existing protocol stack. The protocol tolerates partial losses in multimedia transmissions, while supporting control of the delay sensitivity of such transmissions in WMNs. The key issue in this protocol is the ability to detect packet loss, anticipate retransmission requests, and use the anticipated retransmission requests to transmit the lost packets prior to requests from other receiving agents. The proposed protocol allows for the receiver to determine if retransmission of lost packets is required, ensuring the greatest flexibility needed for a reliable multicast protocol. Copyright © 2009 John Wiley & Sons, Ltd.     

Frame distortion estimation for unequal error protection methods in scalable video coding (SVC)

In many multimedia applications, coded video is transmitted over error prone heterogeneous networks. Because of the predictive mechanism used in video coding, transmission error would propagate temporally and spatially and would result in significant quality losses. In order to address this problem, different error resilience methods have been proposed. One of the techniques, which is commonly used in video streaming, is unequal error protection (UEP) of scalable video coding (SVC). In this technique, different independent layers of an SVC stream are protected differently and based on their importance by using forward error correction (FEC) codes. Accurately analyzing the importance or utility of each video part is a critical component and would lead to a better protection and higher quality of the received video. Calculation of the utility is usually based on multiple decoding of sub-bitstreams and is highly computationally complex. In this work, we propose an accurate low complexity utility estimation technique that can be used in different applications. This technique estimates the utility of each network abstraction layer (NAL) by considering the error propagation to future frames. We utilize this method in an UEP framework with the scalable extension of H.264/AVC codec and it achieves almost the same performance as highly complex estimation techniques (an average loss of 0.05 dB). Furthermore, we propose a low delay version of this technique that can be used in delay constrained application. The estimation accuracy and performance of our proposed technique are studied extensively.