Adaptive low-latency peer-to-peer streaming and its application

Peer-to-peer (P2P) streaming is emerging as a viable communications paradigm. Recent research has focused on building efficient and optimal overlay multicast trees at the application level. A few commercial products are being implemented to provide voice services through P2P streaming platforms. However, even though many P2P protocols from the research community claim to be able to support large scale low-latency streaming, none of them have been adopted by a commercial voice system so far. This gap between advanced research prototypes and commercial implementations shows that there is a lack of a practical and scalable P2P system design that can provide low-latency service in a real implementation. After analyzing existing P2P system designs, we found two important issues that could lead to improvements. First, many existing designs that aim to build a low-latency streaming platform often make the unreasonable assumption that the processing time involved at each node is zero. However in a real implementation, these delays can add up to a significant amount of time after just a few overlay hops and make interactive applications difficult. Second, scant attention has been paid to the fact that even in a conversation involving a large number of users, only a few of the users are actually actively speaking at a given time. We term these users, who have more critical demands for low-latency, active users. In this paper, we detail the design of a novel peer-to-peer streaming architecture called ACTIVE. We then present a complete commercial scale voice chat system called AudioPeer that is powered by the ACTIVE protocol. The ACTIVE system significantly reduces the end-to-end delay experienced among active users while at the same time being capable of providing streaming services to very large multicast groups. ACTIVE uses realistic processing assumptions at each node and dynamically optimizes the streaming structure while the group of active users changes over time. Consequently, it provides virtually all users with the low-latency service that before was only possible with a centralized approach. We present results from both simulations and our real implementation, which clearly show that our ACTIVE system is a feasible approach to scalable, low-latency P2P streaming.     

Video streaming over the internet with optimal bandwidth resource allocation

In this paper, an adaptive framework for video streaming over the Internet is presented. The framework is a joint design of packet scheduling and rate control with optimal bandwidth resource allocation. The transmission rate is dynamically adjusted to obtain maximal utilization of the client buffer and minimal allocation of the bandwidth. Under the constraint of the transmission rate, a prioritized packet scheduling is designed to provide a better visual quality of video frames. The packet scheduling is a refined bandwidth allocation which takes into account of varying importance of the different packets in a compressed video stream. Moreover, the proposed approach is scalable with increasing multimedia flows in the distributed Internet environment. Comparisons are made with the most current streaming approaches to evaluate the performance of the framework using the H.264 video codec. The extensive simulation results show that the average Peak Signal to Noise Ratio (PSNR) increases in our proposed approach. It provides a better quality of the decoded frames, and the quality of the decoded frames changes more smoothly. The achieved video quality among different users also has a lower fluctuation, which indicates a fair sharing of network resources.

Storage technique for real-time streaming of layered video

Scalable streaming technology has been proposed to effectively support heterogeneous devices with dynamically varying bandwidth. From the file system’s point of view, scalable streaming introduces another dimension of complexity in disk scheduling. Most of the existing efforts on multimedia file systems are dedicated to I/O scheduling algorithm and data placement scheme that efficiently guarantee I/O bandwidth. The important underlying assumption in these efforts is that most of the multimedia file accesses are simple playback operations and therefore are sequential. However, this workload characteristic is not valid in scalable streaming environment. In a scalable streaming environment, i.e., when only a subset of imagery is retrieved, the playback does not necessarily coincide with the sequential access on the file. The current file structure and the file system organization leaves much to be desired for supporting scalable streaming service. In this work, we propose a file system scheme, Harmonic Placement to efficiently support scalable streaming. The basic idea of Harmonic placement is to cluster the frequently accessed layers together to avoid unnecessary disk seeks. The data blocks are partitioned into two sets with respect to the layers: lower layers and upper layers. In Harmonic placement, the data blocks in the lower layers are placed with respect to their frame sequence and the data blocks in the upper layers are clustered according to the layers they belong to. We develop elaborate performance models for three different file system schemes: Progressive placement, Interleaved Placement and Harmonic Placement. We investigate the performance of the file server with different file system schemes. It was found that file system performance is very sensitive to the file organization scheme. When most of the service requests are for low-quality video (e.g., 128 Kbits/s ISDN), Progressive placement scheme supports twice as many sessions as the Interleaved placement scheme. When most of the service requests are for high-quality video (e.g., 1.5 Mbits/s MPEG-2 DVD quality), Interleaved placement can support twice as many requests as Progressive placement. In both cases, Harmonic placement scheme yields the most promising performance. Primitive version of this work has appeared on Proceedings of NOSSDAV ’06, Providence, Rhode Island, USA. This work is in part funded by KOSEF throught National Research Lab (ROA-2007-000-200114-0) and by HY-SDR center at Hanyang University.     

视频实时传输中的速率控制研究

视频在互联网上的实时传输常因其要求高带宽、低延迟而造成网络拥塞。传统TCP基于窗口的拥塞控制已经不适用于实时传输中的拥塞控制,该文讨论了基于速率的网络拥塞控制方法,它采用自适应码率和分层的视频编码技术,详细说明了基于源端和收端的各种速率控制方法。     

CollectCast: A peer-to-peer service for media streaming

We present CollectCast, a peer-to-peer (P2P) service for media streaming where a receiver peer is served by multiple sender peers. CollectCast operates at the application level but infers underlying network properties to correlate end-to-end connections between peers. The salient features of CollectCast include: (1) a novel multisender selection method that exploits the performance correlation and dependency among connections between different candidate senders and the receiver, (2) a customization of network tomography techniques and demonstration of improved practicality and efficiency, and (3) an aggregation-based P2P streaming mechanism that sustains receiver-side quality in the presence of sender/network dynamics and degradation. We have performed both real-world (on PlanetLab) and simulation evaluation of CollectCast. Our simulation results show that for a receiver, CollectCast makes better selection of multiple senders than other methods that do not infer underlying network properties. Our PlanetLab experiments are performed using a P2P media streaming application (called PROMISE) which we developed on top of CollectCast. Both packet-level and frame-level performance of MPEG-4 video streaming demonstrates the practicality and effectiveness of CollectCast.     

Transmission of layered video streaming via multi-path on ad hoc networks

Mobile ad hoc networks without centralized infrastructure change their topology rapidly because of node mobility, making multimedia applications difficult to run across wireless networks. Moreover, video transmission over ad hoc networks causes frequent transmission loss of video packets owing to end-to-end transmission with a number of wireless links, and requires essential bandwidth and restricted delay to provide quality-guaranteed display. This paper presents an architecture supporting transmission of multiple video streams in ad hoc networks by establishing multiple routing paths to provide extra video coding and transport schemes. This study also proposes an on-demand multicast routing protocol to transport layered video streams. The multicast routing protocol transmits layered video streaming based on a weight criterion, which is derived according to the number of receivers, delay and expiration time of a route. A simulation is performed herein to indicate the viability and performance of the proposed approach. The simulation results demonstrate that the proposed transport scheme is more effective than other video transport schemes with single or multiple paths.

Analysis of a CDN–P2P hybrid architecture for cost-effective streaming media distribution

To distribute video and audio data in real-time streaming mode, two different technologies – Content Distribution Network (CDN) and Peer-to-Peer (P2P) – have been proposed. However, both technologies have their own limitations: CDN servers are expensive to deploy and maintain, and consequently incur a cost for media providers and/or clients for server capacity reservation. On the other hand, a P2P-based architecture requires sufficient number of seed supplying peers to jumpstart the distribution process. Compared with a CDN server, a peer usually offers much lower out-bound streaming rate and hence multiple peers must jointly stream a media data to a requesting peer. Furthermore, it is not clear how to determine how much a peer should contribute back to the system after receiving the media data, in order to sustain the overall media distribution capacity.In this paper, we propose and analyze a novel hybrid architecture that integrates both CDN- and P2P-based streaming media distribution. The architecture is highly cost-effective: it significantly lowers the cost of CDN capacity reservation, without compromising the media quality delivered. In particular, we propose and compare different limited contribution policies for peers that request a media data, so that the streaming capacity of each peer can be exploited on a fair and limited basis. We present: (1) in-depth analysis of the proposed architecture under different contribution policies, and (2) extensive simulation results which validate the analysis. Our analytical and simulation results form a rigorous basis for the planning and dimensioning of the hybrid architecture.     

Towards efficient video chunk dissemination in peer-to-peer live streaming

There are substantial differences in chunk dissemination manner between P2P live streaming and BitTorrent, and inappropriate algorithms will result in inefficiency of live streaming systems. In this paper, we study the chunk dissemination of P2P live streaming, and introduce a discrete and slotted mathematical model to analyze chunk selection algorithms, including rarest first algorithm and greedy algorithm. Moreover, we present a performance metric to evaluate chunk selection algorithms, as well as the optimization function for the exploration of chunk dissemination strategies. We point out the causes of poor performance of these algorithms, and propose a service request randomization mechanism to promote the use of peer resources, which can prevent chunk requests from rendezvous on a few of peers. Simultaneously, we employ weight assignment strategies to avoid excessive requests for rare chunks. Besides, we present an enhanced model, which adds node degree constraint, to improve our model. We revisit the chunk selection algorithms based on the enhanced model. The results of simulation experiments validate our theoretical analysis and indicate that the weighted randomization mechanism is resilient to flash crowd and peer churn, and can improve the performance of P2P live streaming.     

Quality adaptive end-to-end packet scheduling to avoid playout interruptions in Internet video streaming systems

In Internet multimedia streaming, the quality of the delivered media can be adapted to the Quality of Service provided by the underlying network, thanks to encoding algorithms. These allow a fine grained enhancement of a low quality base layer at streaming time. The main objective that should be satisfied in such systems is to avoid the starvation of the decoding process and consequent playout interruptions. In this work, we tackle the problem using a control theoretic approach. In particular, we design and implement the novel end-to-end Quality Adaptive Scheduler for properly distributing the network available bandwidth among base and enhancement layers. The developed solution can be adopted in many contexts given that it has been designed without assumptions on the delivered media nor on the protocol stack. Anyway, to test its effectiveness, we have casted it in a H.264/AVC SVC based video streaming architecture for unicast Internet applications. The performance of the scheduler has been experimentally evaluated in both a controlled testbed and several “wild” Internet scenarios, including also UMTS and satellite radio links. Results have clearly demonstrated that our Quality Adaptive Scheduler is able to significantly improve the performance of the video streaming system in all operative conditions.     

多媒体直播课堂软件系统的设计

针对目前多媒体教室的资源不能被远程教育的学生有效利用的现象,结合音频、视频和屏幕图像压缩算法的压缩比率不断提高的特点,提出了一种多媒体直播课堂软件系统,给出了其系统框架,并详细介绍了该系统的Web服务器、编码服务器、直播服务器、流媒体代理服务器和客户端的设计.该系统与类似系统相比,不受电子讲稿种类多样性的限制,可适应更多的多媒体压缩算法,在传输效率上也有一定的优势.     

视频流自适应传输技术研究

对当前视频流技术所采取的各种压缩算法及标准、传输协议、拥塞控制等技术进行了总结与分析,在此基础上对IP网络上的视频流自适应传输技术进行了研究,建立了实现该技术的框架。此技术框架综合了实时的视频自适应编码技术和有效的带宽自适应传输技术。实验结果表明,在对视频流的实时性要求高的应用上具有较突出的优势。     

自适应流媒体传输方案研究及其应用

提出了一种自适应流媒体传输方案，根据网络状况调整视频数据的发送速率和视频流的码率，以保证视频数据顺利及时的传输到客户端，使客户端能够顺利进行高质量的媒体播放。该方案涵盖了码率整形、拥塞控制和视频质量自适应等功能模块及其相互关系，各功能模块可以依据应用需求选择算法加以具体实现，从而提高了方案的可扩展性和灵活性。所以该方案不依赖于具体的视频编码标准和网络协议，适用范围较广。文中还基于实际流媒体服务器给出了该方案的一个实际应用，点播测试实验汪明该方案是有效的。     

视频通信的网络自适应传输控制技术

针对视频传输中的拥塞控制问题,对网络自适应传输控制技术进行了研究,该技术综合了UDP与TCP的传输特性,能够对网络拥塞进行自适应控制。提出了该技术的技术框架和实现方案,并将其应用在一个端到端的视频传输系统中。实验结果表明,网络自适应传输控制技术可以为视频通信提供良好的传输质量保障。     

流媒体自适应传输策略

本文阐述了在互联网环境下进行实时流媒体传输的技术以及面临的困难,阐述了AIMD算法的特点,并且改进了该算法使之更适应于流媒体传输。     

自适应的分段代理缓存传输策略

结合现有的代理缓存策略和传输方案,针对现有的网络条件,提出了一种自适应的分段方法,解决了已有方法对于流媒体对象流行性的变化和用户访问模式的不确定缺乏自身调整能力的缺欠,和一种优化的传输方案,采用了单播和多播相结合, 主动预取和补丁传输相结合的方法,对于缩短启动延时、提高字节命中率以及节省骨干网带宽等方面取得了较明显的效果.     

A P4P-integrated data-driven P2P system for the live multimedia streaming service

The Proactive network Provider Participation for the P2P (P4P) architecture deploys central servers, which perceives network status and provides peering suggestions to P2P systems in order to achieve better network resource utilization while supporting best possible application performance. However, P4P alone may not be able to make appropriate peering suggestions for live multimedia streaming since it does not include mechanisms to reflect some of the parameters that are important to the QoS of live multimedia streaming such as upload bandwidth and stability of a peer as a stream deliverer. Furthermore, peer synchronization and parent replacement in the middle of a session, which are critical issues to the QoS of live multimedia streaming, are also left as the matters to be dealt with by the P2P systems alone. Most of the existing data-driven P2P systems leverage periodic information exchanges among neighboring peers in order to cope with these problems, which may incur long delay and high control overhead. In this paper, we proposed P4P-integrated data-driven P2P system for live multimedia streaming service. The proposed system includes not only the peering suggestion mechanism appropriate for live multimedia streaming but also the peer synchronization and parent replacement mechanisms, which exploit the centralized P4P framework and do not require periodic control information exchanges. We implemented the system in NS-2 simulator and compared its performance to the P4P and existing data-driven P2P systems. The results from experiments show that the proposed system enhances QoS compared to the existing data-driven P2P systems while maintaining the same level of network efficiency of the original P4P.     

基于MPEG4的自适应实时流媒体传输

论文阐述了在互联网环境下进行实时流媒体传输中存在的一些难点,并且给出了一个基于MPEG4的实时流媒体传输系统。并且希望能充分利用网络带宽,并且尽力减少媒体流由于网络时延和丢包所产生的影响。同时也关注由于丢包而引起的回放质量的下降问题。     

Proxy-assisted periodic broadcast for video streaming with multiple servers

Large scale video streaming over the Internet requires a large amount of resources such as server I/O bandwidth and network bandwidth. A number of video delivery techniques can be used to lower these requirements. Periodic broadcast by a central server combined with proxy caching offers a significant reduction of the aggregate network and server I/O bandwidth usage. However, the resources available to a single server are still limited. In this paper we propose a system with multiple geographically distributed servers. The problem of multiple servers for periodic broadcast is quite different from the problem of object location for multiple web servers. Multiple servers offer increased amount of resources and service availability and may potentially allow a further reduction of network bandwidth usage. On the other hand, the benefit of periodic broadcast mostly comes from high demand videos. With multiple servers holding a video, the demand of the video at each server is reduced. Therefore, it is a challenge to use multiple servers efficiently. We first analyze the dependence of the resource requirements on the number and locations of the servers. Based on the character of the function describing such a dependence, we formulate and solve the problem of video location and delivery, in a way that minimizes resource usage. We explore a trade-off between network and I/O bandwidth requirements. We evaluate our proposed solutions through a number of tests.