Multipath routing for video delivery over bandwidth-limited networks

The delivery of quality video service often requires high bandwidth with low delay or cost in network transmission. Current routing protocols such as those used in the Internet are mainly based on the single-path approach (e.g., the shortest-path routing). This approach cannot meet the end-to-end bandwidth requirement when the video is streamed over bandwidth-limited networks. In order to overcome this limitation, we propose multipath routing, where the video takes multiple paths to reach its destination(s), thereby increasing the aggregate throughput. We consider both unicast (point-to-point) and multicast scenarios. For unicast, we present an efficient multipath heuristic (of complexity O(|V|/sup 3/)), which achieves high bandwidth with low delay. Given a set of path lengths, we then present and prove a simple data scheduling algorithm as implemented at the server, which achieves the theoretical minimum end-to-end delay. For a network with unit-capacity links, the algorithm, when combined with disjoint-path routing, offers an exact and efficient solution to meet a bandwidth requirement with minimum delay. For multicast, we study the construction of multiple trees for layered video to satisfy the user bandwidth requirements. We propose two efficient heuristics on how such trees can be constructed so as to minimize the cost of their aggregation subject to a delay constraint.     

Continuous multicast push of Web documents over the Internet

The success of the World Wide Web has led to a steep increase in the user population and the amount of traffic on the Internet. Popular Web pages create “hot spots” of network load due to their great demand for bandwidth and increase the response time because of the overload on the Web servers. We propose the distribution of very popular and frequently changing Web documents using continuous multicast push. The benefits of CMP in the case of such documents are a very efficient use of network resources, a reduction of the load on the server, lower response times, and scalability for an increasing number of receivers. We present a quantitative evaluation of the continuous multicast push for a wide range of parameters     

基于多时间服务器的时控性加密体制研究

时控性加密(TRE)是一种被称为“向未来发送消息”的密码原语,接收方在未来指定时间之前无法解密密文。目前,大部分TRE方案采用非交互式单时间服务器方法,系统用户能够正常解密,依赖于单一时间服务器在预定解密时间计算并广播的时间陷门。如果单一的时间服务器遭受攻击,或被腐败,则容易直接威胁TRE的安全应用。因此,需要将1个时间服务器“分散”成多个。但已有多时间服务器TRE方案既没有给出安全性分析,也没有给出严格的安全性证明。为此,该文给出一种随机预言机模型下基于双线性迪菲·赫尔曼(BDH)问题的多时间服务器的TRE模型MTSTRE,构造出一种可证明安全的具体和通用方案,并严格证明所提具体方案在自适应选择明文攻击下是安全的。效率分析表明,与已有最有效的多时间服务器TRE解决方案相比,所提具体方案的计算效率也略有提高。     

Adaptive chaining scheme for distributed VOD applications

In distributed video on demand (VOD) applications, a client station buffers a shifting window of its displaying video so that the video stream can be chained from the client to another one arriving within the window, instead of consuming a server stream for each new request. This scheme is called video chaining that can reduce the load of video servers significantly. In this paper, we propose a novel adaptive chaining scheme that extends the basic chaining scheme with two new techniques: two-way bridging and multicast chaining. The two-way bridging method employs video buffers as forward and/or backward bridges to extend each video chain as long as possible. It provides nearly twice the performance gain than basic chaining in terms of server I/O load reduction. Mathematical analyses for both schemes are also given. The multicast chaining method maximizes the multicast degree of each video chain so that lower data delivery cost per video session can be achieved. Our scheme maintains the video chains optimally to shift the load to active clients so that the I/O bottleneck of video servers is released     

Upstream bandwidth optimization of thin client protocols through latency‐aware adaptive user event buffering

Thin client computing trades local processing for network bandwidth by off‐loading application logic to remote servers. User input and display updates are exchanged between client and server through a thin client protocol. In a wireless device context, it is important to achieve bandwidth efficient thin client protocols because bandwidth availability is limited and the power consumption of the wireless network interface card is directly related to the amount of data that is sent and received. This paper presents and evaluates a novel client‐based mechanism which is transparent to the server to reduce upstream bandwidth consumption. Typically, thin client protocols encode user input as a series of small packets, resulting in a major packetization overhead. By buffering user events at the thin client protocol layer, this overhead can be reduced. However, buffering strategies might result in increased response delays for the user. Therefore, models of the upstream bandwidth and the user perceived responsiveness of pull thin client protocols are presented and validated. These models are used in an adaptive framework, which determines the appropriate buffering time to minimize the bandwidth as much as possible without degrading the responsiveness. For lower network roundtrip times and users actively generating input, it is shown how bandwidth savings up to 78% can be achieved while keeping the average user perceived responsiveness below 150 ms. Copyright © 2010 John Wiley & Sons, Ltd.     

Dynamic agent-based hierarchical multicast for wireless mesh networks

We propose and analyze a multicast algorithm named Dynamic Agent-based Hierarchical Multicast (DAHM) for wireless mesh networks that supports user mobility and dynamic group membership. The objective of DAHM is to minimize the overall network cost incurred. DAHM dynamically selects multicast routers serving as multicast agents for integrated mobility and multicast service management, effectively combining backbone multicast routing and local unicast routing into an integrated algorithm. As the name suggests, DAHM employs a two-level hierarchical multicast structure. At the upper level is a backbone multicast tree consisting of mesh routers with multicast agents being the leaves. At the lower level, each multicast agent services those multicast group members within its service region. A multicast group member changes its multicast agent when it moves out of the service region of the current multicast agent. The optimal service region size of a multicast agent is a critical system parameter. We propose a model-based approach to dynamically determine the optimal service region size that achieves network cost minimization. Through a comparative performance study, we show that DAHM significantly outperforms two existing baseline multicast algorithms based on multicast tree structures with dynamic updates upon member movement and group membership changes.     

A Trusted Key Management Scheme for Digital Rights Management

In this paper, we propose a key management scheme which can provide delivery of the key used to encrypt a digital content from the package server to digital rights management (DRM) clients in a secure manner. The proposed scheme can protect digital content from attacks since an encrypted digital content is sent by a package server and only DRM clients can decrypt the encrypted digital content. It protects the key not only from purchasers but also among the other principals who manage the distribution and license servers.     

Network tomography from measured end-to-end delay covariance

End-to-end measurement is a common tool for network performance diagnosis, primarily because it can reflect user experience and typically requires minimal support from intervening network elements. However, pinpointing the site of performance degradation from end-to-end measurements is a challenging problem. We show how end-to-end delay measurements of multicast traffic can be used to infer the under-lying logical multicast tree and the packet delay variance on each of its links. The method does not depend on cooperation from intervening network elements; multicast probing is bandwidth efficient. We establish desirable statistical properties of the estimator, namely consistency and asymptotic normality. We evaluate the approach through simulations, and analyze its failure modes and their probabilities.     

Resource-efficient delivery of on-demand streaming data using UEP codes

We propose and analyze a new multicast scheme for delivering on-demand streaming data using unequal protection codes. The scheme allows an end user to join only one multicast channel for a data stream at any time to play out the requested data stream from its beginning after a fixed initial playout delay. The scheme tolerates packet loss during transmission, and thus, significantly reduces the cost of implementing a reliable multicast network layer to ensure delivery of all packets. Meanwhile, resource usage of the scheme, including server computing bandwidth, network bandwidth, and client's buffer space, is determined only by the original data stream length and the initial playout delay, but is independent of either the number or the arrival pattern of individual end-user requests. Thus, the scheme is totally scalable with the number of end users, fully utilizing the data delivery efficiency of a multicast network. The scheme also uses resources efficiently, e.g., with an initial playout delays of 30 s and 60 s, multicasting a 2 h video using this scheme needs only about 5.5 and 4.8 times, respectively, the server computing bandwidth and network bandwidth of those for a single unicast delivery of the same original data stream.     

Video staging: a proxy-server-based approach to end-to-end videodelivery over wide-area networks

Real-time distribution of stored video over wide-area networks (WANs) is a crucial component of many emerging distributed multimedia applications. The heterogeneity in the underlying network environments is an important factor that must be taken into consideration when designing an end-to-end video delivery system. We present a novel approach to the problem of end-to-end video delivery over WANs using proxy servers situated between local-area networks (LANs) and a backbone WAN. A major objective of our approach is to reduce the backbone WAN bandwidth requirement. Toward this end, we develop an effective video delivery technique called video staging via intelligent utilization of the disk bandwidth and storage space available at proxy servers. Using this video staging technique, only part of a video stream is retrieved directly from the central video server across the backbone WAN whereas the rest of the video stream is delivered to users locally from proxy servers attached to the LANs. In this manner, the WAN bandwidth requirement can be significantly reduced, particularly when a large number of users from the same LAN access the video data. We design several video staging methods and evaluate their effectiveness in trading the disk bandwidth of a proxy server for the backbone WAN bandwidth. We also develop two heuristic algorithms to solve the problem of designing a multiple video staging scheme for a proxy server with a given video access profile of a LAN. Our results demonstrate that the proposed proxy-server-based approach provides an effective and scalable solution to the problem of the end-to-end video delivery over WANs     

ATM based multimedia servers

Next-generation multiprocessors will be deployed as servers in a multimedia environment. Current servers cannot handle multimedia traffic internally and effectively deliver the network's high bandwidth to the processing and storage subsystems. Three proposed scalable, subsystem-based, multimedia server architectures use ATM to tackle this problem     

Reliable Multimedia Transmission Over Cognitive Radio Networks Using Fountain Codes

With the explosive growth of wireless multimedia applications over the wireless Internet in recent years, the demand for radio spectral resources has increased significantly. In order to meet the quality of service, delay, and large bandwidth requirements, various techniques such as source and channel coding, distributed streaming, multicast etc. have been considered. In this paper, we propose a technique for distributed multimedia transmission over the secondary user network, which makes use of opportunistic spectrum access with the help of cognitive radios. We use digital fountain codes to distribute the multimedia content over unused spectrum and also to compensate for the loss incurred due to primary user interference. Primary user traffic is modelled as a Poisson process. We develop the techniques to select appropriate channels and study the trade-offs between link reliability, spectral efficiency and coding overhead. Simulation results are presented for the secondary spectrum access model.     

Constrained light-tree design for WDM mesh networks with multicast traffic grooming

Multicast applications such as IPTV, video conferencing, telemedicine and online multiplayer gaming are expected to be major drivers of Internet traffic growth. The disparity between the bandwidth offered by a wavelength and the bandwidth requirement of a multicast connection can be tackled by grooming multiple low bandwidth multicast connections into a high bandwidth wavelength channel or light-tree. Light-trees are known to be especially suited for networks that carry ample multicast traffic. In this paper, we propose new algorithms to address the problem of multicast traffic grooming. In particular, an Integer Linear Programming (ILP) formulation is proposed for optimal assignments of hop constrained light-trees for multicast connections so that network throughput can be maximized. Hop constrained light-trees improve the scalability of the approach by reducing the search space of the ILP formulation. Since solving the ILP problem is very time consuming for realistically large networks, we are motivated to propose a heuristic algorithm with a polynomial complexity, called Dividable Light-Tree Grooming (DLTG) algorithm. This algorithm is based on grooming traffic to constrained light-trees and also divides a light-tree to smaller constrained light-trees on which traffic is groomed for better resource utilization. Simulations show that the proposed DLTG heuristic performs better than other algorithms. It achieves network throughputs which are very close to the ILP formulation results, but with far lower running times.     

SD‐WLB: An SDN‐aided mechanism for web load balancing based on server statistics

Software‐defined networking (SDN) is a modern approach for current computer and data networks. The increase in the number of business websites has resulted in an exponential growth in web traffic. To cope with the increased demands, multiple web servers with a front‐end load balancer are widely used by organizations and businesses as a viable solution to improve the performance. In this paper, we propose a load‐balancing mechanism for SDN. Our approach allocates web requests to each server according to its response time and the traffic volume of the corresponding switch port. The centralized SDN controller periodically collects this information to maintain an up‐to‐date view of the load distribution among the servers, and incoming user requests are redirected to the most appropriate server. The simulation results confirm the superiority of our approach compared to several other techniques. Compared to LBBSRT, round robin, and random selection methods, our mechanism improves the average response time by 19.58%, 33.94%, and 57.41%, respectively. Furthermore, the average improvement of throughput in comparison with these algorithms is 16.52%, 29.72%, and 58.27%, respectively.     

A cost comparison of distributed and centralized approaches tovideo-on-demand

With video-on-demand (VoD) regarded as one of the drivers for the deployment of broadband integrated service digital networks (B-ISDNs), an important issue is how to provide wide area VoD services most efficiently. A large contributor to the cost of a VoD system is storage. The nature of the service requires massive amounts of storage and bandwidth to be supplied from a video server located within the network. We examine the costs of storage in such servers and develop an efficient allocation scheme designed to minimize this cost. Using this scheme, centralized and distributed approaches to VoD are compared. We conclude that a distributed approach to storage costs no more than a centralized approach and offers considerable advantages in terms of bandwidth requirements and service quality     

Muslin: A QoE‐aware CDN resources provisioning and advertising system for cost‐efficient multisource live streaming

Delivering video content with a high and fairly shared quality of experience is a challenging task in view of the drastic video traffic increase forecasts, as live video traffic will grow 15‐fold by 2022. Currently, content delivery networks provide numerous servers hosting replicas of the video content, and consuming clients are redirected to the closest server. Then, the video content is streamed using adaptive streaming solutions. However, servers and network links often become overloaded during major events, and users may experience a poor or unfairly distributed quality of experience, unless more servers are provisioned. In this paper, we propose Muslin , a streaming solution supporting a high, fairly shared end users' quality of experience for live streaming, while minimizing the required content delivery platform scale. Muslin leverages on MS‐Stream, a content delivery solution, which aggregates video content from multiple servers to offer a high quality of experience for its users. Muslin dynamically provisions servers and replicates content into servers and advertises servers to clients based on real‐time delivery conditions. We have used Muslin to replay a 1‐day video‐games event, with hundreds of clients and several test beds. Our results show that our approach outperforms traditional content delivery schemes by increasing the fairness and quality of experience at the user side with a smaller infrastructure scale.     

A cost effective approach for survivable multicast traffic grooming in bidirectional multiwavelength ring networks

In this paper, we address the problem of survivable multicast traffic grooming in WDM bidirectional ring networks. The rapid growth of multicast applications such as video conferencing, distance learning, and online auction, has initiated the need for cost-effective solutions to realize multicasting in WDM optical networks. Many of these applications, being time critical and delay sensitive, demand robust and fault-tolerant means of data communication. The end user traffic demands in metro environment are in fractional bandwidth as compared to the wavelength channel capacity. Providing survivability at connection level is resource intensive. Hence cost-effective solutions that require minimum resources for realizing survivable multicasting are in great demand. In order to realize multicast traffic grooming in bidirectional ring networks, we propose a node architecture based on Bidirectional Add Drop Multiplexers (BADM) to support bidirectional add/drop functionality along with traffic duplication at each node. We also propose two traffic grooming algorithms, namely Survivable Grooming with Maximum Overlap of Sessions (SGMOS) and Survivable Grooming with Rerouting of Sessions (SGRS). Extensive simulation studies reveal that the proposed algorithms consume minimum resources measured in terms of BADM grooming ports, backup cost, and wavelengths.     

BroadCatch: a periodic broadcast technique for heterogeneous video-on-demand

Many essential multimedia applications rely on video-on-demand technology to deliver a video to different users. A number of periodic broadcast techniques have been proposed for the cost-effective implementation of such systems. Most of these techniques would either try to minimize the server bandwidth, user bandwidth, user storage, user access latency to the video, or a combination of some of the aforementioned parameters. On the other hand, the implementation strategies of these broadcast schemes would necessitate a minimum bandwidth requirement for all users. Multi-resolution techniques address the heterogeneity problem by sacrificing user video quality. In this paper, we consider a different approach that does not possess this disadvantage. Using an incremental channel design at the server side, and a specific broadcast schedule, users can choose among a range of bandwidths to use to download the video at the cost of their access latency and not to the video quality. We prove the correctness of the proposed solution; provide mathematical analysis to demonstrate its heterogeneous behavior, and present performance studies to illustrate its efficiency.     

一对多场景下的公钥时控性可搜索加密

为有效解决多接收者时间相关密文检索问题,采用广播加密技术提出一对多公钥时控性可搜索加密机制--发送者将加密的数据发送至云服务器,使得仅授权用户组成员可检索下载包含特定关键词的密文,但只能在指定的未来时间之后解密.给出方案及其安全游戏模型的形式化定义,提出两种基于q-DBDHI问题的可证明安全方案,并严格证明所提方案在自适应选择明文攻击下是安全的.效率分析表明,两种方案在执行过程中,实现了计算、存储、传输规模与用户规模无关;与相关方案相比,方案2具有更高效率.     

Protocol design for scalable and reliable group rekeying

We present the design and specification of a protocol for scalable and reliable group rekeying together with performance evaluation results. The protocol is based upon the use of key trees for secure groups and periodic batch rekeying. At the beginning of each rekey interval, the key server sends a rekey message to all users consisting of encrypted new keys (encryptions, in short) carried in a sequence of packets. We present a scheme for identifying keys, encryptions, and users, and a key assignment algorithm that ensures that the encryptions needed by a user are in the same packet. Our protocol provides reliable delivery of new keys to all users eventually. It also attempts to deliver new keys to all users with a high probability by the end of the rekey interval. For each rekey message, the protocol runs in two steps: a multicast step followed by a unicast step. Proactive forward error correction (FEC) multicast is used to reduce delivery latency. Our experiments show that a small FEC block size can be used to reduce encoding time at the server without increasing server bandwidth overhead. Early transition to unicast, after at most two multicast rounds, further reduces the worst-case delivery latency as well as user bandwidth requirement. The key server adaptively adjusts the proactivity factor based upon past feedback information; our experiments show that the number of NACKs after a multicast round can be effectively controlled around a target number. Throughout the protocol design, we strive to minimize processing and bandwidth requirements for both the key server and users.