A Network Layer Approach to Enable TCP over Multiple Interfaces

The mobile Internet is set to become ubiquitous with the deployment of various wireless technologies. When heterogeneous wireless networks overlap in coverage, a mobile terminal can potentially use multiple wireless interfaces simultaneously. In this paper, we motivate the advantages of simultaneous use of multiple interfaces and present a network layer architecture that supports diverse multi-access services. Our main focus is on one such service provided by the architecture: Bandwidth Aggregation (BAG), specifically for TCP applications.While aggregating bandwidth across multiple interfaces can improve raw throughput, it introduces challenges in the form of packet reordering for TCP applications. When packets are reordered, TCP misinterprets the duplicate ACKS received as indicative of packet loss and invokes congestion control. This can significantly lower TCP throughput and counter any gains that can be had through bandwidth aggregation. To improve overall performance of TCP, we take a two-pronged approach: (1) We propose a scheduling algorithm that partitions traffic onto the different paths (corresponding to each interface) such that reordering is minimized. The algorithm estimates available bandwidth and thereby minimizes reordering by sending packet pairs on the path that introduces the least amount of delay. (2) A buffer management policy is introduced at the client to hide any residual reordering from TCP. We show through simulations that our network-layer approach can achieve good bandwidth aggregation under a variety of network conditions.Kameswari Chebrolu is an assistant professor in the electrical department of Indian Institute of Technology, Kanpur. Her research interests are in the areas of wireless network architecture, protocol design and analysis. Kameswari obtained her M.S. and Ph.D. degree in Electrical and Computer Engineering from University of California at San Diego, in 2001 and 2004 respectively.Bhaskaran Raman received his B.Tech in Computer Science and Engineering from Indian Institute of Technology, Madras in May 1997. He received his M.S. and Ph.D. in Computer Science from University of California, Berkeley, in 1999 and 2002 respectively. He joined as faculty in the CSE department at Indian Institute of Technology, Kanpur (India) starting June 2003. His research interests and expertise are in communication networks, wireless/mobile networks, large-scale Internet-based systems, and Internet middleware services.Ramesh R. Rao is a professor at the University of California, San Diego (UCSD). He is a member of the faculty of Irwin and Joan Jacobs School of Engineering, since 1984. Professor Rao is the former director of UCSD’s Center for Wireless Communications (CWC), and the current Director of the San Diego Division of the California Institute of Telecommunications and Information Technology [Cal-(IT)²]. As Director of the San Diego Division of Cal-(IT)², he leads several interdisciplinary, collaborative projects. His research interests include architectures, protocols and performance analysis of computer and communication networks, and he has published extensively on these topics. He received his B.E. from the University of Madras and his M.S. and Ph.D. from the University of Maryland at College Park. Most recently, Dr. Rao was honored by being appointed the first holder of the Qualcomm Endowed Chair in Telecommunications and Information Technologies.     

Combined Connection Admission Control and Packet Transmission Scheduling for Mobile Internet Services

Mobile Internet access is expected to be the most popular communication service in the near future. In this paper, we investigate radio resource management for mobile Internet multimedia systems that use the orthogonal frequency division multiple access and adopt the adaptive modulation and coding technique. It is assumed that real-time (RT) service such as streaming video and best-effort (BE) services such as file transfer protocol and hypertext transfer protocol coexist in the systems. We suggest two levels of radio resource management schemes: the connection admission control (CAC) scheme at the first level and the packet transmission scheduler at the second level. The proposed scheduler does not assign higher priority to RT packets over BE packets unconditionally. Instead, only the RT packets that are close to the deadline are given higher priority. Therefore, the performance of BE services is improved at the cost of RT services. To control the performance degradation in RT services within an acceptable level, the CAC algorithm functions as a congestion controller. The combined effects of the proposed CAC and packet scheduling by using the cross-layer simulation that covers from the physical layer to the Internet application layer are evaluated. The numerical results show that the proposed schemes greatly improve the performance of BE services while maintaining the quality of video service at an acceptable level.     

Implementation and performance of VoIP interception based on SIP session border controller

In an effort to provide lawful interception for session initiation protocol (SIP) voice over Internet protocol (VoIP), an interception architecture using session border controller (SBC) is proposed. Moreover, a prototype based on the proposed architecture is implemented. A testbed is set up and tests are carried out in order to analyze the performance and the capability of function entities and interfaces in the proposed architecture. Test results show that SBC interception capability in SIP signaling is superior to that in real-time transport protocol (RTP) media stream. In order to eliminate the possible bottleneck of RTP packets interception in SBC, an analytic model is proposed to investigate the mechanism in which RTP packet’s traffics are shared among different SBC media functions. Analysis results show that multiple SBC media functions can share the RTP packets arrival and can significantly decrease RTP packets service time in SBC. Test results also show that delivery function, collect function and their interfaces in the proposed interception architecture have corresponding interception performance and capability with SBC.     

LTE系统上行链路调度算法研究

长期演进(Long Term Evolution,LTE)已经成为4G无线技术标准。目前,LTE分组调度的下行链路调度被大多数研究者研究,上行链路的研究相对较少。针对上行链路调度无法保证实时业务分组在延迟期限内传输,存在公平性较差、分组丢弃多的问题。因此,提出了一种新的上行链路调度算法。该算法根据实时业务的延迟约束条件建立目标整数线性规划模型,再根据目标整数线性规划模型进行调度。实验结果表明,该算法能保证实时业务分组在延迟期限内传输,适用于实时业务,能确保公平性,最小化分组丢弃,具有较好的适用性。     

Ant colony optimization based packet scheduler for peer-to-peer video streaming

This letter describes a new packet scheduling algorithm for enhancing the quality of distributed peer-to-peer video streaming. The algorithm was designed for when streaming server peers use error recovery such as automatic-repeat-request (ARQ) rather than error protection to avoid overburdening network resources. Ant colony optimization was used for scheduling groups of packets to reflect the channel status and error recovery effect of multiple server peers heuristically. Simulation results show that the proposed algorithm can enhance the quality of distributed video streaming services.     

Downlink Wireless Packet Scheduling with Deadlines

Next-generation cellular wireless communication networks aim to provide a variety of quality-of-service (QoS)-sensitive packet-based services to downlink users. Included among these are real-time multimedia services, which have stringent delay requirements. Downlink packet scheduling at the base station plays a key role in efficiently allocating system resources to meet the desired level of QoS for various users. In this paper, we employ dynamic programming (DP) to study the design of a downlink packet scheduler capable of supporting real-time multimedia applications. Under well-justified modeling reductions, we extensively characterize structural properties of the optimal control associated with the DP problem. We leverage intuition gained from these properties to propose a heuristic scheduling policy, namely, Channel-Aware Earliest Due Date (CA-EDD), which is based on a "quasi- static" approach to scheduling. The per-time-slot implementation complexity of CA-EDD is only O(K) for a system with K downlink users. Experimental results show that CA-EDD delivers up to 50 percent of performance gains over benchmark schedulers. CA-EDD achieves these performance gains by using channel and deadline information in conjunction with application layer information (relative importance of packets) in a systematic and unified way for scheduling.     

An Architecture for Exploiting Multihoming in Mobile Devices for Vertical Handovers & Bandwidth Aggregation

In recent years, mobile devices with multihoming capabilities i.e. equipped with multiple network interfaces have gained large scale popularity. This multihoming capability enables the mobile devices to connect with multiple diverse access networks simultaneously. However, networking protocol stack implemented in current devices is not capable of exploiting the availability of multiple network interfaces. Multihoming can be used to provide two important services: vertical handovers and bandwidth aggregation. Vertical handover enables a multihomed device to switch its connectivity from one access network to another access network without disrupting the communication session. Bandwidth aggregation enables multihomed device to achieve higher throughput by establishing simultaneous connections over multiple available network interfaces. A number of solutions have been proposed to exploit multihoming for vertical handovers and bandwidth aggregation. However, most of these solutions either require the support of additional network entities such as host agent, foreign agent, mobility gateway, proxy, etc. or they require changes in current widely deployed protocol stack in operating system kernels. Dependence on either network operator, administrator or operating system vendors hinders the large scale deployment of these solutions. This paper presents an end-to-end architecture that offers the vertical handover and bandwidth aggregation services to TCP applications. This architecture neither requires any additional network entity nor it requires the changes in current networking protocol stack in operating system kernels. The paper presents the design, implementation and performance analysis of the proposed architecture.     

High-performance template tracking

Tracking systems are important in computer vision, with applications in video surveillance, human computer interfaces (HCI), etc. Consumer graphics processing units (GPUs) have experienced an extraordinary evolution in both computing performance and programmability, leading to a greater use of the GPU for non-rendering applications, such as image processing and computer vision tasks. In this work we show an effective particle filtering implementation for real-time template tracking based on the use of a graphics card as a streaming architecture in a translation-rotation-scale model.     

Performance analysis of an RSVP-capable router

RSVP is a bandwidth reservation protocol that allows distributed real-time applications such as videoconferencing software to make bandwidth reservations over packet-switched networks. Coupled with real-time scheduling mechanisms built into packet routers, the network guarantees to provide the reserved bandwidth throughout the lifetime of the applications. Although guaranteed services are of great value to both end users and carrier providers, their performance cost, due to additional control and data processing overhead, can potentially have a negative impact on the packet throughput and latency of RSVP-capable routers. The goal of this article is to examine the performance cost of RSVP based on measurements from an industrial-strength RSVP implementation on a commercial IP router. The focus is on the detailed evaluation of the performance implications of various architectural decisions in RSVP. We found that RSVP's control messages do not incur significant overhead in terms of processing delay and bandwidth consumption. However, the performance overhead of real-time packet scheduling is noticeable in the presence of a large number of real-time connections. In extreme cases, the performance guarantees of existing real-time connections may not be kept, and some best-effort packets are actually dropped, although the overall bandwidth requirement from these connections is smaller than the available link bandwidth     

An On-Line Scheduler over Hard Real-Time Communication System

By thorough research on the prominent periodic and aperiodic scheduling algorithms,anon-line hard real-time scheduler is presented,which is applicable to the scheduling of packets over a link.This scheduler,based on both Rate Monotonic,pinwheel scheduling algorithm Sr and Polling Serverscheduling algorithms,can rapidly judge the schedulability and then automatically generate a bus tablefor the scheduling algorithm to schedule the packets as the periodic packets.The implementation of thescheduler is simple and easy to use,and it is effective for the utilization of bus link.The orderly executionof the bus table can not only guarantee the performance of the hard real time but also avoid the blockageand interruption of the message transmission.So the scheduler perfectly meets the demand of hard real-time communication system on the field bus domain.     

Efficient variable partitioning and scheduling for DSP processors with multiple memory modules

Multiple on-chip memory modules are attractive to many high-performance digital signal processing (DSP) applications. This architectural feature supports higher memory bandwidth by allowing multiple data memory accesses to be executed in parallel. However, making effective use of multiple memory modules remains difficult. The performance gain in this kind of architecture strongly depends on variable partitioning and scheduling techniques. In this paper, we propose a graph model known as the variable independence graph (VIG) and algorithms to tackle the variable partitioning problem. Our results show that VIG is more effective than interference graph for solving variable partitioning problem. Then, we present a scheduling algorithm known as the rotation scheduling with variable repartition (RSVR) to improve the schedule lengths efficiently on a multiple memory module architecture. This algorithm adjusts the variable partitions during scheduling and generates a compact schedule based on retiming and software pipelining. The experimental results show that the average improvement on schedule lengths is 44.8% by using RSVR with VIG. We also propose a design space exploration algorithm using RSVR to find the minimum number of memory modules and functional units satisfying a schedule length requirement. The algorithm produces more feasible solutions with equal or fewer number of functional units compared with the method using interference graph.     

Improving Security of Real-Time Wireless Networks Through Packet Scheduling [Transactions Letters]

Modern real-time wireless networks require high security level to assure confidentiality of information stored in packages delivered through wireless links. However, most existing algorithms for scheduling independent packets in real-time wireless networks ignore various security requirements of the packets. Therefore, in this paper we remedy this problem by proposing a novel dynamic security-aware packet-scheduling algorithm, which is capable of achieving high quality of security for realtime packets while making the best effort to guarantee realtime requirements (e.g., deadlines) of those packets. We conduct extensive simulation experiments to evaluate the performance of our algorithm. Experimental results show that compared with two baseline algorithms, the proposed algorithm can substantially improve both quality of security and real-time packet guarantee ratio under a wide range of workload characteristics.     

Medium access protocol design for delay-guaranteed multicode CDMA multimedia networks

To enable multimedia real-time applications over next-generation wireless code-division multiple access (CDMA) packet-switching networks, previous efforts show a proper scheduling policy is the key to provide delay-guaranteed access services to various traffic types with different bit error rate (BER) requirements. Considering the support of the prevailing Internet protocol (IP) with variable-length packets in future mobile networks, we develop a mathematically delay-optimal medium access control (MAC) protocol over multicode CDMA (MC-CDMA) environments under the continuous-time assumption. From our investigations, we suggest that a good MAC protocol should be designed by using a proper single-server scheduling policy to guarantee packet-delay, and controlling the maximal number of simultaneous spreading-code transmissions to maintain the required BER. We further evaluate the performance of some MC-CDMA MAC protocols supporting QoS on BER and packet-delay, and show that MAC schemes conforming to our design rules give better performance on packet-delay when maintaining acceptable BER of various traffic types.     

Performance evaluation of the RQMA-CDMA scheme for multimedia traffic with QoS guarantees

The rapid growth of cellular mobile technology in recent years, coupled with the explosive growth of the Internet, has significantly increased the demand for wireless data services. Traffic on mobile devices is expected to be a mix of real-time traffic such as video, voice, and data, with users requiring diverse quality of service (QoS) guarantees for different types of traffic (video, voice and data). One of the primary challenges of providing QoS is how to prioritize and allocate network resources among contending applications. In order to achieve these goals, a scheduling scheme that can provide equitable and effective packet routing is required. This paper proposes a scheduling scheme called remote queuing multiple access-code division multiple access (RQMA-CDMA), whose purpose is to equitably assign bandwidth resources with QoS guarantees to different mobile devices. RQMA-CDMA is a rate scheduling scheme that can be used to assign bandwidth resources in conjunction with GPS (generalized processor sharing). Additionally, we analyze an admission control that is based on signal to interference plus noise ratio (SINR) for multimedia traffic. Finally, the performance of RQMA-CDMA is evaluated and compared to schemes based on CDMA-GPS according to dropped packets, delay, and throughput.     

一种基于紧急程度的自时钟开始时间公平排队分组调度算法

为了克服目前GPS (Generalized Processor Sharing)类调度算法中实时应用分组的排队时延较大且不稳定的局限性,该文提出一种新的分组排队调度算法,该调度算法在计算分组服务标签时添加了一个紧急程度函数,调整了到达分组间的竞争关系,从而可以按照实时性应用的要求来调整到达分组的转发先优级,由此显著降低了实时性应用分组的排队时延和抖动幅度。分析和仿真实验表明,与GPS类其它调度算法相比,该调度算法对于实时应用的分组能提供较低的、更稳定的排队时延保证,同时还继承了GPS类算法的公平性和排队时延有界等特性,而且对系统虚拟时间的跟踪计算更为简捷高效。     

一种IEEE 802.16无线网络的优化带宽调度策略

提出了一种基于预测的自适应实时轮询业务带宽分配机制:SS依据当前带宽需求和以往的实际分配带宽,对实时业务数据包所需求的确切带宽进行预测并调度.给出了数学分析模型和仿真,结果表明,与传统的带宽分配机制相比,该自适应的带宽分配算法能更好地改善系统性能,提高吞吐量,减少时延,减少缓冲区需求.     

EMUNE: Architecture for Mobile Data Transfer Scheduling with Network Availability Predictions

With the mobile communication market increasingly moving towards value-added services, the network cost will need to be included in the service offering itself. This will lead service providers to optimize network usage based on real cost rather than simplified network plans sold to consumers traditionally. Meanwhile, today’s mobile devices are increasingly containing multiple radios, enabling users on the move to take advantage of the heterogeneous wireless network environment. In addition, we observe that many bandwidth intensive services such as video on demand and software updates are essentially non real-time and buffers in mobile devices are effectively unlimited. We therefore propose EMUNE, a new transfer service which leverages these aspects. It supports opportunistic bulk transfers in high bandwidth networks while adapting to device power concerns, application requirements and user preferences of cost and quality. Our proposed architecture consists of an API, a transport service and two main functional units. The well defined API hides all internal complexities from a programmer and provides easy access to the functionalities. The prediction engine infers future network and bandwidth availability. The scheduling engine takes the output of the prediction engine as well as the power and monetary costs, application requirements and user preferences into account and determines which interface to use, when and for how long for all outstanding data transfer requests. The transport service accordingly executes the inferred data transfer schedule. The results from the implementation of EMUNE’s and of the prediction and scheduling engines evaluated against real user data show the effectiveness of the proposed architecture for better utilization of multiple network interfaces in mobile devices.     

异构网络化汽车电子系统中多DAG离线任务调度

公平性和通信开销已成为影响调度性能的主要瓶颈,首先在任务优先级排序阶段,提出基于通信开销权值的轮转调度的公平排序标准;在处理器选择阶段,提出在插入法的基础上将任务分配到具有最小选择值的选择标准;综合上述2个阶段,提出以降低调度长度和减少通信开销为目标的多DAG离线公平任务调度(MDOFTS, multiple DAGs off-line and fairness task scheduling)算法。异构网络化汽车电子系统是一个典型的混合关键级嵌入式系统,既要确保实时性又要降低调度长度,提出了以满足安全关键DAG的多DAG离线优先级任务调度(MDOPTS, multiple DAGs off-line and Priority task scheduling)算法。综合MDOFTS和MDOPTS,提出多DAG离线自适应任务调度(MDOATS, multiple DAGs off-line and adaptive task scheduling)算法,在满足实时性的基础上提高调度性能。实例分析和实验结果表明,提出的算法在调度长度、通信开销、不公平性、最差响应时间和实时性上都优于其他算法。     

低速网络中实时补偿型差额循环调度算法的设计和实现

服务质量(QoS)是目前网络应用研究的一个热点。由于低速链路在当前整个网络中占有相当大的比例,因此研究如何在低速链路上为用户提供具有QoS保证的实时业务已经成为一个重要的课题,其中采取何种调度算法则是实现QoS保证的关键因素之一。该文根据低速链路的特点,提出了一种适合实时分组转发的公平排队调度算法实时补偿型差额循环调度(RCDRR)算法,并用ns2软件对RCDRR算法和DRR算法进行了模拟对比。实验及仿真结果表明: RCDRR调度算法具有公平性好、算法复杂度低、可以降低实时分组在低速链路下的排队时延等特点。     

移动网络中基于两级缓存的流媒体调度算法

提出一种适用于移动通信网的两级缓存流媒体系统结构2CMSA（two—level cache mobile streaming architecture），它突破了移动流媒体系统中终端缓存空间小、无线接入网带宽窄的局限；针对2CMSA结构设计了基于两级缓存的移动流媒体调度算法2CMSS（two—level cache based mobile streaming scheduling algorithm），建立数学模型分析了其性能；仿真实验证明，与原有的移动流媒体系统相比，使用2CMSS调度算法能够有效地节省网络传输开销，降低用户启动时延。