Multimedia group synchronization protocols for integrated servicesnetworks

Unlike traditional data traffic, real time multimedia traffic requires synchronization. Temporal relationships among media must be maintained. Yet delay jitter and the absence of a global clock may disrupt these temporal relationships. This paper introduces new group synchronization protocols for real-time, multimedia applications, including teleconference, teleorchestration and multimedia on demand services. The proposed protocols achieve synchronization for all configurations (one-to-one, one-to-many, many-to-one, and many-to-many), and does so without prior knowledge of the end-to-end delay distribution, or the distribution of the clock drift. The only a-priori knowledge the protocols require is an upper bound on the end-to-end delay. The paper concludes with simulation experiments showing that the protocols work effectively in both LAN and WAN environments     

Multimedia Group synchronization based in standard protocols

Most of actual multimedia tools use RTP/RTCP for inter-stream synchronization, but not for group synchronization. A new proposal of modification of RTCP packets to provide a sender-based method for synchronization of a group of receivers is described and evaluated both objectively and subjectively. The solution takes advantage of the feedback RR RTCP messages and the malleability of RTP/RTCP to provide the information required by the synchronization approach, defining a few new APP RTCP packets useful for synchronization purpose. This modification hardly increases the workload of the network and helps to avoid the asynchronies, between receivers (distributed) and between streams (locally), exceeding the limits, in accordance with the related literature.     

Secure time synchronization protocols for wireless sensor networks

Time synchronization is essential in wireless sensor networks as it is needed by many applications for basic communication. The inherent characteristics of sensor networks do not permit simply applying traditional time synchronization algorithms. Therefore, many new time synchronization algorithms have been proposed, and a few of them provide security measures against various degrees of attacks. In this article we review the most commonly used time synchronization algorithms and evaluate these algorithms based on factors such as their countermeasures against various attacks and the types of techniques used.     

Multimedia traffic characteristics in broadband networks

It is pertinent to develop a set of simple parameters that can best characterize the variability and the statistical correlations of the packet arrival process. These parameters are then used by the network to allocate its resources among the different users in order to avoid congestion and maintain a predefined quality of service (QOS) for each user. The traffic characterization and modeling of the real-time bursty traffic, mainly voice and video traffic sources, using statistical methods are described. It is shown that the variability of the variance of the sum of consecutive packet interarrival times leads to significant queuing delays and therefore is a major cause of congestion in broadband networks     

宽带OFDM系统中的频率同步算法与FPGA实现

频率同步是宽带OFDM系统中的关键技术,IEEE 802.11a 采用OFDM技术,利用其前导序列的重复结构,提出了一种分步的粗细频偏捕获与校正算法,同时为了补偿时域同步后的残余相差和适应信道的时变特性,还提出了一种基于频域导频的相位跟踪方案.通过仿真对算法性能进行了分析,并通过FPGA实现验证了算法在宽带OFDM无线接入系统中应用的有效性.结果表明,算法可以对±2个子载波频率间隔内的载波频偏实施快速捕获与跟踪,估计精度达10-3,满足突发分组传输OFDM系统的快速同步要求.     

An approximate performance model for a multislot integratedservices system

An approximate model (with finite or infinite waiting room) is presented for an integrated service system with three types of traffic: a first-offered narrowband traffic, an overflow narrowband traffic, and a wideband traffic. A narrowband call requires a single server, while the number of servers required to serve a wideband call is N. The blocked narrowband calls are lost while the blocked wideband calls are delayed in a finite or infinite waiting room. Based on two assumptions with regard to the characteristics of the system, the system is resolved by decomposition. The corresponding improvements in numerical efficiency as well as in computational storage requirements are significant enough to enable use of the model within network optimization algorithms. The model provides a very good approximation for the system performance, i.e. the blocking probabilities of the two narrowband traffics, the loss probability (in the case of finite waiting room), the probability of nonwaiting, and the average waiting time of wideband traffic     

On the scalability of data synchronization protocols for PDAs andmobile devices

Personal digital assistants and other mobile computing devices rely on synchronization protocols in order to maintain data consistency. These protocols operate in environments where network resources such as bandwidth, memory and processing power are limited. We examine a number of popular and representative synchronization protocols, such as Palm's HotSync, Pumatech's Intellisync and the industry-wide SyncML initiative. We investigate the scalability performance of these protocols as a function of data and network sizes and compare them to a novel synchronization approach, CPISync, which addresses some of their scalability concerns. The conclusions of this survey are intended to provide guidance for handling scalability issues in synchronizing data on large, heterogeneous, tetherless networks     

Guest editorial broadband access networks: architectures and protocols

The 13 papers in this special issue cover a diversity of topics on the design of network architectures and protocols for passive optical networks (PONs), WiMAX networks, and integrated optical and wireless access networks.     

A SYNCHRONIZATION ALGORITHM FOR HF （HIGH FREQUENCY） BROADBAND OFDM SYSTEM

In this letter, a kind of associated synchronization algorithm which is suitable for HF （High Frequency） broadband OFDM （Orthogonal Frequency Division Multiplexing） system is presented based on describing and constructing the GMW （Gorden, Mills and Welch） sequence. The algorithm is based on the Schmidl and Minn＇s symbol timing principle, the constructed GMW sequence is transmitted and disposed, and the synchronization is adjudicated using the correlation of GMW sequence. The simulation result indicates that this algorithm has high performance synchronization ability under the low SNR （Signal to Noise Ratio） at two different kinds of channel models.     

宽带OFDM系统定时同步的硬件实现

针对宽带OFDM系统中的定时同步问题，提出了一种基于DSP和FPGA混合结构的硬件实现方案。相对于全FPGA的实现方案，该方案具有资源消耗少，定时精度高等一系列特点，并且具有灵活的可扩展性，可以满足多种通信协议的定时同步实现。在无线环境下的实验结果验证了该方案在实际系统中的可行性，以及优良的定时和调整能力。     

A multimedia synchronization model and its implementation intransport protocols

An implementation of a synchronization mechanism in transport protocol to support multimedia applications over a packet or cell switched network is proposed. In designing such a mechanism for practical use, ease of implementation and capability of handling random delay of packets are two key issues for success. Since the random delay of packet or cell switched networks makes synchronization among media more complicated after the transmission across the network, a model which considers the random transmission delay is hence required to specify the temporal relationship among media. Therefore, a real-time synchronization model is presented to satisfy this requirement in the paper. Based on the proposed synchronization model, a transport protocol, namely the multimedia synchronization transport protocol (MSTP), is designed and implemented. We have implemented a prototype system using the MSTP protocol and built a simulation system for the wide area network (WAN) environment in order to evaluate the performance of the MSTP protocol     

基于DSP的无线宽带通信接收机信号同步模块设计

基于目前无线通信领域飞速发展,为促进嵌入技术在无线通信领域的应用,DSP是目前应用最热门的嵌入式处理器之一,在无线通信领域应用很广.本文采用DSP嵌入式处理器,通过软硬件相结合的方法,结合软件设计与验证试验,得出无线宽带通信接收机信号同步模块的设计方案.本文重点是从信号同步原理和信号同步实现两方面入手研究了利用DSP实现无线宽带通信系统信号同步模块的设计.     

Start-time fair queueing: a scheduling algorithm for integratedservices packet switching networks

We present a start-time fair queueing (SFQ) algorithm that is computationally efficient and achieves fairness regardless of variation in a server capacity. We analyze its single server and end-to-end deadline guarantee for variable rate fluctuation constrained (FC) and exponentially bounded fluctuation (EBF) servers. To support heterogeneous services and multiple protocol families in integrated services networks, we present a hierarchical SFQ scheduler and derive its performance bounds. Our analysis demonstrates that SFQ is suitable for integrated services networks since it: (1) achieves low average as well as maximum delay for low-throughput applications (e.g., interactive audio, telnet, etc.); (2) provides fairness which is desirable for VBR video; (3) provides fairness, regardless of variation in server capacity, for throughput-intensive, flow-controlled data applications; (4) enables hierarchical link sharing which is desirable for managing heterogeneity; and (5) is computationally efficient     

Cost-based scheduling and dropping algorithms to support integratedservices

Applications with diverse performance objectives must be supported on a single packet-switched network. The efficiency of such networks can be greatly improved through the use of sophisticated scheduling and dropping algorithms within the queues that form at the network access points and in switches throughout the network. In the present approach, arbitrary performance objectives are expressed in the form of cost functions, which map the queueing delay experienced by each packet to a cost incurred. The heuristic algorithms, cost-based scheduling (CBS) and cost-based dropping (CBD), then attempt to optimize network performance as perceived by the applications by minimizing the total cost incurred by all packets. Appropriate cost functions are presented for common applications. Scheduling and dropping algorithms are defined from these cost functions. It is demonstrated that network performance is better when these algorithms are used as opposed to the common alternatives. Also, contrary to conventional wisdom, some evidence is presented indicating that sophisticated scheduling may be preferable to sophisticated dropping as a means of adjusting loss rates     

Dynamic bandwidth allocation on a slotted ring with integratedservices

The authors discuss what they consider the fundamental issue of bandwidth allocation on an integrated local area network. An approach is introduced for dynamic bandwidth allocation which is based on traffic prediction concepts. It is especially well suited for real-time services such as video and voice. Using a control model two allocation schemes are proposed: the first is based on an analytical model of the traffic flow; the second is a simpler version that can be easily implemented on very high-speed systems. The results of simulation studies indicate a marked improvement in performance. The presented approach is especially effective when used in systems with large transmission path latencies as the network performance does not deteriorate with increasing latency. This is very useful if the network is to be used as a metropolitan area network     

Multimedia transmission with adaptive QoS based on real‐time protocols

In this paper, we describe a mechanism for adaptive transmission of multimedia data, which is based on real‐time protocols. The proposed mechanism can be used for unicast or multicast transmission of multimedia data over heterogeneous networks, like the Internet, and has the capability to adapt the transmission of the multimedia data to network changes. In addition, the implemented mechanism uses an inter‐receiver fairness function in order to treat the group of clients with fairness during the multicast transmission in a heterogeneous environment. The proposed mechanism uses a ‘friendly’ to the network users congestion control policy to control the transmission of the multimedia data. We implement a prototype application based on the proposed mechanism and we evaluate the proposed mechanism both in unicast and multicast transmission through a number of experiment and a number of simulations in order to examine its fairness to a group of clients and its behaviour against transport protocols (TCP) and UDP data streams. Copyright © 2003 John Wiley & Sons, Ltd.     

Priority queueing strategies and buffer allocation protocols fortraffic control at an ATM integrated broadband switching system

The problem of furnishing an asynchronous transfer mode (ATM) based broadband-ISDN (B-ISDN) with two bearer services supporting different grades of transfer quality is addressed. The focus is on priority bandwidth and buffer management in the ATM communications nodes (switches, multiplexers or concentrators, and expanders) in the context of a multichannel network architecture. Detailed queueing analyses and simulations and results are provided to evaluate the differentiation between traffic classes that can be achieved by different strategies. The implementation complexity of the different schemes is discussed. Various priority queueing strategies characterized mainly by different degrees of resource sharing and a general system model for performance evaluation are introduced. Performance comparisons and design tradeoffs are addressed     

网同步设备同步方式管理算法

针对专业通信网络的同步方式,提出了一种规划与调整网络中网同步设备同步方式的管理算法,并对其进行了仿真验证。文中提出了主从同步工作时时钟跟踪路经的择优原则以及对应算法,该算法可以用于同步管理方式的预先规划和运行中的调整。