IP网络链路带宽的测量

对网络带宽的测量是非常困难的,目前还没有一种很好的方法能对网络链路带宽进行精确的测量。本文详细讨论了现在流行的pathchar带宽测量算法,并针对其不足提出了一种改进的测量算法－－E－pathchar。最后我们通过试验仿真对两种算法进行了对比,证明E－pathchar算法比pathchar算法的测量精度更高,测量速度更快,耗费网络资源更少。     

网络瓶颈带宽的测量

分析了网络时延与网络带宽的关系,推导出了一定时间间隔的网络报文对通过网络时间间隔的变化与瓶颈带宽的数学关系,并以此设计出了一种测量网络瓶颈带宽的方法。     

IP网络时延变化参数的分析与测量

IP时延变化是一个重要的网络性能参数,分析并准确测量了时钟同步前后的IP网络时延和时延变化,讨论了时钟同步对IPDV测量的影响,并得到了选取3种不同参考时延所计算出的IPDV值。通过深入分析得知,选取前后包时延作为参考能更加准确地测量IPDV。最后,给出了一种消除IPDV的方法。     

时延约束下的等效带宽估计

本文利用流体流方法计算不实时媒体流在缓冲区中的延概率分布函数,给出了时延约束下实时媒体流的等效带宽的表达式。讨论了相关参数对等效带宽的影响。进行了仿真分析和验证。该表达式计算简单,便于实时带宽估计和控制。     

IP网络可用带宽测量技术的研究

网络测量技术是了解网络行为、进行QoS控制、提高网络性能的重要环节和前提基础,而可用带宽测量技术又是网络测量研究中的热点和难点.本文着重分析可用带宽测量的基本原理和算法,介绍现有的测量工具及其实现方法,并提出了进一步的研究方向.     

IP网带宽测量技术研究与进展

网络带宽表征网络传输信息的能力,是衡量网络服务质量的重要指标.本文在分析IP网带宽测量的重要作用及其技术分类的基础上给出了链路/路径带宽、可用带宽、批量数据传输能力等测度定义,系统总结了当前最具代表性的各种IP网带宽测量技术,剖析了它们的内在机理及各自优缺点,同时介绍了多种对应的开源带宽测量工具,并对影响带宽测量收敛速度、准确度的因素进行了讨论,最后对带宽测量的应用前景和研究趋势进行了展望.     

一种时延和时延抖动受约束的启发式多播路由算法

多播路由算法在组播应用中是至关重要的,对视频会议等交互式实时组播业务来说,不仅要考虑时延约束,而且要考虑时延抖动约束。本文提出了一种基于最短时延路径的时延和时延抖动约束的启发式算法,仿真结果表明该算法复杂度较低,而且性能也较好,在算法复杂度和性能之间达到了很好的折中。     

一种基于带宽和时延约束的分布式组播路由算法

针对已有分布式组播路由算法在寻找QoS路由时的低成功率问题,本文提出了一种新的基于带宽和时延约束的分布式组播路由算法-QDMR(QoS-based Distributed Multicast Routing).在为新组播成员搜索连接到组播树的可行路径时,QDMR算法使用RBMF(Reverse Best Metric Forwarding)转发算法代替RPF(Reverse Path Forwarding)转发算法,从而优先搜索满足带宽和时延约束要求的路径,然后才考虑代价的优化.模拟分析表明,QDMR提高了路由搜索的成功率,并且降低了协议开销.     

基于可用带宽满足时延和时延差别 约束的实时多播路由算法

实时多播路由中具有可加性的代价(Cost)不能确切反映网络本质特性,尤其不能反映路径带宽的凹性(Concave).已有基于代价的算法不能很好适应多播应用,需要新的模型和算法.本文采用可用带宽代替代价作为主要度量,并满足实时多播中二个重要约束度量:时延和时延差别.同时基于此三个度量,本文提出二种新的具有多项式复杂性的实时多播路由算法并比较其性能.新算法通过分析得到每路径时延和二约束之间的关系,有效降低涉及时延和时延差别此类问题的复杂性.新算法采用度量反映实时多播本质特性而具有实际推广性.     

网络瓶颈带宽测量的噪声分析

包对/串技术通过两个/多个背靠背发送的测量包所经历的散布间隔来估测一条路径的瓶颈带宽.本文分析了瓶颈带宽测量中的噪声特性,解释了在不同测量包长和测量包数情况下测量样点的分布.基于噪声分析的结论,文中提出了一种基于信号模式的滤波算法(MBFA,Model-Based Filtering Algorithm),它利用信号模式/特征来增强信号、过滤噪声.     

一种长时延网络中移动节点间的时间同步算法

长时延网络的同步对于提升网络传输效率,增加网络吞吐量具有重要意义。现有用于长时延网络典型时间同步协议有TSHL,Tri-Message等,这些同步协议都是针对传播时延保持稳定或者保持短时稳定的长时延网络,但在实际情况中,有时传播时延会发生变化,如节点间相对位移的变化,如何在这样的长时延网络中进行时间同步也是一个值得研究的问题,文中提出了一种运用于长时延网络中移动节点间的时间同步算法,利用节点间多次的信息交互计算出传播时延变化的速率,然后对时延变化进行估算来进行时间偏移的补偿以达到时间同步。     

Adaptive bandwidth management in ATM networks

A framework for adaptive bandwidth management in ATM based networks is proposed. It is based on a layered approach which includes bandwidth allocation to virtual networks. The central concept of this approach is adaptive estimation of the effective bandwidth required, by connections carried in the network. To achieve reliable results the estimation process takes into account both the traffic source declarations and the connection superposition process measurements on the network links. This is done in an optimization framework provided by estimation theory. A study, based on a linear two-state Kalman filter, shows that the proposed approach provides good adaptation to undeclared changes in traffic parameters and that the network performance is significantly improved when compared to the effective bandwidth allocation based solely on the source parameters declarations. These features allow more relaxed source parameter declarations and at the same time permit less stringent source policing. Thus the two bottlenecks influencing bandwidth management in ATM networks can be significantly widened.     

Performance analysis for the dynamic virtual path bandwidth control method in ATM networks

Broadband ISDN (integrated services digital network) should provide various kinds of communication services for multimedia traffic, including voice, computer data, still picture and motion video, and an ATM (asynchronous transfer mode) technology is expected to satisfy those demands. In ATM networks, a VP (virtual path) concept is introduced for simplifying the network resource management. However, if the bandwidth of each VP is fixed, it cannot absorb traffic load fluctuation. In this case, it is likely to happen that one VP has no remaining bandwidth while other VPs on the same physical link have free capacities. Another extremity is that a VP is not introduced, and that all VCs along the same physical link share the whole bandwidth of the link. This can achieve an efficient use of the link, but it apparently requires complicated call set-up procedures. In this paper, we propose a new dynamic VP bandwidth control method, in which the bandwidth allocated to each VP is dynamically changed according to traffic fluctuation. More specifically, in the case that multiple VPs are multiplexed on the same link between two nodes, when the utilization of some VP is increased, that VP requests an additional bandwidth. When the traffic load becomes low and the utilization of a VP is decreased, a part of the assigned bandwidth is released. We consider two methods for this purpose; one is the basic method in which the bandwidth increase request is issued after the VP bandwidth is exhausted. The other is the look-ahead method where the bandwidth increase is requested in advance before the bandwidth starvation. An approximate analysis for the above methods is provided, and the validation of its accuracy is assessed by comparing with simulation results. Through numerical examples, we show that our methods can provide performance improvement in terms of call loss probabilities of each VP.     

一种新的低延迟OBS突发组装算法

提高带宽利用率和降低端到端的时间延迟是下一代网络性能的重要指标,文章简单扼要地介绍了目前OBS网络中在提高带宽利用率和降低端到端的时间延迟方面比较典型的两种组装算法:前向资源预留算法(Forward Resource Reservation,FRR)和精确组装FRR算法(Precision Assembly with Forward Resource Reservation,PAFRR).分析了这两种算法在带宽利用率提高以及平均时间延迟降低方面存在的问题.针对这些问题,提出了一种新的降低延迟的组装周期预测算法(As sembly Duration Forecasting,ADC)并进行了网络仿真.仿真结果验证了该算法在进一步提高带宽利用率方面和降低平均时间延迟方面的有效性.     

同步无线Mesh网络带宽申请与分配策略的改进

数据通信具有突发性与带宽计算的不确定性,现有同步无线Mesh网络带宽申请与分配策略未对此问题进行充分考虑,在设计上存在带宽申请速度慢、带宽分配没有最大化、将数据时隙区分了上下行等不足,导致了QoS保障能力低与网络性能的下降。提出了带宽申请与分配策略的改进方案,包括结合各类数据业务类型的带宽申请条件的描述、检查数据发送队列的时间间隔描述、带宽分配最大化的描述、数据时隙不再区分上、下行的描述。理论分析表明,改进后的方案能够提供精确的QoS保障并提高网络性能。     

A versatile model for TCP bandwidth sharing in networks with heterogeneous users

Enabled by the emergence of various access technologies (such as ADSL and wireless LAN), the number of users with high-speed access to the Internet is growing rapidly, and their expectation with respect to the quality-of-service of the applications has been increasing accordingly. With TCP being the ubiquitous underlying end-to-end control, this motivates the interest in easy-to-evaluate, yet accurate, performance models for a TCP-based network shared by multiple classes of users. Building on the vast body of existing models, we develop a novel versatile model that explicitly captures user heterogeneity, and takes into consideration dynamics at both the packet level and the flow level. It is described as to how the resulting multiple time-scale model can be numerically evaluated. Validation is done by using NS2 simulations as a benchmark. In extensive numerical experiments, we study the impact of heterogeneity in the round-trip times on user-level characteristics such as throughputs and flow transmission times, thus quantifying the resulting bias. We also investigate to what extent this bias is affected by the networks’ ‘packet-level parameters’, such as buffer sizes. We conclude by extending the single-link model in a straightforward way to a general network setting. Also in this network setting the impact of heterogeneity in round-trip times is numerically assessed.     

Network selection based on available link bandwidth in multi-access networks

One of the important aspects of seamless communication for ubiquitous computing is the dynamic selection of the best access network for a multimodal device in a heterogeneous wireless environment. In this paper, we consider available bandwidth as a dynamic parameter to select the network in heterogeneous environments. A bootstrap approximation based technique is firstly utilized to estimate the available bandwidth and compare it with hidden Markov model based estimation to check its accuracy. It is then used for the selection of the best suitable network in the heterogeneous environment consisting of 2G and 3G standards based wireless networks. The proposed algorithm is implemented in temporal and spatial domains to check its robustness. The numerical results show that the proposed algorithm gives improved performance in terms of estimation error (less than 15%), overhead (varies from 0.45% to 72.91%) and reliability (approx. 99%)as compared to the existing algorithm.     

A new bandwidth allocation mechanism for next generation wireless cellular networks

In this paper we design and study the performance of a Medium Access Control (MAC) scheme for the multiplexing and the integrated delivery of voice, mobile messaging, IP, gaming and H.264 videoconference traffic over a high-speed cellular TDMA channel with errors and capture. To the best of our knowledge, this is one of the first papers in the literature investigating the integration of actual H.264 video traces and gaming traffic with other types of traffic over wireless networks. Our results show that the proposed scheme achieves high throughput results while preserving the strict Quality of Service (QoS) requirements of each traffic type, and outperforms two efficient schemes previously proposed in the literature.     

An analytical model for jitter in IP networks

Traditionally, IP network planning and design is mostly based on the average delay or loss constraints which can often be easily calculated. Jitter, on the other hand, is much more difficult to evaluate, but it is particularly important to manage the QoS of real-time and interactive services such as VoIP and streaming video. In this paper, we present simple formulas for the jitter of Poisson traffic in a single queue that can be quickly calculated . It takes into account the packets delay correlation and also the correlation of tandem queues that have a significant impact on the end-to-end jitter. We then extend them to the end-to-end jitter of a tagged stream based on a tandem queueing network. The results given by the model are then compared with event-driven simulations. We find that they are very accurate for Poisson traffic over a wide range of traffic loads and more importantly that they yield conservative values for the jitter so that they can be used in network design procedures. We also find some very counter-intuitive results. We show that jitter actually decreases with increasing load and the total jitter on a path depends on the position of congested links on that path. We finally point out some consequences of these results for network design procedures.