A scheduler ASIC for a programmable packet switch

While the Internet is successful in supporting traditional data-only traffic, an integrated services Internet is inevitable with the emergence of new applications such as voice, video, multimedia, and interactive video conferencing. Such an integrated services network should support a wide range of applications with diverse quality of service requirements and traffic characteristics. Provision for quality of service in packet networks in general, and in the Internet in particular, is the focus of most of the recent developments in switching and routing system design. We designed a generic, single-queue scheduler engine for use in a programmable packet switch/router to handle IP packets, ATM cells, or a combination of both. Comprising 275,000 gates, the 0.35-micron ASIC is incorporated into a prototype programmable packet switch     

Performance analysis of an integrated voice/data cut-through switching network

In this paper, the performance of an integrated voice/data cut-through switching network is studied. We first derive cut-through probabilities of voice and data packets at intermediate nodes. Then, the Laplace transform for the network delay is obtained. According to our numerical results, the cut-through switching method is superior in its delay characteristics to the conventional packet switching3for voice and data in integrated voice/data networks.     

一种基于DiffServ的IP电话QoS实现方案

如何为IP网络中的业务提供QoS保证正成为lP技术所要解决的关键问题。文中基于Diffserv提出了一种IP电话QoS方案,并对其基本思想及实现方法作了详细介绍。该方案将不同的分组数据包设置成不同的优先级,其中系统控制分组数据包优先级最高,语音分组数据包次之,普通数据分组数据包最低,使得系统控制分组和语音分组数据包的平均等待时间缩短。研究结果表明,对不同数据包进行优先级设置是改善IP电话QoS的一种可行方法。     

基于SDN的OpenFlow交换机数据包流水线处理机制

目前,软件定义网络(Software Defined Networking,SDN)已成为网络研究与开发的重点,但相关的研究与开发工作还仅仅局限于园区网络和数据中心网络等。由于SDN控制层与数据层处理效率的限制,SDN面向互联网这样超大规模网络的研究还基本处于空白阶段。为了提升SDN的性能以使其适应大规模网络的特点,挖掘SDN数据层中并行加速处理的可能性,提出了将流水线技术应用到SDN数据层中交换机对数据包的转发过程。另外,结合SDN南向接口OpenFlow协议提供的交换机工作规范,设计了适用于OpenFlow交换机数据包转发的三级流水线处理机制。仿真实验说明,将流水线应用到SDN中能有效加快OpenFlow交换机的数据包转发速度。     

A multi-path mechanism for reliable VoIP transmission over wireless networks

With the advances in audio encoding standards and wireless access networks, voice over IP (VoIP) is becoming quite popular. However, real-time voice data over lossy networks (such as WLAN and UMTS), still posses several challenging problems because of the adverse effects caused by complex network dynamics. One approach to provide QoS for VoIP applications over the wireless networks is to use multiple paths to deliver VoIP data destined for a particular receiver. This paper introduced cmpSCTP, a transport layer solution for concurrent multi-path transfer that modifies the standard stream control transmission protocol (SCTP). The cmpSCTP aims at exploiting SCTP’s multi-homing capability by selecting several best paths among multiple available network interfaces to improve data transfer rate to the same multi-homed device. Through the use of path monitoring and packet allotment techniques, cmpSCTP tries to transmit an amount of packets corresponding to the path’s ability. At the same time, cmpSCTP updates the transmission strategy based on the real-time information of all of paths. Using cmpSCTP’s flexible path management capability, we may switch the flow between multiple paths automatically to realize seamless path handover. The theoretical analysis evaluated the model of cmpSCTP and formulated optimal traffic fragmentation of VoIP data. Extensive simulations under different scenarios using OPNET verified that cmpSCTP can effectively enhance VoIP transmission efficiency and highlighted the superiority of cmpSCTP against the other SCTP’s extension implementations under performance indexes such as throughput, handover latency, packet delay, and packet loss.     

基于二项式概率的冗余传输策略

分析现有音视频传输的丢包恢复技术,结合前向纠错Reed-Solomon冗余编码及交织恢复丢包技术提出二项式概率模型。该模型根据接收端反馈的结果计算需要编码的冗余包个数,使用交织技术将音频冗余包与原始数据混合传输,有效节省了带宽资源。实验结果表明,该模型在网络拥塞的情况下,能根据实际情况产生足够的冗余数据包,使接收端收到数据后还原出原始数据并播放,提高了音视频的传输质量和播放质量。     

基于E-Model的语音帧分组传输性能研究

voIP的语音帧分组大小是实时语音传输的关键参数。为提高网络效率和最大话路数,采用EModel的方法分析了RTP包中语音帧个数、语音长度、丢包概率和抖动缓冲区大小对语音质量的影响,给出了不同带宽时的最佳传输分组大小。仿真结果表明,在保证最基本的话音质量情况下,为不同链路确定合适的分组语音帧数能有效提高链路的最大话路数。     

自适应低轨卫星MAC协议性能分析

针对低轨卫星通信信道碰撞检测能力弱,时延较长和大业务量的特点,提出一种具有接入控制机制的自适应APRMA MAC协议。通过对信道负载和业务优先级判断来确定不同业务的接入概率函数,并且接入概率在每个时隙中通过更新来动态适应系统资源的变化。该MAC协议确保多个终端合理共享有限的无线资源同时,系统能达到高容量。通过仿真对语音业务丢包概率、数据包平均时延和数据业务吞吐量三个衡量协议性能指标与传统协议进行分析对比,证明了APRMA MAC协议显著改善系统性能。     

Performance considerations for integrated voice and data networks

Performance considerations, particularly network delays, for integrated voice and data networks are reviewed. The nature of the delay problem is discussed, followed by a review of concepts, objectives and advances in enhanced circuit, packet and hybrid switching techniques, including fast circuit switching (FCS), virtual circuit switching (VCS), buffered speech interpolation (SI), packetized virtual circuit (PVC), cut-through switching (CTS), composite packets and various frame-management strategies for hybrid switching.In particular, the concept of introducing delay to resolve contention in SI is emphasized and, when applied to both voice talkspurts and data messages, this forms a basis for a relatively new approach to network design called transparent message switching (TMS). This approach and its potential performance advantages are reviewed in terms of various architectural aspects of integrated services networks, such as packet structure, multiplexing scheme, server structure and queuing performance, network topology and network protocols.A number of traffic-management strategies and their grade-of-service implications for voice service are discussed. These strategies include voice call and data session blocking, voice talkspurt and data message buffering, speech loss and data integrity and speech processing techniques, including variable quality, rate, speed and entropy coding. Emphasis is placed on the impact of variable delays on voice traffic, especially the importance of generating and preserving appropriate length speech talkspurts in order to mitigate the effects of variable network delay.     

Availability of end-to-end ideal QoS in IP packet networks

Network processor technology has advanced to the point where high-precision time-based store-and-forward logic is readily incorporated into packet switches and routers. With appropriate scheduling, packets from multiple flows can be serviced without contending for link resources. Accordingly, packet flows traversing a network of switching elements can have both path and time determinacy attributes which support ideal end-to-end QoS (zero jitter, zero loss, acceptable end-to-end latency) for real-time UDP packet flows and guaranteed goodput for TCP flows. One approach to packing a network with a relatively large number of such deterministic flows, i.e. achieving high availability of the ideal QoS service in a network, uses precise buffering of packets at each switch, which introduces latency. This paper describes analysis methods for quantifying how much buffering may be necessary to achieve high (99.999%) availability. For typical network topologies the analysis shows that buffering latency requirements are very small compared to transport delays, even when the network is highly utilized with heterogeneous (e.g. voice, video, circuit emulation, and data) traffic. Actual physical implementations have empirically validated the analysis results as well as the scalability of the end-to-end, time-based forwarding approach and the end-to-end availability of ideal QoS services in IP packet networks.     

Adaptive playout scheduling and loss concealment for voice communication over IP networks

The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score.     

Priority queues and sorting methods for parallel simulation

The authors examine the design, implementation, and experimental analysis of parallel priority queues for device and network simulation. They consider: 1) distributed splay trees using MPI; 2) concurrent heaps using shared memory atomic locks; and 3) a new, more general concurrent data structure based on distributed sorted lists, designed to provide dynamically balanced work allocation and efficient use of shared memory resources. We evaluate performance for all three data structures on a Cray-TSESOO system at KFA-Julich. Our comparisons are based on simulations of single buffers and a 64×64 packet switch which supports multicasting. In all implementations, PEs monitor traffic at their preassigned input/output ports, while priority queue elements are distributed across the Cray-TBE virtual shared memory. Our experiments with up to 60000 packets and two to 64 PEs indicate that concurrent priority queues perform much better than distributed ones. Both concurrent implementations have comparable performance, while our new data structure uses less memory and has been further optimized. We also consider parallel simulation for symmetric networks by sorting integer conflict functions and implementing a packet indexing scheme. The optimized message passing network simulator can process ~500 K packet moves in one second, with an efficiency that exceeds ~50 percent for a few thousand packets on the Cray-T3E with 32 PEs. All developed data structures form a parallel library. Although our concurrent implementations use the Cray-TSE ShMem library, portability can be derived from Open-MP or MP1-2 standard libraries, which will provide support for one-way communication and shared memory lock mechanisms     

实时语音的自适应回放算法

由于丢包和延时抖动的引入而使网络传输的实时语音质量让人难以接受，目前对丢包和延时抖动提出了很多的解决方案．但是却很少把这两者结合在一起进行研究。本文提出了一种新的自适应回放算法，通过监测接收和回放队列，结合丢包的自适应恢复技术，达到语音高质量的连续回放。实验证明，该算法能在严格的平均回放延时条件下努力减小由于超时而引起的丢包，获得较好的重建语音质量。     

面向链路洪泛攻击的多维检测与动态防御方法

针对现有链路洪泛攻击检测存在的不足，提出了多维指标检测算法，通过会话连接时长、数据分组低速比例、数据分组距离均匀性、平均低速率数据分组占比、低速数据分组占比变化率5维要素对存在异常的转发链路进行多维检测，改善了现有方法误报率高的情况。进一步，提出基于染色理论的“控制器-交换机”动态部署方法，解决了现有防御缓解机制存在的“难以实际部署在交换机变体类型受限的实际环境中”问题。最后，实验验证所提方法的有效性。     

DoS攻击下的信息物理系统事件触发预测控制设计

针对信息物理系统(CPS)安全控制设计问题,提出拒绝服务(DoS)攻击下具有任意有界丢包的事件触发预测控制(ETPC)方法.首先,考虑DoS攻击能量的有限性及攻击行为的任意性,将DoS攻击描述为事件触发通信机制下的任意有界丢包;其次,在控制器端利用最近一次收到的状态信息进行控制器增益序列的预测设计以补偿DoS攻击造成的数据包丢失;随后,基于Lyapunov稳定性理论及切换系统分析方法考虑了DoS攻击下CPS的安全性并给出了控制序列设计方法.所提出的ETPC设计方法只需利用最近时刻收到的状态信息,无需满足传统CPS稳定性对最大允许丢包数的约束,为大时滞CPS的稳定性分析及控制提供了有效的解决方案.最后,通过仿真实例验证所提出的基于事件触发预测控制设计方法的有效性.     

高速地址Cache--散列表的应用

路由交换机由IP包进行转发时，需要查找路由表获得转发路径。但在网络层上实现此功能是一个耗费时间的过程，特别是在一个比较大的网络中进行路由交换时，其路由表会相当庞大，路由查找就成了交换机的一个瓶颈。为了解决这个问题，可采用高速地址缓存来加快路由查找过程。其基本思路是第一次IP包的路由确定后，以后的包直接转发。在具体实现中，需要有一个高速地址缓存路由信息，以便使后续的到达同一目的地的IP包块快速通过路交换机。对高速地址缓存的实现进行了探讨。     

Uplink Scheduling for Supporting Real Time Voice Traffc in IEEE 802.16 Backhaul Networks

In this paper we propose simple enhancements to the bandwidth (BW) request messages in IEEE 802.16 for supporting real-time packet voice traffc. Three different BW request formats are proposed, each requiring a different amount of latency information about the buffered packets at the SS. On this basis, packet scheduling schemes are proposed for the BS to make resource allocations for real-time traffc. Our results show that the proposed BW request and scheduling schemes achieve significantly lower packet loss probability than the standard IEEE 802.16 BW request with round robin scheduling. The results further show that there is an optimum point about how much delay information the SS should report to the BS in order to best utilize the uplink resources while the SS provides satisfactory real-time performance for the voice traffc.     

基于硬件的乱序报文重组算法

在硬件实现TCP时,TCP协议对接收报文的乱序管理使硬件实现复杂度大为增加。针对该问题,提出一种利于硬件执行的算法,将数据直接基于序列号进行放置,同时进行序列号映射比较。实验结果表明,该算法具有易于实现、执行效率高的优点,使10G接口的TCP数据传输率达到4.64 Gb/s。     

HMM delay prediction technique for VoIP

This paper proposes a new algorithm for predicting audio packet playout delay for voice conferencing applications that use silence suppression. The proposed algorithm uses a hidden Markov model (HMM) to predict the playout delay. Several existing algorithms are reviewed to show that the HMM technique is based on a combination of various desirable features of other algorithms. Voice over Internet protocol (VoIP) applications produce packets at a deterministic rate but various queuing delays are added to the packets by the network causing packet interarrival jitter. Playout delay prediction techniques schedule audio packets for playout and attempt to make a reasonable compromise between the number of lost packets, the one-way delay and the delay variation since these criteria cannot be optimized simultaneously. In particular, this paper will show that the proposed HMM technique makes a good compromise between the mean end-to-end delay, end-to-end delay standard deviation and average packet loss rate.     

Performance evaluation of two multiple-access protocols

A real-time, microprocessor-based simulator was designed to study the packet transmission of voice on a broadcast type local area network, based on the CSMA/CD and Hymap multiple-access protocols. The speech quality is evaluated subjectively. A packetization-frozen protocol is used to eliminate the successive collisions due to possible synchronization of packet generation among stations. The variance of the network delay is bounded by discarding packets which have not been transmitted within a certain amount of time. Smooth speech output can be obtained by introducing additional buffer delay at the receiver.