Performance of a distributed control LAN access protocol supporting voice and data traffic

This paper describes a local area network access protocol for integrated voice and data communications in a token passing ring. The transmission scheme is decentralized and suitable for both voice and data services. In this system access delay for voice packets can be bounded to a desired upper limit, even at heavy network load. The quality of speech remains unaffected at any level of offered data load and all voice packets are served within the specified delay limit. The protocol provides equal priority to pure data at normal network load while voice is given priority over pure data at heavy network load. The delay-throughput characteristic of both voice and data traffic has been evaluated at a transmission speed of 10 Mb s^-1 and it is shown that the mean delay for voice packets, at heavy data load, can be limited to a value equal to half the threshold token rotation time.     

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.     

Analysis of playout strategies for voice transmission using packet switching techniques

Due to variations in network delay, a stream of voice packets with deterministic interarrival times to a data network may not have deterministic interdeparture times at the destination. Two playout schemes which are designed to remove such variations in delay are considered. Analytic results for the performance of these two schemes are obtained. Numerical examples showing the effect of coefficient of variation of interdeparture time on performance are presented.     

PROFIBUS通信性能测试与参数设置分析

PROFIBUS现场总线协议的实时性是评价其性能的关键因素,因此,如何研究和计算实时性能参数显得尤为重要. 针对这一问题,在对PROFIBUS总线存取协议深入分析的基础上,设计了基于PROFIBUS-DP通信性能测试平台. 通过测试平台分析和计算了包信息率、传输效率、网络平均利用率、网络吞吐量、传输延迟和令牌循环时间等实时性能参数,并给出了总线循环时间与主站个数及报文数量之间的关系之间的关系,从而定性定量分析了PROFIBUS-DP现场总线的实时性能.     

Load Estimation and Control in Best-Effort Network Domains

A mechanism for the estimation of the available bandwidth between two end-points of a best-effort network is presented. The estimation is obtained by a simple statistical analysis of the effect that the network has on a synchronous train of packets. The possibility of exploiting self-similar characteristics of the delay jitters is also discussed, and a possible use of the estimates for management actions is suggested.     

Neural and fuzzy computation techniques for playout delay adaptation in VoIP networks

Playout delay adaptation algorithms are often used in real time voice communication over packet-switched networks to counteract the effects of network jitter at the receiver. Whilst the conventional algorithms developed for silence-suppressed speech transmission focused on preserving the relative temporal structure of speech frames/packets within a talkspurt (intertalkspurt adaptation), more recently developed algorithms strive to achieve better quality by allowing for playout delay adaptation within a talkspurt (intratalkspurt adaptation). The adaptation algorithms, both intertalkspurt and intratalkspurt based, rely on short term estimations of the characteristics of network delay that would be experienced by up-coming voice packets. The use of novel neural networks and fuzzy systems as estimators of network delay characteristics are presented in this paper. Their performance is analyzed in comparison with a number of traditional techniques for both inter and intratalkspurt adaptation paradigms. The design of a novel fuzzy trend analyzer system (FTAS) for network delay trend analysis and its usage in intratalkspurt playout delay adaptation are presented in greater detail. The performance of the proposed mechanism is analyzed based on measured Internet delays.     

VoWLAN中QoS延迟性能分析与改进

VoWLAN（VoIP over WLAN）技术是一种利用无线局域网（WLAN）来传输VoIP帧的技术。VoWLAN中的语音业务是对延时较为敏感的实时业务，因此对其作延迟分析尤为重要。重点介绍了VoWLAN系统中AP节点处延迟产生过程，并采用一种新的具有优先级的M/G/1排队分析模型对AP节点处的延迟进行详细分析，该模型比以往所采用的单一的基于泊松过程的分析模型更精确；探讨了减少延迟的方法，提出新的点协调功能（PCF）方式下站点轮询算法，该算法可以更好的支持VoWLAN系统的VoIP等实时业务；最后用OPNET仿真工具进行了验证。     

Performance evaluation of client-server systems

A client-server system is a distributed system where a server station receives requests from its client stations, processes the requests and returns replies to the requesting stations. The authors consider client-server systems in which a set of workstations access a file server over a local area network. The systems are modelled by a class of stochastic Petri nets. The mean response time, the throughput and the parametric sensitivities are evaluated for a client-server system based on token ring network and a system based on CSMA/CD network. These models are different from the prevalent performance models of token ring or CSMA/CD network systems because of the message interdependencies introduced by the clients-server structure. An approximate analytic-numeric method rather than simulation is used to solve the models. The solution method and the accuracy of approximation are also discussed     

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.     

分组交换网络语音回放控制及性能分析

语音通信中的回放控制是保证语音平滑性的关键。本文深入研究分组交换网中的语音回放控制及其性能分析问题。在总结语音通信系统的特点和端到端时延的组成及其计算方法的基础上，提出了一个简单可行的语音业务回放控制模型，采用双令牌桶对语音流量进行整形。基于最新的网络演算理论，推导出了给定端到端时延、语音到达曲线和网络服务曲线条件下的语音回放时延、需要分配的速率和缓冲区长度的计算公式。最后通过应用实例分析验证了本文的分析结论。     

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.     

An extended token bus protocol for embedded networks

The network environment considered in this paper exists in commercial and military products such as Autonomous Aircraft and Vehicles. In these products sensors and processors are utilized for control, diagnosis, repair and communication. We refer to such networks as embedded networks and present a bus protocol suitable for these networks. Two modes of operation have been defined for the proposed protocol (IMAP). Normal mode is defined as that in which token passing is done in a random order and the token remains within a cluster of active stations. The other mode of operation is called the interrupt mode. In this mode, token bus operation is carried out and the token is passed through every station. Performance in terms of channel utilization and delay characteristics of IMAP is compared to the token bus and CSMA/CD performance. The performance models are developed by obtaining time-distance diagrams of token bus and IMAP. It is observed that IMAP has better performance characteristics than the conventional token bus and CSMA/CD.     

Analysis of voice and low-priority data traffic by means of brisk periods and slack periods

Differentiating data traffic in an integrated voice and data network into high and low priority classes, it is noted that spare voice transmission capacity may be used to transmit low priority data packets, which can tolerate large delays. This paper introduces the notion of brisk periods and slack periods for studying the fluctuation in the number of voice calls in progress and the number of customers, in general, in a multiserver queue with Poisson arrivals and exponential service times. Recurrence relations are derived for the Laplace-Stieltjes transform of the probability density functions and the mean and variance of the length of brisk periods and slack periods of voice calls and the mean and variance of the increase in low priority data packets during brisk periods of voice calls. Computation of spare transmission capacity available during slack periods of voice calls is outlined.     

Subjective Quality of Speech Over Packet Networks as a Function of Packet Loss, Delay and Delay Variation

An expanding proportion of voice traffic is being carried by packet networks. Speech quality can be impaired in qualitatively new ways in packet networks when packets are lost or the spacing between them is distorted. Three parameters that characterize the performance of packet networks were examined for their relative impact on speech quality as judged by human observers: network delay or latency, packet loss, and packet delay variation or jitter. We manipulated these variables via a network emulator made available by NIST. This report summarizes five laboratory experiments that examined the variables in a variety of experimental procedures for presenting and judging speech. The experiments agreed in showing that the relative importance of the variables for affecting speech quality was, in decreasing order: packet loss, jitter, delay. The effect on speech quality of 200 ms of network delay was shown to be equivalent to the effect of one percentage point of packet loss. Many consumers also traded off some speech quality for a free, added feature, unified messaging.     

On the challenge and design of transport protocols for MMORPGs

Although MMORPGs are becoming increasingly popular as well as a highly profitable Internet business, there is still a fundamental design question: Which transport protocol should be used—TCP, UDP, or some other protocol? In this paper, we first evaluate whether TCP is suitable for MMORPGs, and then propose some novel transport strategies for this genre of games. Our analysis of a trace collected from a TCP-based MMORPG called ShenZhou Online indicates that TCP is unwieldy and inappropriate for MMORPGs. We find that the degraded network performance problems are due to the following characteristics of MMORPG traffic: 1) tiny packets, 2) a low packet rate, 3) application-limited traffic generation, and 4) bi-directional traffic. Since not all game packets require reliable transmission or in-order delivery, transmitting all packets with a strict delivery guarantee causes high delays and delay jitters. Therefore, our proposed transport strategies assign game packets with appropriate levels of transmission guarantee depending on the requirements of the packets’ contents. To compare the performance of our approach with that of existing transport protocols, we conduct network simulations with a real-life game trace from Angel’s Love. The results demonstrate that our strategies significantly reduce the end-to-end delay and delay jitter of packet delivery. Finally, we show that our strategies effectively raise satisfaction levels of the game players.     

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。     

Scheduling and admission control for integrated-services networks: the Priority Token Bank

This paper proposes a new mechanism called the Priority Token Bank for admission control, scheduling and policing in integrated-services networks. In such networks, both arrival processes and performance objectives can vary greatly from one packet stream to another. There are two principal components to the Priority Token Bank: accepting or rejecting requests to admit entire packet streams, where acceptance means guaranteeing that the packet stream's performance objectives will be met, and scheduling the transmission of packets such that performance objectives are met, even under heavy loads. To the extent possible, the performance of traffic is also optimized beyond the requirements. The performance achieved with the Priority Token Bank is compared to that of other typical algorithms. It is shown that, when operating under the constraint that the performance objectives of applications such as packet voice, video and bulk data transfer must be met in an ATM network, the mean delay experienced by other traffic is much better with the Priority Token Bank. Furthermore, the admission control algorithm can guarantee requirements will be met, and admit more traffic than the common alternatives.     

Synchronous bandwidth allocation in FDDI networks

It is well known that an FDDI token ring network provides a guaranteed throughput for synchronous messages and a bounded medium access delay for each node/station. However, this fact alone cannot effectively support many real-time applications that require the timely delivery of each critical message. The reason for this is that the FDDI guarantees a medium access delay bound to nodes, but not to messages themselves. The message-delivery delays may exceed the medium-access delay bound even if a node transmits synchronous messages at a rate not greater than the guaranteed throughput. We solve this problem by developing a synchronous bandwidth allocation (SEA) scheme which calculates the synchronous bandwidth necessary for each application to satisfy its message-delivery delay requirement. The result obtained in this paper is essential for effective use of the FDDI token ring networks in supporting such real-time communication as digital video/audio transmissions, and distributed control/monitoring     

基于WiFi的语音矿灯设计

针对现有矿灯大多只具有照明、定位、环境感知等功能,没有语音对讲功能的问题,设计了一种基于WiFi的具有语音对讲功能的语音矿灯。该语音矿灯以工业以太环网和WiFi网络为传输平台,采用VoIP语音通信技术实现语音播放、音频采集、与调度台对讲功能;通过音频编解码芯片实现语音模拟信号与数字信号的转换,采用UDP协议将信号传输至调度台,完成语音数据的双向传输,实现语音对讲和矿灯照明一体化。详细介绍了语音对讲功能实现的关键技术:音频数据的编码格式和缓存管理、语音数据的可靠传输机制,用于确保语音播放的准确性;WiFi模块与微控制器STM32L151的低功耗休眠技术,用于降低语音矿灯平均电流,延长工作时间。测试结果表明:该语音矿灯能够满足调度台与井下工作人员之间的语音对讲需求,与WiFi基站通信距离可达400 m,与调度台之间的对讲传输时延小于1 s,语音矿灯之间的组播传输时延小于3 s;语音矿灯对讲时平均电流小于70 mA,空闲时平均电流小于5 mA。     

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.