Perceptive admission control for wireless network quality of service

As wireless networks become more widely used, there is a growing need to support advanced services, such as multimedia streaming and voice over IP. Traditional approaches to guarantee quality of service (QoS) work well only with predictable channel and network access. In wireless mobile networks, where conditions dynamically change as nodes move about the network, a stateless, high level approach is required. Since shared wireless resources are easily over-utilized, the load in the network must be controlled so that an acceptable QoS for real-time applications can be maintained. If minimum real-time requirements are not met, these unusable packets waste scarce bandwidth and hinder other traffic, compounding the problem. To enable high QoS for all admitted traffic, we propose the Perceptive Admission Control (PAC) protocol. PAC monitors the wireless channel and dynamically adapts admission control decisions to enable high network utilization while preventing congestion. Through discussion, simulations and testbed experiments, we demonstrate that PAC ensures low packet loss and delay for all admitted flows.     

Prevention of deadlocks and livelocks in lossless backpressured packet networks

No packets will be dropped inside a packet network, even when congestion builds up, if congested nodes send backpressure feedback to neighboring nodes, informing them of unavailability of buffering capacity-stopping them from forwarding more packets until enough buffer becomes available. While there are potential advantages in backpressured networks that do not allow packet dropping, such networks are susceptible to a condition known as deadlock in which throughput of the network or part of the network goes to zero (i.e., no packets are transmitted). In this paper, we describe a simple, lossless method of preventing deadlocks and livelocks in backpressured packet networks. In contrast with prior approaches, our proposed technique does not introduce any packet losses, does not corrupt packet sequence, and does not require any changes to packet headers. It represents a new networking paradigm in which internal network losses are avoided (thereby simplifying the design of other network protocols) and internal network delays are bounded.     

基于瞬时相位差和深度学习的丢包隐藏方法

实时IP 语音通信在数据包会丢失的情况下,语音质量会受到严重影响。为了恢复传输过程中丢失的语音信息,本文提出了一种基于瞬时相位差(Instantaneous Phase Deviation, IPD)和深度神经网络(Deep Neural Network, DNN)的丢包隐藏 (Packet Loss Concealment, PLC)方法。在训练阶段,将语音的对数功率谱(Log Power Spectrum, LPS)和IPD作为训练DNN的输入特征,以学习从接收包到丢失包的映射关系;在重构阶段,将丢包前接收到的语音包送入训练好的DNN中,恢复出丢失包的语音。实验结果表明,在不同丢包率下,所提方法的性能优于传统的基于LPS和DNN的PLC方法。      

基于Windows的网络安全监控系统的设计与实现

为了对付网络入侵行为,保障内网安全,设计了一个基于W indows的实时网络安全监控系统。本系统利用W insock抓包技术,在VB6.0环境中开发实现。在千兆网卡80%的负载环境下,对系统进行了应用测试,其平均丢包率小于0.05。实验结果显示,系统能有效监测进出网络的数据,对数据包进行严格访问控制,实现入侵检测和流量监测,具有良好的实时安全监控能力。     

A self-timed real-time sorting network

High-speed networks are expected to carry traffic classes with diverse quality of service (QoS) guarantees. For efficient utilization of resources, sophisticated scheduling protocols are needed; however, these must be implemented without sacrificing the maximum possible bandwidth. This paper presents the architecture and implementation of a self-timed real-time sorting network to be used in packet switches that support a diverse mix of traffic. The sorting network receives packets with appropriately assigned priorities and schedules the packets for departure in a highest-priority-first manner. The circuit implementation uses zero-overhead, self-timed, and self-precharging domino logic to minimize the circuit latency. An experimental sorting network chip has been designed using the techniques described in this paper to support 10 Gb/s links with ATM-size packets     

Distributed network embedded FEC for real-time multicast applications in multi-hop wireless networks

Multi-hop wireless networks are becoming popular because of their flexibility and low deployment cost. Emerging technologies such as orthogonal frequency division and multiple in and multiple out have significantly increased the bandwidth of a wireless channel. Further, as device cost decreases, a communication terminal can have multiple radios and transmit/receive data simultaneously, which improves the capacity of a wireless network. This makes the support of real-time multicast applications over multi-hop wireless networks viable and practical. Meanwhile, wireless links are prone to random and burst losses due to multipath fading and cross channel interference, real-time multicast over a wireless network remains a challenging problem. Traditional end-to-end FEC is less efficient in multi-hop wireless networks, as packets may suffer from random or burst losses in more than one hop before they arrive at their destination. In this paper, we advocate the deployment of distributed network-embedded FEC (DNEF) for real-time multicast distribution over multi-hop wireless networks. We first develop a packet loss model of multi-hop wireless networks using a system analysis approach. We then propose a distributed codec placement algorithm and evaluate its performance. Our simulation shows that multicast using DNEF significantly outperforms both traditional multicast and application-level peer-to-peer multicast that can be deployed over multi-hop wireless networks.     

A novel congestion control protocol with AQM support for IP-based networks

Multimedia services (Real-time and Non real-time) have different demands, including the need for high bandwidth and low delay, jitter and loss. TCP is a dominant protocol on the Internet. In order to have the best performance in TCP, the congestion window size must be set according to some parameters, since the TCP source is not aware of the window size. TCP emphasizes more on reliability than timeliness, so TCP is not suitable for real-time traffic. In this paper an active Queue management support TCP (QTCP) model is presented. Source rate is regulated based on the feedback which is received from intermediate routers. Furthermore, in order to satisfy the requirements of multimedia applications, a new Optimization Based active Queue management (OBQ) mechanism has been developed. OBQ calculates packet loss probabilities based on the queue length, packets priority and delay in routers and the results are sent to source, which can then regulate its sending rate. Simulation results indicate that the QTCP reduces packet loss and buffer size in intermediate nodes, improves network throughput and reduces delay.     

网络流量丢包率预测模型

无线信道的网络流量的及时预测,是目前国内外研究的前沿课题。通过对实时网络流量的测量分析,提出了一种基于最小二乘法的多项式拟合网络流量丢包率预测模型。该模型根据当前获得的网络流量信息,对网络丢包率趋势进行预测,为无线网络下信号丢包率的预测供了可行的手段。实测丢包率与预测丢包率的对比分析表明提出的预测模型能很好的预测丢包率趋势。     

Packet sequencing: a deterministic protocol for QoS in IP networks

We describe a deterministic protocol for routing delay and loss-sensitive traffic through an IP network. Unlike traditional approaches, the method described here - packet sequencing - does not rely on queue management. Instead, it uses a temporally-based deterministic protocol to coordinate and switch IP packets on a systemwide basis. As a result, end-to-end throughput is guaranteed, without packet loss, loss variance, or accumulated performance impairment; additionally, end-to-end delay is minimized, and jitter is essentially eliminated. We also show that packet sequencing can complement conventional IP networks: sequencing does not negate the use of queue management QoS methods that are the subject of considerable ongoing study. This article describes the fundamental approach, issues associated with scalability, illustrative performance in the context of storage networking, and attributes related to the security and reliability of IP networks.     

Dynamic Forwarding over Tree-on-DAG for Scalable Data Aggregation in Sensor Networks

Computing and maintaining network structures for efficient data aggregation incurs high overhead for dynamic events where the set of nodes sensing an event changes with time. Moreover, structured approaches are sensitive to the waiting-time which is used by nodes to wait for packets from their children before forwarding the packet to the sink. Although structure-less approaches can address these issues, the performance does not scale well with the network size. We propose ToD, a semi-structured approach that uses Dynamic Forwarding on an implicitly constructed structure composed of multiple shortest path trees to support network scalability. The key principle behind ToD is that adjacent nodes in a graph will have low stretch in one of these trees in ToD, thus resulting in early aggregation of packets. Based on simulations on a $2000$ nodes network and real experiments on a $105$ nodes Mica2-based network, we conclude that efficient aggregation in large scale networks can be achieved by our semi-structured approach.     

A new TCP for persistent packet reordering

Most standard implementations of TCP perform poorly when packets are reordered. In this paper, we propose a new version of TCP that maintains high throughput when reordering occurs and yet, when packet reordering does not occur, is friendly to other versions of TCP. The proposed TCP variant, or TCP-PR, does not rely on duplicate acknowledgments to detect a packet loss. Instead, timers are maintained to keep track of how long ago a packet was transmitted. In case the corresponding acknowledgment has not yet arrived and the elapsed time since the packet was sent is larger than a given threshold, the packet is assumed lost. Because TCP-PR does not rely on duplicate acknowledgments, packet reordering (including out-or-order acknowledgments) has no effect on TCP-PR's performance. Through extensive simulations, we show that TCP-PR performs consistently better than existing mechanisms that try to make TCP more robust to packet reordering. In the case that packets are not reordered, we verify that TCP-PR maintains the same throughput as typical implementations of TCP (specifically, TCP-SACK) and shares network resources fairly. Furthermore, TCP-PR only requires changes to the TCP sender side making it easier to deploy.     

Threshold-based selective drop for shared buffer packet switches

We propose a novel buffer management scheme called threshold-based selective drop (TSD) to improve the overall loss performance and fairness by regulating the buffer sharing in a packet switch. A transient analysis of TSD is derived to prove the fairness of buffer allocation. Computer simulation shows that the overall loss performance of TSD approaches to the pushout (PO) scheme, which is considered as an optimal solution with implementation difficulties in high-speed Internet. However, unlike the PO, the TSD will block the unwanted packets before they enter the queue, and does not need to pre-empty the queue for accepting new packets.     

Analysis of packet loss processes in high-speed networks

The packet loss process in a single-server queueing system with a finite buffer capacity is analyzed. The model used addresses the packet loss probabilities for packets within a block of a consecutive sequence of packets. An analytical approach is presented that yields efficient recursions for the computation of the distribution of the number of lost packets within a block of packets of fixed or variable size for several arrival models and several numbers of sessions. Numerical examples are provided to compare the distribution obtained with that obtained using the independence assumption to compute the loss probabilities of packets within a block. The results show that forward error correction schemes become less efficient due to the bursty nature of the packet loss processes; real-time traffic might be more sensitive to network congestion than was previously assumed; and the retransmission probability of ATM messages has been overestimated by the use of the independence assumption     

A statistical trust system in wireless mesh networks

Most trust and reputation solutions in wireless mesh networks (WMNs) rely on the intrusion detection system (IDS) Watchdog. Nevertheless, Watchdog does not consider packet loss on wireless links and may generate false positives. Consequently, a node that suffers from packet loss on one of its links may be accused wrongly, by Watchdog, of misbehaving. To deal with this issue, we propose in this paper a novel trust system which considers packet loss of links. Our trust system is based on a statistical detection method (SDM) implemented on each node of the network. Firstly, the SDM, via CUSUM test, analyzes the behavior of the packets loss in order to detect a dropping attack. Secondly, the SDM, through the Kolmogorov-Smirnov test, compares the behavior of the total packets loss with that of the control packets in order to identify the attack type. Our system allows every WMN’s node to assign to each of its neighbors, a trust value which reflects its real behavior. We have validated the proposed SDM method via extensive simulations on ns2 and have compared our trust system with an existing solution. The results display that our SDM solution offers better performance.     

QoS evaluation of sender-based loss-recovery techniques for VoIP

Voice over Internet protocol (VoIP) is a technology that transports voice data packets across packet-switched networks using the Internet protocol (IP). Losing packets in the network is inevitable, and losing voice packets degrades audio quality. There are many loss-recovery techniques that designers can use to mitigate the undesired effects of packet loss. Some of these loss-recovery techniques use sender-based procedures, and others use receiver-based procedures. We examine several well-known sender-based loss-recovery techniques and evaluate the feasibility and effectiveness of each one in real-time interactive VoIP applications. We analyze the bandwidth requirements, buffering delays, and perceptual sound qualities of these techniques. We study the effectiveness of these approaches under various packet-loss conditions, and we also compare the effectiveness of these techniques against a speech codec that has high degree of packet-loss robustness     

Study of information loss in packet voice systems

Once a voice buffer is full, it remains full for a certain period, during which many packets are possibly blocked, resulting in consecutive clippings in voice. The packet loss rate during this period changes slowly and has large fluctuations. It is shown that the temporal behavior of packet loss, especially at high rate, is inherently determined by voice correlation and system capacity and is independent of buffer size. Buffering may reduce the occurrence of short blocking periods associated with low rates packet loss but does not affect long ones associated with high packet loss rates. In fact, increasing the buffer size merely extends nonblocking periods, and thereby reduces the overall average packet loss rate, but packet-loss performance within existing blocking periods is not significantly improved. A simple tool is developed for calculating the boundary performance. It is found that it is possible to design a packet-switched voice system without buffering only at the expense of supporting a fewer number of calls. The issue of voice delay allocation between source and network is discussed, and it is shown that it is more effective to keep the network delay short while extending the source delay     

Application layer error-correction coding for rate-distortion optimized streaming to wireless clients

This paper addresses the problem of streaming packetized media over a lossy packet network to a wireless client, in a rate-distortion optimized way. We introduce an incremental redundancy error-correction scheme that combats the effects of both packet loss and bit errors in an end-to-end fashion, without support from the underlying network or from an intermediate base station. The scheme is employed within an optimization framework that enables the sender to compute which packets it should send, out of all the packets it could send at a given transmission opportunity, in order to meet an average transmission-rate constraint while minimizing the average end-to-end distortion. Experimental results show that our system is robust and maintains quality of service over a wide range of channel conditions. Up to 8 dB performance gains are registered over systems that are not rate-distortion optimized, at bit-error rates as large as 10/sup -2/.     

Performance evaluation of a protocol for packet radio network in mobile computer communications

The need to provide computer network access to mobile terminals and computer communications in the mobile environment has stimulated and motivated the current developments in this area. Packet radio technology has developed over the past decade in response to the need for real-time, interactive communications among mobile users and shared computer resources. In computer communication systems we have a great need for sharing expensive resources among a collection of high peak-to-average (i.e., bursty) users. Packet radio networks provide an effective way to interconnect fixed and mobile resources. The results of an attempt to study the performance of the mobile packet radio network for computer communications over degraded channels are presented. We develop a model under fading conditions and derive a protocol for evaluating the performance of the mobile packet radio network (MPRNET) in terms of the packet error rate, packet delay, throughput and average number of retransmitted packets per cycle. The analytical results are presented and numerical examples are given to illustrate the behavior of these performance criteria as a function of packet transmission rate, packets transmitted per cycle, packet size, and vehicle speed with the help of appropriate plots.     

A slotted frequency-hopped multiple-access network with packetcombining

A frequency-hopped spread-spectrum (FH/SS) packet radio network is considered. Instead of discarding the received packets which were unsuccessfully decoded, the receiver keeps and combines the different received copies of the same packet to improve the throughput and reduce the delay necessary to transmit a packet successfully. It is shown that the throughput is substantially increased over a similar system where the uncorrectable packets are discarded. This system is equivalent to a fast frequency-hopped system where the diversity level is adapted according to the channel conditions. Reed-Solomon (RS) codes are used for error correction and detection     

An optical router with multistage management functions for asynchronous optical packet switching network

An optical router with multistage distributed management features for the asynchronous optical packet switching (OPS) network is presented, which can improve switching capacity and all-optical scalability. A compact recycling-fiber-delay-line (Rec-FDL) based collision resolution mechanism is proposed to resolve the contentions for asynchronous and variable length optical packets. The analysis models of stabilities, packet loss rates (PLR) and average packet waiting latencies (PWL) for the router are developed based on the timer based optical packet assembly algorithm. The simulation shows that PLR and PWL for a 400-byte optical packet transmitted in the 32 wavelengths dense wavelength division multiplexing (DWDM) system equal to 3.48 × 10⁻⁴ and 0.072 ns, respectively. The non-blocking switching can be realized for the packets with lengths less than the buffer granularity of the Rec-FDL, and the optimized performance for the proposed router can be obtained through properly selecting of the system parameters.