Variable bit-rate coding of video signals for ATM networks

Statistical characteristics of video signals for video packet coding, are clarified and a variable-bit-rate coding method for asynchronous transfer mode (ATM) networks is described that is capable of compensating for packet loss. ATM capabilities are shown to be greatly affected by delay, delay jitter, and packet loss probability. Packet loss has the greatest influence on picture quality. Packets may be lost either due to random bit error in a cell header or to network control when traffic is congested. A layered coding technique using discrete-cosine transform (DCT) coding is presented which is suitable for packet loss compensation. The influence of packet loss on picture quality is discussed, and decoded pictures with packet loss are shown. The proposed algorithm was verified by computer simulations     

CAPF: coded anycast packet forwarding for wireless mesh networks

In wireless mesh networks, delay and reliability are two critical issues in the support of delay-sensitive applications. Due to sleep scheduling designed for energy efficiency, a node along an end-to-end path needs to wait for its next hop to wake up before it can transmit, which incurs extra delay. In addition, because of unreliable wireless communications, a node may not successfully receive the packet even when it is in active mode. In this paper, we propose a coded anycast packet forwarding (CAPF) scheme for both unicast and multicast communications such that the delay can be reduced and the reliability can be improved. We theoretically analyze the impact of nodes’ awake probability and the link loss probability on the end-to-end delay and the reliability. A tradeoff between the end-to-end delay and the reliability is also investigated. Simulation results demonstrate that CAPF provides a flexible mechanism to make good delay-reliability tradeoff and is effective to reduce the end-to-end delay and enhance the reliability.     

Interleaved coding and sequential transmission scheme for packet‐level forward error correction over burst loss channels

Burst packet loss is a common problem over wired and wireless networks and leads to a significant reduction in the performance of packet‐level forward error correction (FEC) schemes used to recover packet losses during transmission. Traditional FEC interleaving methods adopt the sequential coding‐interleaved transmission (SCIT) process to encode the FEC packets sequentially and reorder the packet transmission sequence. Consequently, the burst loss effect can be mitigated at the expense of an increased end‐to‐end delay. Alternatively, the reversed interleaving scheme, namely, interleaved coding‐sequential transmission (ICST), performs FEC coding in an interleaved manner and transmits the packets sequentially based on their generation order in the application. In this study, the analytical FEC model is constructed to evaluate the performance of the SCIT and ICST schemes. From the analysis results, it can be observed that the interleaving delay of ICST FEC is reduced by transmitting the source packets immediately as they arrive from the application. Accordingly, an Enhanced ICST scheme is further proposed to trade the saved interleaving time for a greater interleaving capacity, and the corresponding packet loss rate can be minimized under a given delay constraint. The simulation results show that the Enhanced ICST scheme achieves a lower packet loss rate and a higher peak signal‐to‐noise‐ratio than the traditional SCIT and ICST schemes for video streaming applications.     

Small‐block interleaving for low‐delay cross‐packet forward error correction over burst‐loss channels

By adding the redundant packets into source packet block, cross‐packet forward error correction (FEC) scheme performs error correction across packets and can recover both congestion packet loss and wireless bit errors accordingly. Because cross‐packet FEC typically trades the additional latency to combat burst losses in the wireless channel, this paper presents a FEC enhancement scheme using the small‐block interleaving technique to enhance cross‐packet FEC with the decreased delay and improved good‐put. Specifically, adopting short block size is effective in reducing FEC processing delay, whereas the corresponding effect of lower burst‐error correction capacity can be compensated by deliberately controlling the interleaving degree. The main features include (i) the proposed scheme that operates in the post‐processing manner to be compatible with the existing FEC control schemes and (ii) to maximize the data good‐put in lossy networks; an analytical FEC model is built on the interleaved Gilbert‐Elliott channel to determine the optimal FEC parameters. The simulation results show that the small‐block interleaved FEC scheme significantly improves the video streaming quality in lossy channels for delay‐sensitive video. Copyright © 2013 John Wiley & Sons, Ltd.     

On the effects of the packet size distribution on the packet loss process

Real-time multimedia applications have to use forward error correction (FEC) anderror concealment techniques to cope with losses in today’s best-effort Internet. The efficiency of these solutions is known however to depend on the correlation between losses in the media stream. In this paper we investigate how the packet size distribution affects the packet loss process, that is, the distribution of the number of lost packets in a block, the related FEC performance and the average loss run length. We present mathematical models for the loss process of the MMPP+M/D/1/K and the MMPP+M/M/1/K queues; we validate the models via simulations, and compare the results to simulation results with an MPEG-4 coded video trace. We conclude that the deterministic packet size distribution (PSD) not only results in lower stationary loss probability than the exponential one, but also gives a less correlated loss process, both at a particular average link load and at a particular stationary loss probability as seen by the media stream.Our results show that for applications that can only measure the packet loss probability, the effects of the PSD on FEC performance are higher in access networks, where a single multimedia stream might affect the multiplexing behavior. Our results show that the effects of the PSD on FEC performance are higher in access networks, where a single multimedia stream might affect the multiplexing behavior and thus can improve the queuing performance by decreasing the variance of its PSD.     

ATM网络中漏桶算法在突发业务输入时的性能分析

本文首先引入了描述分组图像业务的突发业务模型;分析了漏桶算法在突发业务输入时的性能;研究了各参数对业务服务质量的影响。研究结果表明,增大缓冲器容量可以降低信元丢失率,但会增大时延和时延抖动。     

一种新的基于网络丢包率的主动队列管理策略

提出了一种新的基于网络丢包率的动态自适应的主动队列管理的改进算法。该算法首先用早期网络的丢包率标记到达的数据包,并作为数据包的丢弃概率。这样使得到达数据包的丢弃概率逼近当前实际网络的丢包率,然后经过自适应调整丢弃概率使得缓冲队列长度保持在一定范围内。通过NS仿真实验表明了该算法可以很好的控制队列长度,降低延迟抖动。     

PERFORMANCE ANALYSIS OF LEAKY BUCKET ALGORITHM WITH BURSTY TRAFFIC INPUT IN ATM NETWORKS

A bursty traffic model is introduced in this paper to describe the statistical characteristics of packet video. The performance of leady bucket algorithm with bursty traffic input is analyzed. The influences of various parameters on QOS (Quality of Service) are investigated. The analysis shows that although the loss probability decreases through expanding the buffer capacity, the delay and delay jitter increase, whose effect on QOS will not be negligible.     

软件定义卫星光网络蚁群优化波长路由技术

针对卫星光网络中网络拓扑动态时变和业务类型多样化的问题,研究了在软件定义网络架构下保障服务质量的路由技术,提出了一种基于多业务的卫星光网络蚁群优化波长路由算法。通过改进蚁群算法的启发函数,将波长空闲率、时延、时延抖动、丢包率作为蚂蚁选路的重要依据,为业务选择了满足多种服务质量的最优路径;采用分组波长分配方法对不同等级的业务进行了区分服务,为不同业务分配了不同的波长集。仿真结果表明:与CL-ACRWA算法和Dijkstra算法相比,降低了卫星光网络的平均时延、平均时延抖动、平均丢包率,提高了波长利用率,同时也降低了高优先级业务的网络拥塞概率。     

Denial of service attacks on network-based control systems: impact and mitigation

Replacing specialized industrial networks with the Internet is a growing trend in industrial informatics, where packets are used to transmit feedback and control signals between a plant and a controller. Today, denial of service (DoS) attacks cause significant disruptions to the Internet, which will threaten the operation of network-based control systems (NBCS). In this paper, we propose two queueing models to simulate the stochastic process of packet delay jitter and loss under DoS attacks. The motivation is to quantitatively investigate how these attacks degrade the performance of NBCS. The example control system consists of a proportional integral controller, a second-order plant, and two one-way delay vectors induced by attacks. The simulation results indicate that Model I attack (local network DoS attack) impairs the performance because a large number of NBCS packets are lost. Model II attack (nonlocal network DoS attack) deteriorates the performance or even destabilizes the system. In this case, the traffic for NBCS exhibits strong autocorrelation of delay jitter and packet loss. Mitigating measures based on packet filtering are discussed and shown to be capable of ameliorating the performance degradation.     

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.     

Improving link layer performance on satellite channels with shadowing via delayed two-copy selective repeat ARQ

This paper focuses on improving performance of land mobile satellite channels (LMSCs) at high band (Ka-band or EHF band), where shadowing is the primary impediment to reliable data transmission. Compared with multipath fading, shadowing exists on a longer time scale; hence, interleaving to combat shadowing introduces unacceptably large decoding delay. We use Lutz's model to investigate bit-error rate/packet-error rate (BER/PER) performance of interleaving with various forward error correction (FEC) coding as a function of different channel parameters to demonstrate its limited effectiveness for combatting burst errors whose mean duration significantly exceed a link layer (LL) packet. We propose a delayed two-copy selective repeat ARQ (DTC-SR-ARQ) scheme, whereby two copies of a packet are sent-the second with a delay relative to the first-in every transmission or retransmission. Closed-form expressions for mean transmission time, success probability, and residual loss probability are provided and simulations used to validate the analysis. Furthermore, the issue of optimum delay is addressed as well, and a simple yet effective strategy is suggested to support transmission control protocol (TCP) traffic over this data link layer. DTC-SR-ARQ is shown to achieve much shorter additional delay than interleaving and compared with normal SR-ARQ, reduces mean transmission time at expense of a small increase in residual packet loss probability. Furthermore, ns2 simulation results show that for TCP traffic, DTC-SR-ARQ acquires higher end-to-end throughput than normal SR-ARQ.     

Performance of Transport Protocols for Multimedia Communications in Wireless Sensor Networks

Many wireless sensor network (WSN) applications require efficient multimedia communication capabilities. However, the existing communication protocols in the literature mainly aim to achieve energy efficiency and reliability objectives and do not address multimedia communication challenges in WSN. In this paper, comprehensive performance evaluation of the existing transport protocols is performed for multimedia communication in WSN. Performance metrics such as packet delivery rate, end-to-end packet delay, bandwidth and energy efficiency, frame peak signal-to-noise ratio (PSNR), delay-bounded frame PSNR, frame delivery probability, frame end-to-end delay and jitter are investigated. The results clearly show that the existing transport protocols are far from satisfying the requirements of multimedia communication in WSN and hence there is a need for new effective multimedia delivery protocols for WSN.     

一种环境感知的无线Mesh网络自适应QoS路径选择算法

本文针对如何改善无线多跳Mesh网络的服务质量,满足无线多媒体业务对数据传输的带宽、时延、抖动的要求等问题,研究了一种基于无线信道状态和链路质量统计的MAC层最大重传次数的自适应调整算法。该算法通过对无线Mesh网络的无线信道环境的动态感知,利用分层判断法区分无线分组丢失的主要原因是无线差错还是网络拥塞导致,实时调整MAC层的最佳重传次数,降低无线网络中的分组冲突概率。基于链路状态信息的统计和最大重传策略,提出了一种启发式的基于环境感知的QoS路由优化机制HEAOR。该算法通过动态感知底层链路状态信息,利用灰色关联分析法自适应选择最优路径,在不增加系统复杂度的基础上,减少链路误判概率,提高传输效率。NS2仿真结果表明,HEAOR算法能有效减少重路由次数,降低链路失效概率,提高网络的平均吞吐率。本文提出的方法不仅能够优化MAC层的重传,而且通过发现跨层设计的优化参数实现对路径的优化选择。      

Delay jitter bounds and packet scale rate guarantee for expedited forwarding

We consider the definition of the expedited forwarding per-hop behavior (EF PHB) as given in RFC 2598 and its impact on worst case end-to-end delay jitter. On the one hand, the definition in RFC 2598 can be used to predict extremely low end-to-end delay jitter, independent of the network scale. On the other hand, the worst case delay jitter can be made arbitrarily large, while each flow traverses at most a specified number of hops, if we allow networks to become arbitrarily large; this is in contradiction with the previous statement. We analyze where the contradiction originates and find the explanation. It resides in the fact that the definition in RFC 2598 is not easily implementable in known schedulers, mainly because it is not formal enough and also because it does not contain an error term. We propose a new definition for the EF PHB, called "packet scale rate guarantee" (PSRG) that preserves the spirit of RFC 2598 while allowing a number of reasonable implementations and has very useful properties for per-node and end-to-end network engineering. We show that this definition implies a rate-latency service curve property. We also show that it is equivalent, in some sense, to the stronger concept of "adaptive service guarantee". Then we propose some proven bounds on delay jitter for networks implementing this new definition in cases without loss and with loss.     

A novel adaptive traffic prediction AQM algorithm

In the Internet, network congestion is becoming an intractable problem. Congestion results in longer delay, drastic jitter and excessive packet losses. As a result, quality of service (QoS) of networks deteriorates, and then the quality of experience (QoE) perceived by end users will not be satisfied. As a powerful supplement of transport layer (i.e. TCP) congestion control, active queue management (AQM) compensates the deficiency of TCP in congestion control. In this paper, a novel adaptive traffic prediction AQM (ATPAQM) algorithm is proposed. ATPAQM operates in two granularities. In coarse granularity, on one hand, it adopts an improved Kalman filtering model to predict traffic; on the other hand, it calculates average packet loss ratio (PLR) every prediction interval. In fine granularity, upon receiving a packet, it regulates packet dropping probability according to the calculated average PLR. Simulation results show that ATPAQM algorithm outperforms other algorithms in queue stability, packet loss ratio and link utilization.     

KALP: a Kalman filter-based adaptive clock method with low-passprefiltering for packet networks use

We consider the issue of source clock frequency recovery in packet networks and propose a new adaptive clock method based on the Kalman filter (KF) with low-pass prefiltering-KALP (Kalman filter-based Adaptive clock method with Low-pass Prefiltering) in short. Noting that because of the difficulty in modeling as well as the nonwhite Gaussian nature of the packet jitter most existing adaptive clock methods could not successfully adopt the Kalman filter, we take a new approach to packet jitter modeling for the KALP. We model the packet jitter not directly but after shaping its characteristics by low-pass prefiltering. This low-pass prefiltering is an important arrangement as it helps to convert the packet jitter into a low-pass signal regardless of its original characteristics, thus enabling to model the prefiltered packet jitter using a simple first-order autoregressive [AR(1)] process. The low-pass prefilter should be carefully selected not to lose the timing information while prefiltering, and the moving averager employed in this paper satisfies this requirement. The AR(1)-modeled jitter component is amenable to the KF-based processing, which in this case becomes an optimal estimator. The design parameters including the initial conditions of the KF and AR(1) parameters can be determined based on the service clock specification and packet interarrival times during the delay smoothing process. We carry out various simulations to compare the performance of the KALP with the existing buffer-based adaptive clock method and demonstrate that the KALP can significantly reduce the fluctuation in the level of receiving buffer as well as the time to recover the source clock frequency     

Maximum-likelihood decoding and code combining for DS/SSMA slottedALOHA

This paper considers the combination of multiple copies of a packet to improve the performance of a slotted direct-sequence spread-spectrum multiple-access (DS/SSMA) ALOHA packet radio system with coherent binary phase-shift keying (BPSK) modulation. Both slotted DS/SSMA ALOHA with and without forward error correction (FEC) are considered. For the case with FEC, maximum-likelihood decoding with code combining is used. Code combining allows for the combination of multiple copies of the same packet (which are typically discarded), to obtain a lower code rate for that specific packet, and therefore an improved probability of successful decoding. In both cases, combining multiple copies of the same packet results in a throughput which is an increasing function over a broad range of offered load, so that the system is more reliable from the point of view of stability. In addition, combining provides a higher throughput and a smaller time delay for packet transmission. This is illustrated by means of analytical and simulation results     

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.