Video coding with fixed-length packetization for a tandem channel.

A robust scheme is presented for the efficient transmission of packet video over a tandem wireless Internet channel. This channel is assumed to have bit errors (due to noise and fading on the wireless portion of the channel) and packet erasures (due to congestion on the wired portion). First, we propose an algorithm to optimally switch between intracoding and intercoding for a video coder that operates on a packet-switched network with fixed-length packets. Different re-synchronization schemes are considered and compared. This optimal mode selection algorithm is integrated with an efficient channel encoder, a cyclic redundancy check outer coder concatenated with an inner rate-compatible punctured convolutional coder. The system performance is both analyzed and simulated. Last, the framework is extended to operate on a time-varying wireless Internet channel with feedback information from the receiver. Both instantaneous feedback and delayed feedback are evaluated, and an improved method of refined distortion estimation for encoding is presented and simulated for the case of delayed feedback.     

Accumulation-based congestion control

This paper generalizes the TCP Vegas congestion avoidance mechanism and uses accumulation , buffered packets of a flow inside network routers, as a congestion measure based on which a family of congestion control schemes can be derived. We call this model Accumulation-based Congestion Control (ACC), which fits into the nonlinear optimization framework proposed by Kelly. The ACC model serves as a reference for packet-switching network implementations. We show that TCP Vegas is one possible scheme under this model. It is well known that Vegas suffers from round trip propagation delay estimation error and reverse path congestion. We therefore design a new Monaco scheme that solves these problems by employing an out-of-band, receiver-based accumulation estimator, with the support of two FIFO priority queues from the (congested) routers. Comparisons between these two schemes demonstrate that Monaco does not suffer from the problems mentioned above and achieves better performance than Vegas. We use ns-2 simulations and Linux implementation experiments to show that the static and dynamic performance of Monaco matches the theoretic results. One key issue regarding the ACC model in general, i.e., the scalability of bottleneck buffer requirement, and a solution using a virtual queueing algorithm are discussed and evaluated.     

A Timeout-Based Congestion Control Scheme for Window Flow-Controlled Networks

During overload, most networks drop packets due to buffer unavailability. The resulting timeouts at the source provide an implicit mechanism to convey congestion signals from the network to the source. On a timeout, a source should not only retransmit the lost packet, but it should also reduce its load on the network. Basedon this realization, we have developed a simple congestion control scheme using the acknowledgment timeouts as indications of packet loss and congestion. This scheme does not require any new message formats, therefore, it can be used in any network with window flow control, e.g., ARPAnet or ISO.     

On network bandwidth sharing for transporting rate‐adaptive packet video using feedback

While existing research shows that feedback‐based congestion control mechanisms are capable of providing better video quality and higher link utilization for rate‐adaptive packet video, there has been relatively little study on how to share network bandwidth among competing rate‐adaptive video connections, when feedback control is used in a fully distributed network. This paper addresses this issue by presenting a framework of network bandwidth sharing for transporting rate‐adaptive packet video using feedback. We show how a weight‐based bandwidth sharing policy can be used to allocate network bandwidth among competing video connections and design a feedback control algorithm using an Available Bit Rate (ABR)‐like flow control mechanism. A novel video source rate adaptation algorithm is also introduced to decouple a video source's actual transmission rate from the rate used for distributed protocol convergence. Our feedback control algorithm provides guaranteed convergence and smooth source rate adaptation to our weight‐based bandwidth sharing policy under any network configuration and any set of link distances. Finally, we show the on‐line minimum rate renegotiation and weight adjustment options in our feedback control algorithm, which offer further flexibility in network bandwidth sharing for video connections. Copyright © 2000 John Wiley & Sons, Ltd.     

Scalable and reliable ATM multicast employing RM cell consolidation

The authors present a new scalable and reliable ATM multicast algorithm (SRAM) that can provide reliable data transfer services. This algorithm employs the resource management cell consolidation mechanism for available bit rate point-to-multipoint connections which resolves the feedback implosion problem without increasing the number of connections according to the number of receivers. The proposed scheme is compared with other reliable multicast schemes through numerical analysis under reasonable assumptions     

Fair packet discarding for controlling ABR traffic in ATM networks

The asynchronous transfer mode (ATM) Forum has chosen rate-based control as the flow control scheme for the available bit-rate (ABR) service. However, rate-based schemes can achieve congestion control only if all users act in a cooperative manner. Even a limited number of uncooperative users can cause congestion collapse. We propose a mechanism called fair packet discarding to provide incentives to users to participate in network congestion control so that the network can operate in a more efficient manner     

Predictive Congestion Control Protocol for Wireless Sensor Networks

Available congestion control schemes, for example transport control protocol (TCP), when applied to wireless networks, result in a large number of packet drops, unfair scenarios and low throughputs with a significant amount of wasted energy due to retransmissions. To fully utilize the hop by hop feedback information, this paper presents a novel, decentralized, predictive congestion control (DPCC) for wireless sensor networks (WSN). The DPCC consists of an adaptive flow and adaptive back-off interval selection schemes that work in concert with energy efficient, distributed power control (DPC). The DPCC detects the onset of congestion using queue utilization and the embedded channel estimator algorithm in DPC that predicts the channel quality. Then, an adaptive flow control scheme selects suitable rate which is enforced by the newly proposed adaptive backoff interval selection scheme. An optional adaptive scheduling scheme updates weights associated with each packet to guarantee the weighted fairness during congestion. Closed-loop stability of the proposed hop-by-hop congestion control is demonstrated by using the Lyapunov-based approach. Simulation results show that the DPCC reduces congestion and improves performance over congestion detection and avoidance (CODA) [3] and IEEE 802.11 protocols.     

Active queue management with flow proportional buffering

Most active queue management schemes maintain an average of the queue length which they use together with a number of queue thresholds to detect congestion. However, the setting of the queue thresholds is problematic because the required buffer size for good sharing among TCP connections is dependent on the number of TCP connections using the buffer. This paper describes an improved active queue management scheme which dynamically changes its threshold settings as the number of connections and system load changes. This technique allows network devices to effectively control packet losses and TCP timeouts while maintaining high link utilization. Copyright © 2003 John Wiley &Sons, Ltd.     

信道公平分配的局部拥塞控制算法

提出基于信道公平分配的局部拥塞控制算法FCA(fair channel allocation),在缓解局部拥塞的同时增强信道分配的公平性。为减少获取邻居节点实时缓存信息的通信开销和提高以单一节点缓存是否溢出为检测模型的准确性,FCA采用以节点实时缓存长度预测为基础的邻居节点缓存总长度和分组平均传输延迟作为检测指标的拥塞检测模型。为避免使用独立拥塞通告消息增加信道负载,FCA采用在ACK控制帧中增加一个节点地址位携带拥塞信息。在去拥塞阶段,FCA采用基于实时缓存长度和队列优先权值的信道分配机制保证公平传输和防止部分节点因缓存增速过快导致溢出分组丢失。实验结果表明,FCA在碰撞次数、分组传递率、吞吐量和公平性等方面相比802.11、CODA和PCCP具有显著优势。     

A fuzzy-logic control algorithm for active Queue Management in IP networks

Active Queue Management （AQM） is an active research area in the Internet community. Random Early Detection （RED） is a typical AQM algorithm, but it is known that it is difficult to configure its parameters and its average queue length is closely related to the load level. This paper proposes an effective fuzzy congestion control algorithm based on fuzzy logic which uses the predominance of fuzzy logic to deal with uncertain events. The main advantage of this new congestion control algorithm is that it discards the packet dropping mechanism of RED, and calculates packet loss according to a preconfigured fuzzy logic by using the queue length and the buffer usage ratio. Theoretical analysis and Network Simulator （NS） simulation results show that the proposed algorithm achieves more throughput and more stable queue length than traditional schemes. It really improves a router＇s ability in network congestion control in IP network.     

Cooperative congestion control schemes in ATM networks

One of the basic problems faced in the design of efficient traffic and congestion control schemes is related to the wide variety of services with different traffic characteristics and quality of service (QoS) requirements supported by ATM networks. The authors propose a new way of organizing the control system so that complexity is easier to manage. The multi-agent system approach, which provides the use of adaptative and intelligent agents, is investigated. The authors show, through the two congestion control schemes proposed, how to take advantage of using intelligent agents to increase the efficiency of the control scheme. First, TRAC (threshold based algorithm for control) is proposed, which is based on the use of fixed thresholds which enables the anticipation of congestion. This mechanism is compared with the push-out algorithm and it is shown that the authors' proposal improves the network performance. Also discussed is the necessity of taking into account the network dynamics. In TRAC, adaptative agents with learning capabilities are used to tune the values of the thresholds according to the status of the system. However, in this scheme, when congestion occurs, the actions we perform are independent of the nature of the traffic. Subsequently, we propose PATRAC (predictive agents in a threshold based algorithm for control) in which different actions are achieved according to the QoS requirements and to the prediction of traffic made by the agents. Specifically, re-routing is performed when congestion is heavy or is expected to be heavy and the traffic is cell loss sensitive. This re-routing has to deflect the traffic away from the congestion point. In this scheme, we propose a cooperative and predictive control scheme provided by a multi-agent system that is built in to each node     

An adaptive congestion control scheme for real time packet videotransport

We show that modulating the source rate of a video encoder based on congestion signals from the network has two major benefits: the quality of the video transmission degrades gracefully when the network is congested and the transmission capacity is used efficiently. Source rate modulation techniques have been used in the past in designing fixed rate video encoders used over telephone networks. In such constant bit rate encoders, the source rate modulation is done using feedback information about the occupancy of a local buffer. Thus, the feedback information is available instantaneously to the encoder. In the scheme proposed, the feedback may be delayed by several frames because it comes from an intermediate switching node of a packet switched network. The paper shows the proposed scheme performs quite well despite this delay in feedback. We believe the use of such schemes will simplify the architecture used for supporting real time video services in future nationwide gigabit networks     

Estimation and Prediction‐Based Connection Admission Control in Broadband Satellite Systems

We apply a “sliding‐window” Maximum Likelihood (ML) estimator to estimate traffic parameters of On‐Off source and develop a method for estimating stochastic predicted individual cell arrival rates. Based on these results, we propose a simple Connection Admission Control (CAC) scheme for delay sensitive services in broadband onboard packet switching satellite systems. The algorithms are motivated by the limited onboard satellite buffer, the large propagation delay, and low computational capabilities inherent in satellite communication systems. We develop an algorithm using the predicted individual cell loss ratio instead of using steady state cell loss ratios. We demonstrate the CAC benefits of this approach over using steady state cell loss ratios as well as predicted total cell loss ratios. We also derive the predictive saturation probability and the predictive cell loss ratio and use them to control the total number of connections. Predictive congestion control mechanisms allow a satellite network to operate in the optimum region of low delay and high throughput. This is different from the traditional reactive congestion control mechanism that allows the network to recover from the congested state. Numerical and simulation results obtained suggest that the proposed predictive scheme is a promising approach for real time CAC.     

Seamless Handover of IPTV Streams in a Wireless LAN Network

A robust mechanism to enable seamless handover of streamed IPTV in a WLAN is presented. Handover in a wireless network is usually based on signal strength measurements, but that approach does not consider levels of congestion within the network. Here, the case of stationary nodes with varying levels of network congestion is considered. A scheme that analyzes the jitter is used to establish the relationship between congestion and loss in WLANs. This Moving Average of Negative Jitter is used as the basis of a handover scheme which can minimize loss. The handover scheme uses a client with two simultaneous connections to the same server through two separate WLANs, and it is shown that the client can compare the jitter in the two streams to determine which network delivers the best performance; this information is then used to determine when to perform a handover. The proposed scheme is implemented and results are presented that show the successful handover of an RTP over UDP stream in a live WLAN environment. The test scenario used is aimed at “Over-The-Top” service delivery, but the core algorithm is expected to be more broadly applicable.      

ATM多播ABR业务拥塞控制的公平准则探讨

本文介绍了在ATM网络中多播ABR（Available Bit Rate）业务的基于ER（Explicit Cell Rate）并采用一种新的公平准则的拥塞控制模式,分析并推导出了在既有单播又有多播、实际情况MCR（Minimum Cell Rate）不为零时缓存长度的递推公式,并说明了网络的稳定性。在这种以平准则下,一个连接的ACR（Allowable Cell Rate）与该连接的PCR（Peak Cell Rate）与MCR的差成比例。而且,分支连接的剩余带宽容量被分配给单播连接,网络资源的利用达到最大化。     

Engineering ATM networks for congestion avoidance

With the combination of telecommunication, entertainment and computer industries, computer networking is adopting a new method called Asynchronous Transfer Mode (ATM) networking. Congestion control plays an important role in the effective and stable operation of ATM networks. Traffic management concerns with the design of a set of mechanisms which ensure that the network bandwidth, buffer and computational resources are efficiently utilized while meeting the various Quality of Service (QoS) guarantees given to sources as part of a traffic contract. In this paper, the most widely recognized congestion control schemes for ABR service are investigated. Some of these schemes show either lack of scalability or fairness while other well‐behaved schemes may require a highly complex switch algorithm that is unsuitable for implementation in cell‐switching high‐speed ATM networks. A new and improved congestion control scheme is proposed to support the best‐effort ABR traffic. This algorithm provides the congestion avoidance ability with high throughput and low delay, in addition to achieving the max–min fairness allocation. This revised version was published online in June 2006 with corrections to the Cover Date.     

一种支持多媒体通信QoS的拥塞控制机制

本文针对Internet传输协议TCP的和式增加积式减少(AIMD)拥塞控制机制不适应多媒体通信,而目前拥塞控制的研究又大多集中在尽量做好(Best-effort)服务上的问题,结合Internet上多媒体通信的特点及其对QoS的要求,提出了一种将多媒体通信服务质量(QoS)控制和基于速率拥塞控制结合起来的拥塞控制的新机制.本文详细地研究了这一机制,并提出了源端多媒体数据流的带宽控制策略、基于动态部分缓存共享(DPBS)的数据包丢失控制方案和接收端计算包丢失率p的方法.最后给出了整个拥塞控制机制的系统结构.     

Convergence of REM flow control at a single link

Various TCP congestion control schemes can be interpreted as approximately carrying out a certain basic algorithm to maximize aggregate source utility, different schemes corresponding to different choices of utility functions. The basic algorithm consists of a link algorithm that updates a congestion measure based on its traffic load, and a source algorithm that adapts the source rate to congestion in its path. Though convergent, this algorithm can lead to large equilibrium backlogs. This problem can be eliminated by modifying the basic algorithm to include backlog in the update of the congestion measure. This article proves that the modified algorithm converges when the network can be modeled as a single bottleneck link. Moreover, in equilibrium, the source rates are matched to the link capacity while the buffer is cleared     

ATM网中流量管理策略的比较研究

论述了有关资源预留对比走入业务，开环控制对比反馈控制，基于速率控制对比基于窗口控制以及基于路由器控制对比基于信源控制等几个关于高速网中拥塞控制和拥塞避免方法的争论双方的论点。     

一种基于Markov及反馈机制的网络拥塞控制方法

在研究现有网络数据传递拥塞控制问题的基础上,提出了一种基于路由器缓冲区队列长度的网络拥塞预测机制,并以该预测机制为基础提出了一种计费策略(PPBC,Per-Packet Billing Control)来控制网络拥塞.在拥塞控制模式中,将路由器缓冲区队列长度作为拥塞指标,通过马尔可夫链建模预测网络拥塞度.在服务器终端和数据发送方之间建立反馈机制,利用网络拥塞度,通过数据包计费方式控制网络拥塞.最后采用NS2仿真软件对PPBC性能进行仿真测评,仿真结果表明:PPBC不仅能够有效缓解网络拥塞,还能够使网络保持较低的丢包率.