A scalable low-overhead rate control algorithm for multirate multicast sessions

In multirate multicasting, different users (receivers) within the same multicast group can receive service at different rates, depending on the user requirements and the network congestion level. Compared with unirate multicasting, this provides more flexibility to the user and allows more efficient usage of the network resources. We address the rate control problem for multirate multicast sessions, with the objective of maximizing the total receiver utility. This aggregate utility maximization problem not only takes into account the heterogeneity in user requirements, but also provides a unified framework for diverse fairness objectives. We propose an algorithm for this problem and show, through analysis and simulation, that it converges to the optimal rates. In spite of the nonseparability of the problem, the solution that we develop is completely decentralized, scalable and does not require the network to know the receiver utilities. The algorithm requires very simple computations both for the user and the network, and also has a very low overhead of network congestion feedback.     

Congestion control for fair resource allocation in networks with multicast flows

We consider the problem of congestion control in networks which support both multirate multicast sessions and unicast sessions. We present a decentralized algorithm which enables the different rate-adaptive receivers in different multicast sessions to adjust their rates to satisfy some fairness criterion. A one-bit ECN marking strategy to be used at the nodes is also proposed. The congestion-control mechanism does not require any per-flow state information for unicast flows at the nodes. At junctions nodes of each multicast tree, some state information about the rates along the branches at the node may be required. The congestion-control mechanism takes into account the diverse user requirements when different receivers within a multicast session have different utility functions, but does not require the network to have any knowledge about the receiver utility functions.     

A Unified Approach to Heterogeneous Video-on-Demand Broadcasting

Video-on-demand (VoD) is an essential technology for many multimedia applications. However, it remains challenging to efficiently deliver on-demand streams to a large number of receivers in a heterogeneous network environment. Although a number of VoD broadcasting protocols have been proposed for heterogeneous receivers, these protocols considers only single-bit-rate VoD services, which cannot satisfy the diverse quality-of-service requirements of heterogeneous receivers. In this paper, we propose a unified approach to heterogeneous VoD broadcasting, called unified heterogeneous broadcasting (UHB). As a unified approach, UHB is design to integrate different VoD broadcasting protocols and different scalable video coding schemes. The main merits of UHB are threefold. First, unlike conventional VoD broadcasting protocols, UHB supports multiple-bit-rate VoD services. Second, UHB is capable of adapting to network bandwidth fluctuations. Third, UHB enables each heterogeneous receiver to obtain the best visual quality by selecting the optimum system joining time. Our performance evaluation results indicate that UHB is superior to conventional heterogeneous VoD broadcasting protocols under various network conditions.     

Intra-Session Fairness in Multicast Communications

Multicast communication achieves scalability by sending data to multiple receivers at the same time. Receivers in a multicast session usually share the fate with each other, even though their processing speed and the capacity of the path they use can be quite different. A conventional multicast session usually consists of a single multicast group and the problem is how to set the group rate so that it is fair to both fast and slow receivers, to some extent. In a replicated multicast service, receivers are divided into groups based on their capacities and a multicast session can consist of multiple multicast groups. The question is how to divide receivers into groups exactly and set appropriate group rates so that it is fair to all the receivers. Most of current work focuses on optimizing the social welfare represented as a sum of some performance measures of receivers [Kar et al., 2002; Stoenescu et al., 2003]. In this paper, we define a new concept called intra-session fairness and give an optimal solution that can achieve fairness among receivers in the same session. The goal is to maximize the minimum fairness value of the receivers. The novelty of the framework is that it is independent of the specific definition of the fairness function on individual receivers. We illustrate a layering method to implement the max-min intra-session fair allocation and demonstrate the significant difference in fairness achieved by the maximal social welfare algorithm and the max-min intra-session fairness algorithm.     

Delay sensitive scheduling schemes for heterogeneous QoS over wireless networks

Future wireless networks will support the growing demands of heterogeneous and delay sensitive applications. In this paper, a users' satisfaction factor (USF) is defined to quantify quality of service (QoS) for different types of services such as voice, data, and multimedia, as well as for different delay constraints. This USF not only predicts the final delivered QoS during transmission, but also take advantages of the fact that different packets can be decoded at different time in the receivers. Based on this USF, four types of scheduling schemes considering tradeoffs between system performance and individual fairness are proposed. These schemes explore the time, channel, and multi-user diversity to guarantee quality of service and enhance the network performance. From the simulation results, the proposed scheduling schemes achieve different tradeoffs between individual fairness and high system performance for the heterogeneous and delay sensitive applications, compared with the weighted round-robin and the modified proportional fairness scheduling schemes     

FLID-DL: congestion control for layered multicast

We describe fair layered increase/decrease with dynamic layering (FLID-DL): a new multirate congestion control algorithm for layered multicast sessions. FLID-DL generalizes the receiver-driven layered congestion control protocol (RLC) introduced by Vicisano et al. (Proc. IEEE INFOCOM, San Francisco, CA, , p.996-1003, Mar. 1998)ameliorating the problems associated with large Internet group management protocol (IGMP) leave latencies and abrupt rate increases. Like RLC, FLID-DL, is a scalable, receiver-driven congestion control mechanism in which receivers add layers at sender-initiated synchronization points and leave layers when they experience congestion. FLID-DL congestion control coexists with transmission control protocol (TCP) flows as well as other FLID-DL sessions and supports general rates on the different multicast layers. We demonstrate via simulations that our congestion control scheme exhibits better fairness properties and provides better throughput than previous methods. A key contribution that enables FLID-DL and may be useful elsewhere is dynamic layering (DL), which mitigates the negative impact of long IGMP leave latencies and eliminates the need for probe intervals present in RLC. We use DL to respond to congestion much faster than IGMP leave operations, which have proven to be a bottleneck in practice for prior work.     

Achieving inter-session fairness for layered video multicast

The Internet is increasingly used to deliver multimedia services. Since there are heterogeneous receivers and changing network conditions, it has been proposed to use adaptive rate control techniques such as layered video multicast to adjust the video traffic according to the available Internet resources. A problem of layered video multicast is that it is unable to provide fair bandwidth sharing between competing video sessions. We propose two schemes, layered video multicast with congestion sensitivity and adaptive join-timer (LVMCA) and layered video multicast with priority dropping (LVMPD), to achieve inter-session fairness for layered video multicast. Receiver-driven layered multicast (RLM), layer-based congestion sensitivity, LVMCA, and LVMPD are simulated and compared. Results show both proposed schemes, especially LVMPD, are fairer and have shorter convergence time than the other two schemes.     

Scalable monitoring support for resource management and service assurance

Continuous monitoring of network status and its resources are necessary to ensure proper network operation. Deployment of QoS-based value-added services in IP networks necessitates the employment of resource management techniques and specifically the use of traffic engineering. The latter typically relies on monitoring data for both offline proactive and dynamic reactive solutions. The variety of data to be collected and analyzed using different measurement methods and tools, and the extent of monitoring information to use demand a proper QoS monitoring infrastructure. A monitoring system should be scalable in terms of network size, speed, and number of customers subscribed to value-added services. This article investigates the requirements of scalable monitoring system architectures, proposes principles for designing such systems, and validates them through the design and implementation of a scalable monitoring system for QoS delivery in IP differentiated services networks. Experimental assessment results prove the accuracy and scalability of the proposed monitoring system.     

区分服务中一种基于聚集流内部公平性的标记算法

在现今异构刚络Internet,区分服务体系结构通过对不同应用和不同要求的数据流部署不同等级的服务,以提供较大粒度的服务质量(Quality of Service,QoS)保证.不同流之间的公平性问题是区分服务中的研究热点,而标记策略是提高公平性的有效方法.目前许多标记算法只考虑了聚集流之间的公平性,而忽略了聚集流内部流之间的公平性.本文针对同一个聚集流中可能包含不同类型的单个流情况(比如存在自适应TCP流和非适应UDP流、不同速率的多媒体UDP流、采用不同TCP协议的数据流、不同分组大小的数据流),提出了一种基于聚集流内部公平性的标记算法(Fair Aggregate Traffic Marker,FATM).论文构造了不同情况下的模拟场景,并给出了实验的相关参数设置.大量模拟实验结果表明:标记算法FATM在保持聚集流之间的公平性和网络吞吐量的基础上,提高聚集流内部单个流之间的公平性.     

The impact of multicast layering on network fairness

Many definitions of fairness for multicast networks assume that sessions are single rate, requiring that each multicast session transmits data to all of its receivers at the same rate. These definitions do not account for multirate approaches, such as layering, that permit receiving rates within a session to be chosen independently. We identify four desirable fairness properties for multicast networks, derived from properties that hold within the max-min fair allocations of unicast networks. We extend the definition of multicast max-min fairness to networks that contain multirate sessions, and show that all four fairness properties hold in a multirate max-min fair allocation, but need not hold in a single-rate max-min fair allocation. We then show that multirate max-min fair rate allocations can be achieved via intra-session coordinated joins and leaves of multicast groups. However, in the absence of coordination, the resulting max-min fair rate allocation uses link bandwidth inefficiently, and does not exhibit some of the desirable fairness properties. We evaluate this inefficiency for several layered multirate congestion control schemes, and find that, in a protocol where the sender coordinates joins, this inefficiency has minimal impact on desirable fairness properties. Our results indicate that sender-coordinated layered protocols show promise for achieving desirable fairness properties for allocations in large-scale multicast networks     

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     

Bandwidth-allocation policies for unicast and multicast flows

Using multicast delivery to multiple receivers reduces the aggregate bandwidth required from the network compared to using unicast delivery to each receiver. However, multicast is not yet widely deployed in the Internet. One reason is the lack of incentive to use multicast delivery. To encourage the use of multicast delivery, we define a new bandwidth-allocation policy, called LogRD, taking into account the number of downstream receivers. This policy gives more bandwidth to a multicast flow as compared to a unicast flow that shares the same bottleneck, without starving the unicast flows, however. The LogRD policy also provides an answer to the question on how to treat a multicast flow compared to a unicast flow sharing the same bottleneck. We investigate three bandwidth-allocation policies for multicast flows and evaluate their impact on both receiver satisfaction and fairness using a simple analytical study and a comprehensive set of simulations. The policy that allocates the available bandwidth as a logarithmic function of the number of receivers downstream of the bottleneck achieves the best tradeoff between receiver satisfaction and fairness     

A simple, scalable, and stable explicit rate allocation algorithmfor max-min flow control with minimum rate guarantee

We present a novel control-theoretic explicit rate (ER) allocation algorithm for the max-min flow control of elastic traffic services with minimum rate guarantee in the setting of the ATM available bit rate (ABR) service. The proposed ER algorithm is simple in that the number of operations required to compute it at a switch is minimized, scalable in that per-virtual-circuit (VC) operations including per-VC queueing, per-VC accounting, and per-VC state management are virtually removed, and stable in that by employing it, the user transmission rates and the network queues are asymptotically stabilized at a unique equilibrium point at which max-min fairness with minimum rate guarantee and target queue lengths are achieved, respectively. To improve the speed of convergence, we normalize the controller gains of the algorithm by the estimate of the number of locally bottlenecked VCs. The estimation scheme is also computationally simple and scalable since it does not require per-VC accounting either. We analyze the theoretical performance of the proposed algorithm and verify its agreement with the practical performance through simulations in the case of multiple bottleneck nodes. We believe that the proposed algorithm will serve as an encouraging solution to the max-min flow control of elastic traffic services, the deployment of which has been debated long due to their lack of theoretical foundation and implementation complexity     

基于网络控制的分层多点播送速率控制机制研究

本文在视频分层编码及分层传输协议的基础上，将基于网络控制与接收端控制机制相结合，提出一种新的分层多点播送速率控制机制。文中给出了该机制的拥塞检测、流行度权衡和数据层增加和丢弃算法。实验结果表明，该速率控制机制通史对拥塞做也快速响应，并在较好利用宽带的基础上保证高流行度的会话流具有较好的服务质量。     

Linear transceiver design for a MIMO interfering broadcast channel achieving max–min fairness

We consider the problem of linear transceiver design to achieve max–min fairness in a downlink MIMO multicell network. This problem can be formulated as maximizing the minimum rate among all the users in an interfering broadcast channel (IBC). In this paper we show that when the number of antennas is at least two at each of the transmitters and the receivers, the min rate maximization problem is NP-hard in the number of users. Moreover, we develop a low-complexity algorithm for this problem by iteratively solving a sequence of convex subproblems. We theoretically establish the global convergence of the proposed algorithm to the set of stationary points, which may be suboptimal due to the non-convexity of the original minimum rate maximization problem. Numerical simulations show that this algorithm is efficient in achieving fairness among all the users.     

A cognitive accountability mechanism for penalizing misbehaving ECN‐based TCP stacks

The introduction of high‐bandwidth demanding services such as multimedia services has resulted in important changes on how services in the Internet are accessed and what quality‐of‐experience requirements (i.e. limited amount of packet loss, fairness between connections) are expected to ensure a smooth service delivery. In the current congestion control mechanisms, misbehaving Transmission Control Protocol (TCP) stacks can easily achieve an unfair advantage over the other connections by not responding to Explicit Congestion Notification (ECN) warnings, sent by the active queue management (AQM) system when congestion in the network is imminent. In this article, we present an accountability mechanism that holds connections accountable for their actions through the detection and penalization of misbehaving TCP stacks with the goal of restoring the fairness in the network. The mechanism is specifically targeted at deployment in multimedia access networks as these environments are most prone to fairness issues due to misbehaving TCP stacks (i.e. long‐lived connections and a moderate connection pool size). We argue that a cognitive approach is best suited to cope with the dynamicity of the environment and therefore present a cognitive detection algorithm that combines machine learning algorithms to classify connections into well‐behaving and misbehaving profiles. This is in turn used by a differentiated AQM mechanism that provides a different treatment for the well‐behaving and misbehaving profiles. The performance of the cognitive accountability mechanism has been characterized both in terms of the accuracy of the cognitive detection algorithm and the overall impact of the mechanism on network fairness. Copyright © 2012 John Wiley & Sons, Ltd.     

Explicit rate flow control for ABR services in ATM networks

We propose a novel explicit rate flow control algorithm intended for available-bit-rate (ABR) service on an ATM network subject to loss and fairness constraints. The goal is to guarantee low cell loss in order to avoid throughput collapse due to retransmission by higher level protocols. The mechanism draws on measuring the current queue length and bandwidth availability, as well as tracking the current number of active sessions contending for capacity, to adjust an explicit bound on the source transmission rates. We identify the factors that affect queue overflows and propose simple design rules aimed at achieving transmission with controlled loss in a dynamic environment. We also discuss how conservative design rules might be relaxed by accounting for statistical multiplexing in bandwidth sharing among bursty ABR sources and variable-bit-rate (VBR) sources     

Adaptive algorithms for feedback-based flow control in high-speed,wide-area ATM networks

This paper gives a class of flow control algorithms for the adaptive allocation of bandwidths to virtual connections (VC) in high-speed, wide-area ATM networks. The feedback rate to the source from the network is parsimonious, with each feedback bit indicating whether the buffer at a distant switch is above or below a threshold. The service discipline at the switch is first-come-first-served. The important goal of adaptability aims to make all of the network bandwidth available to the active VCs, even though the number of such VCs is variable over a given range. Each VC has two parameters, one giving its minimum guaranteed bandwidth and the other is the weight for determining its share of the uncommitted bandwidth. Judicious selection of these parameters defines distinctive services, such as best effort and best effort with minimum bandwidth. We derive design rules for selecting the parameters of the algorithms such that the appropriate guarantees and fairness properties are exhibited in the dynamical behavior. The systematic use of “damping” in right proportion with “gain” is shown to be a powerful device for stabilizing behavior and achieving fairness. Our analyses are based on a simple analytic fluid model composed of a system of first-order delay-differential equations, which reflect the propagation delay across the network. Extensive simulations examine the following: (1) fairness, especially to start-up VCs; (2) oscillations; (3) transient behavior, such as the rate of equalization from different initial conditions; (4) disparate bandwidth allocations; (5) multiple paths with diverse propagation delays; (6) adaptability and robustness with respect to parameters; and (7) interoperability of different algorithms