Control of perceived quality of service in multimedia retrieval services: prediction-based mechanism vs. compensation buffers

In multimedia systems end-to-end delay jitter has a great impact on the continuity of information playback. Therefore, it is necessary to introduce appropriate mechanisms to compensate for delay variations, so that the intramedia and intermedia temporal relationships can be preserved. In this paper, two methods for compensation of the network delay jitter in a distributed multimedia retrieval service are compared: the first is based on prediction of the network delay jitter suffered by each information unit and retrieval time modification at the source site; the second is based on a compensation buffer at the destination site. Comparison is made by assuming a master/slave relationship between the monomedia streams composing the multimedia data flow.     

Multimedia streaming gateway with jitter detection

This paper investigates a novel active buffer management scheme, "Jitter Detection" (JD) for gateway-based congestion control to stream multimedia traffics in packet-switched networks. The quality of multimedia presentation can be greatly degraded due to network delay variation or jitter when transported over a packet-switched network. Jitter degrades the timing relationship among packets in a single media stream and between packets from different media streams and, hence, creates multimedia synchronization problems. Moreover, too much jitter will also degrade the performance of the streaming buffer in the client. Packets received by the client will be rendered useless if they have accumulated enough jitter. The proposed active buffer management scheme will improve the quality of service in multimedia networking by detecting and discarding packets that accumulated enough jitter, such as to maintain a high bandwidth for packets within the multimedia stream's jitter tolerance. Simulation results have shown that the proposed scheme can effectively lower the average received packet jitter and increase the goodput of the received packets when compared to random early detection (RED) and DropTail used in gateway-based congestion control. Furthermore, simulation results have also revealed that the proposed scheme can maintain the same TCP friendliness when compared to that of RED and DropTail used for multimedia streams.     

Impact of queuing discipline on packet delivery latency in ad hoc networks

Delivering live multimedia streaming over ad hoc networks can improve coordination in battlefields, assist in disaster recovery operations, and help prevent vehicular traffic accidents. However, ad hoc networks often experience congestion faster than wired networks, leading to high end-to-end delays and jitter even for moderate traffic. This paper describes a partial remedy that applies to delay sensitive but loss tolerant applications such as live streaming. We find that under relatively high UDP traffic load, the Last-In–First-Out (LIFO) with Frontdrop queuing discipline achieves less than half the delay of the commonly used First-In–First-Out (FIFO) with Taildrop, while maintaining similar jitter. In low traffic situations, FIFO and LIFO have similar delays, but FIFO with Frontdrop has the lowest jitter. The results can be applied to an adaptive queuing mechanism that changes the queuing discipline at nodes function of the locally observed traffic load. The advantage of such an approach is that it does not require new protocols and does not incur any network overhead.     

A routing framework for load balancing of bandwidth sensitive traffic in differentiated service networks

The Differentiated Service (DiffServ) network model has been defined as a scalable framework for providing Quality of Service to applications. In this model, traffic is classified into several service classes with different priorities inside queues of IP routers. The premium service class has the highest priority. Due to the high priority of premium traffic, the global network behaviour against this service class, including routing and scheduling of premium packets, may impose significant influences on traffic of other classes. These negative influences, which could degrade the performance of low-priority classes with respect to some important metrics such as the packet loss probability and the packet delay, are often called the inter-class effects. To reduce the inter-class effects, the premium-class routing algorithm must be carefully selected such that (1) it works correctly (i.e., without loop) under the hop-by-hop routing paradigm; and (2) the congestion resulted from the traffic of premium class over the network becomes minimum. In this paper, we first introduce a novel routing framework, named compatible routing, that guarantees loop-freedom in the context of hop-by-hop routing model. Then, upon this framework, we propose two multipath architectures for load balancing of high-priority traffic on DiffServ networks. Our extensive simulations clearly demonstrate that the proposed methods distribute the premium bandwidth requirements more efficiently over the whole network and perform better than the existing algorithms, especially in the case of complex and highly loaded networks.     

A new design for end-to-end proportional loss differentiation in IP networks

This paper describes the algorithms and the architecture of a network able to provide end-to-end proportional packet loss probabilities at the flow level. We show that the combination of a simple classification technique at the sources, and a network core having two internal service classes, is sufficient to achieve proportional service without the need to deploy coordinated, complex per-hop scheduling schemes or signaling protocols, which is the conventional approach. The proposed architecture is complementary to any differentiation algorithm used by the routers. Our results show that any network endowed with some internal service classes with respect to packet loss probabilities can be exploited to build a set of external service classes with end-to-end and per-flow guarantees.     

高速路由器中一种有效的组播交换排队机制

许多Internet新业务产生的网络流量都属于组播类型．高速路由器中组播交换队列设计是解决组播数据在复杂网络中有效传输的关键问题之一．理想情况下,为保证服务的公平性及有效性,不应该在排队和调度时对组播和单播人为地进行区分．在二者共存的情况下,如何进行公平地排队、交换是亟待解决的问题．基于共享存储交换结构,提出了一种有效的组播交换队列设计方案,同时提供了相应的与组播比例和平均扇出相关的缓存管理机制和队列调度算法．仿真结果表明,与在Cisco高端路由器中广泛应用的ESLIP设计方案相比,在输入负载大于80％的重负载环境中,该系统能够获得更低的平均时延和更小的系统丢包率．     

Delay-jitter control in multimedia applications

The growing needs of multimedia communications are leading to new developments in providing real-time communication with guarantees. Several extensions have been proposed for different layers of the Open Systems Interconnection Reference Model to accomodate these needs. In this paper, we study methods for guaranteeing delay jitter bounds for high-speed networks in the network and application layers of this model. The method proposed for the network layer provides distributed jitter control. The method proposed for the application layer allows the destination application to control delay jitter. We use a simulation to compare the effects on delay jitter in each method for various scenarios, such as constant bit rate, cross traffic, and bursty data. In addition, the buffer space requirements for accommodating real-time channels are monitored at each node in the network.     

保障连续媒体流用户层QoS 的缓存控制

研究了缓存控制对媒体流用户层QoS 的影响.多媒体系统信宿端通常采用播放缓存来补偿时延抖动,提高媒体流播放的连续性.缓存控制虽然能够降低时延抖动的影响,却增加了端到端时延.时延或时延抖动是用户可感知的QoS 参数,缓存控制对用户层QoS 的影响究竟如何呢?利用已有的应用层向用户层QoS 映射的研究结果,分析缓存控制参数与端到端QoS 参数、应用层QoS 参数的关系,获得了缓存控制参数与用户层QoS 参数的关系.从理论上深入挖掘缓存控制对用户层QoS 参数的作用,给出了提供确定时延和时延抖动保障的缓存容量值,论证了在网络环境一定时存在提供最佳用户层QoS 的缓存容量值.实验结果验证了分析.     

基于IP网络规划与设计的抖动模型的研究

提出了一个典型的模型,该模型考虑到包延时的相关性和串联队列的相关性,这对端到端的抖动有重要影响。针对一个单队列的Poisson流量分布的抖动,给出了一个非常易于计算的公式,然后推广到基于串联队列的标记流的端到端抖动。通过模拟实验可以发现,模型的分析值和模拟值基本吻合,在大流量背景下更为精确,更重要的是对于抖动而言该值是可信的,这样就可以用于网络设计过程中。     

基于模糊控制的ATM网络VBR视频传输平滑策略

VBR视频传输的突发性是影响ATM网络服务质量的关键因素,文中通过模糊控制方法对视频传输流量的阈值进行监控,实现了接入的平滑并可动态调整传输速度。文中以传输流量及这发级作为评价指标进行分析,结果表明VBR视频传输自适应平滑策略的实效性。     

On Generalized Processor Sharing With Regulated Multimedia Traffic Flows

Multimedia traffic is becoming an increasing portion of today's Internet traffic due to the flourishing of multimedia applications such as music/video streaming, video teleconferencing, IP telephony, and distance learning. In this paper, we study the problem of supporting multimedia traffic using a generalized processor sharing (GPS) server. By examining the sample path behavior and exploring the inherent feasible ordering of the classes, we derive tight performance bounds on backlog and delay for regulated multimedia traffic classes in a GPS system. Our approach is quite general since we do not assume any arriving traffic model or any specific traffic regulator, other than that each traffic flow is deterministically regulated. Such deterministic regulators, as well as approximations of the GPS server, are widely implemented in commercial routers. In addition, our analysis is very accurate and achieves a high utilization of the server capacity, since we exploit the independence among the traffic flows for higher statistical multiplexing gains. Numerical examples and simulation results are presented to demonstrate the accuracy and merits of our approach, which is practical and well suited for supporting multimedia applications in the Internet     

Energy efficient and QoS based routing protocol for wireless sensor networks

The increasing demand for real-time applications in Wireless Sensor Networks (WSNs) has made the Quality of Service (QoS) based communication protocols an interesting and hot research topic. Satisfying Quality of Service (QoS) requirements (e.g. bandwidth and delay constraints) for the different QoS based applications of WSNs raises significant challenges. More precisely, the networking protocols need to cope up with energy constraints, while providing precise QoS guarantee. Therefore, enabling QoS applications in sensor networks requires energy and QoS awareness in different layers of the protocol stack. In many of these applications (such as multimedia applications, or real-time and mission critical applications), the network traffic is mixed of delay sensitive and delay tolerant traffic. Hence, QoS routing becomes an important issue. In this paper, we propose an Energy Efficient and QoS aware multipath routing protocol (abbreviated shortly as EQSR) that maximizes the network lifetime through balancing energy consumption across multiple nodes, uses the concept of service differentiation to allow delay sensitive traffic to reach the sink node within an acceptable delay, reduces the end to end delay through spreading out the traffic across multiple paths, and increases the throughput through introducing data redundancy. EQSR uses the residual energy, node available buffer size, and Signal-to-Noise Ratio (SNR) to predict the best next hop through the paths construction phase. Based on the concept of service differentiation, EQSR protocol employs a queuing model to handle both real-time and non-real-time traffic.     

优先级队列的缓存管理机制的性能分析

为了解决好计算机网络中的多媒体通信以及对服务质量的要求,提出了一种基于优先级队列的缓存管理机制。详细研究了该机制的实现原理和方法,并给出了相关的性能分析,并建立端对端多媒体通信系统环境,进行了实验,证明了该缓存管理机制能很好地解决多媒体通信的拥塞控制,满足多媒体通信服务质量的要求。     

自相似背景流量模型下的CBR流延迟抖动分析

实时业务是网络中快速增长的业务类型,但网络中需要传送多种业务的混合流量。实时 业务的性能取决于分组延迟抖动。过大的分组延迟抖动将导致语音的中断,画面的停顿和跳跃。延 迟抖动主要是背景流量在边缘路由器的干扰引起的。以往的延迟抖动分析都是假设背景流量为泊松 过程,研究表明这种假设已经不符合当前网络流量的特性。本文将对自相似背景流量下的CBR流的 延迟抖动进行分析,给出其分布函数,并以仿真结果验证其与泊松流量对CBR流的不同影响。     

分布式星群网络中基于蚁群算法的通信量分类路由

给出了一种分布式星群网络(Distributed Satellite Cluster Network,DSCN)体系架构,阐明了DSCN拓扑变化的特点。在分析网络状态获取方式和路由计算方法的基础上,提出了一种适用于DSCN的基于蚁群算法的通信量分类路由(Ant Colony Optimization Based Traffic Classified Routing,ATCR)算法。ATCR算法将通信量分为时延敏感型通信量A、带宽敏感型通信量B以及提供尽力而为服务的通信量C,并对蚁群算法(Ant Colony Optimization,ACO)收敛慢的缺点进行了改进。仿真实验表明,ATCR算法提高了收敛速度,可以有效平衡网络流量。通信量A和C的端到端时延要小于未采用通信量分类的改进ACO算法。由于减少了重负载链路的数量及拥塞引起的丢包,ATCR算法在分组递交率上的表现优于改进的ACO算法。     

Low-cost,delay-bounded point-to-multipoint communication to support multicasting over WDM networks

The confluence of technical advances and multimedia service needs is intensifying the need for high throughput and low latency. Future communication networks will face an increase in traffic driven by multimedia requirements with stringent delay and jitter requirements. Wavelength division multiplexing (WDM) optical networks have the potential for meeting these goals by offering unprecedented high bandwidth and low latency. One very important aspect of the emerging Internet services is the need to support multicasting. This is crucial if WDM networks were to play an efficient role in the next generation Internet.Multicasting in WDM networks supporting multimedia applications can be viewed as the process of taking a group communication request and selecting a multicast tree that satisfies the quality of service requirements, in terms of bandwidth and end-to-end delay, of the underlying application. In this paper, we present a new class of low-cost, bounded-delay multicast heuristics for WDM networks. The heuristics use various techniques to establish a tree of semi-lightpaths between a source and a group of destination nodes. The unique feature of these heuristics is that they decouple the cost of establishing the multicast tree from the delay incurred by data transmission due to light-wave conversion and processing at intermediate nodes along the transmission path. A simulation study shows the performance of the proposed heuristics.     

RCSP and stop-and-go: a comparison of two non-work-conserving disciplines for supporting multimedia communication

To support emerging real-time applications, high-speed integrated services networks must provide end-to-end performance guarantees on a per-connection basis in a networking environment. Resource management algorithms must accommodate traffic that may get burstier as it traverses the network due to complex interactions among packet streams at each switch. To address this problem, several non-work-conserving packet-service disciplines have been proposed. Non-work-conserving servers may be idle and hold packets under certain conditions, to reconstruct, fully or partially, the traffic pattern of the original source inside the network and prevent the traffic from becoming burstier. We compare two non-work-conserving service disciplines. Stop-and-go uses a multilevel framing strategy to allocate resources in a single switch and to ensure traffic smoothness throughout the network. Rate controlled static priority (RCSP) decouples the server functions with two components: (1) a regulator to control traffic distortion introduced by multiplexing effects and load fluctuations in previous servers, and 2) a static priority scheduler to multiplex the regulated traffic. We compare the two service disciplines in terms of traffic specification, scheduling mechanism, buffer space requirement, end-to-end delay characteristics, connection admission-control algorithms, and achievable network utilization. The comparison is first done analytically, and then empirically by using two 10-min traces of MPEG compressed video.     

Investigation of the M2/G2/1/∞,N queue with restricted admission of priority customers and its application to HSDPA mobile systems

This paper investigates a queuing system for QoS optimization of multimedia traffic consisting of aggregated streams with diverse QoS requirements transmitted to a mobile terminal over a common downlink shared channel. The queuing system, proposed for buffer management of aggregated single-user traffic in the base station of High-Speed Downlink Packet Access (HSDPA), allows for optimum loss/delay/jitter performance for end-user multimedia traffic with delay-tolerant non-real-time streams and partially loss tolerant real-time streams. In the queuing system, the real-time stream has non-preemptive priority in service but the number of the packets in the system is restricted by a constant. The non-real-time stream has no service priority but is allowed unlimited access to the system. Both types of packets arrive in the stationary Poisson flow. Service times follow general distribution depending on the packet type. Stability condition for the model is derived. Queue length distribution for both types of customers is calculated at arbitrary epochs and service completion epochs. Loss probability for priority packets is computed. Waiting time distribution in terms of Laplace–Stieltjes transform is obtained for both types of packets. Mean waiting time and jitter are computed. Numerical examples presented demonstrate the effectiveness of the queuing system for QoS optimization of buffered end-user multimedia traffic with aggregated real-time and non-real-time streams.     

Providing traffic smoothness and service guarantees within ATM networks through a rate-controlled service discipline

With the introduction of diverse rate requirements under a variety of statistical multiplexing schemes, traffic burstiness behavior of a source stream and its quality-of-service (QoS) performances within the ATM networks become difficult to model and analyze. In this paper, we address this issue and propose a rate-controlled service discipline that provides control of the traffic burstiness while maintaining QoS guarantees for traffic flows with various rate requirements. According to our analysis, traffic streams from different connections can be well regulated at the output of each network node based on their rate requirements. Traffic envelope and the associated burstiness behavior inside the network can thus be effectively characterized. In addition, by assuming a leaky-bucket constrained input source, we prove that the proposed scheme can provide end-to-end delay and jitter bounds for each connection passing through a multi-hop network. Further, due to the low traffic burstiness, only a small buffer space is required at the internal switches for guaranteeing QoS requirements.     

A fuzzy queue-aware routing approach for wireless mesh networks

Recent advances in Wireless Mesh Networks (WMNs) have overcome the drawbacks of traditional wired and ad-hoc networks and now they are seen as a means of allowing last mile communications with quality level assurance in Future Multimedia Systems. However, new routing schemes are needed to provide end-to-end Quality of Service (QoS) and Quality of Experience (QoE) support for delay/loss/jitter-sensitive multimedia applications. The well-known OLSR (Optimized Link State Routing) protocol with ETX (Expected Transmission Count) metric brings many benefits to the path selection process, but has a drawback with regard to queue availability management, which reduces the system performance. This problem is caused when OLSR-EXT control messages are exchanged and the queues of mesh routers along the end-to-end communication path are overloaded. As a result, multimedia-related packets will suffer from loss/delay/jitter and the overall system performance will decrease. This paper proposes the Optimized Link State Routing-Fuzzy ETX Queue (OLSR-FEQ) protocol to overcome the limitations of OLSR-ETX regarding queue availability, QoS and QoE assurance. OLSR-FEQ optimizes network and user-based parameters by coordinating queue availability, QoS and fuzzy issues in the routing decision process as a way of allocating the best paths for multimedia applications. Performance evaluations were carried out with the Network Simulator (NS-2.34) to show the benefits of the proposed solution when compared with existing routing schemes, namely OLSR-ETX, OLSR-FLC, OLSR-MD and HWMP (IEEE 802.11s standard), regarding QoS (unsuccessful packet delivery and throughput) and QoE (PSNR, SSIM, VQM and MOS) parameters.