A novel prediction technique to improve quality of service (QoS) for heterogeneous data traffic

The present and future high-speed networks are expected to support wide variety real-time applications. However, the current Internet architecture offers mainly best-effort service. It means that the network will do its best to deliver the data at the destination without any guarantee. But the future integrated services networks will require guarantee for transferring heterogeneous data. There are many parameters involved in improving the Quality of Service (QoS). QoS is a set of service requirements to be met by the network while transporting a flow. In this paper, we consider four primary parameters are such as reliability, delay, jitter, bandwidth which together determine the QoS. The requirements of the above parameters will vary from one application to another application. Applications like file transfer, remote login, etc., will require high reliability. But, applications like audio, video, etc., will require low reliability, because they can tolerate errors. The objectives of this paper are to propose a novel technique to predict reason(s) for deterioration in the QoS and to identify the algorithm(s)/mechanism(s) responsible for the deterioration. We are sure that this paper will give better results to improve the QoS and to improve the performance of the network.     

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.     

An efficient bandwidth allocation algorithm for real-time VBR stream transmission under IEEE 802.16 wireless networks

This paper investigates variable rate control strategies for real-time multimedia variable bit rate (VBR) services over IEEE 802.16 broadband wireless networks. A data rate control mechanism is derived for the case where the uplink channel provides real-time services and the traffic rate parameter remains constant. This paper shows that the common queuing scheduling algorithms have some bandwidth allocation fairness problems for the real-time polling service (rtPS) in the MAC layer. In other words, the use of a VBR for the rtPS by a WiMAX system results in additional access latency jitter and bandwidth allocation disorder in the transmitted multimedia streams during the regular time interval polling of subscribe stations (SSs) for the contention bandwidth request period. However, the proposed scheduling algorithm solves these SSs contending with bandwidth resource allocation problems based on an extended rtPS (ertPS) of quality-of-service (QoS) pre-programming for a ranging response non-contention polling period. The adopted bandwidth allocation of max–min fairness queue scheduling uses a time constraint condition to transmit real-time multimedia VBR streaming in an IEEE 802.16 broadband wireless environment. In addition, we use the ns-2 simulation tool to compare the capacity of multimedia VBR stream and show that the proposed ertPS scheduling algorithm outperforms other rtPS scheduling algorithms.     

基于实时传输协议的QOS研究

随着互联网技术的发展与普及,多媒体等实时业务的应用变得越来越广泛,同时实时业务对网络传输的服务质量（QoS）要求也越来越高。在分析了影响网络实时传输QoS因素的基础上,利用丢包率作为判断当前网络拥塞与否的参考量,设计了一种自适应的流量控制机制,该机制使用变常数增长和乘减少的方式对发送速率进行自适应调整。仿真结果表明,该机制有效地提高了实时传输的QoS。     

Scalability and QoS guarantee for streams with (m,k)-firm deadline

Real-time media streams with (m,k)-firm are increasing and attractive as it proposes an alternative quality of service (QoS). In order to meet these ever increasing demands, the problem of scalability should be considered under limited resource. The problem is ignored in the existing solutions all along, because these solutions maintain a separate queue for each stream and per-stream state information. In this paper, we extend the work made in[Ji Ming Chen, Zhi Wang, Ye Qiong Song, and You Xian Sun. A scalable approach for streams with (m,k)-firm deadline. In Proceedings of IEEE Workshop on Quality of Service for Application Servers (2004 October)] and propose loss-rate balance (static) and stream number balance (dynamic) Class Selection Algorithm (CSA) for real-time media streams with (m,k)-firm, which is scalable while offering dynamic performance close to that of existing solutions. The effectiveness of static CSA (S-CSA) that captures the trade-off between scalability and QoS granularity is evaluated through extensive simulation studies. It is also shown that the S-CSA is as effective to guarantee QoS in terms of dynamic failure rate as the dynamic one.     

Impact of video scheduling on bandwidth allocation for multiplexed MPEG streams

We present efficient schemes for scheduling the delivery of variable-bit-rate MPEG-compressed video with stringent quality-of-service (QoS) requirements. Video scheduling is being used to improve bandwidth allocation at a video server that uses statistical multiplexing to aggregate video streams prior to transporting them over a network. A video stream is modeled using a traffic envelope that provides a deterministic time-varying bound on the bit rate. Because of the periodicity in which frame types in an MPEG stream are typically generated, a simple traffic envelope can be constructed using only five parameters. Using the traffic-envelope model, we show that video sources can be statistically multiplexed with an effective bandwidth that is often less than the source peak rate. Bandwidth gain is achieved without sacrificing the stringency of the requested QoS. The effective bandwidth depends on the arrangement of the multiplexed streams, which is a measure of the lag between the GOP periods of various streams. For homogeneous streams, we give an optimal scheduling scheme for video sources at a video-on-demand server that results in the minimum effective bandwidth. For heterogeneous sources, a sub-optimal scheduling scheme is given, which achieves acceptable bandwidth gain. Numerical examples based on traces of MPEG-coded movies are used to demonstrate the effectiveness of our schemes.     

基于网格的流媒体服务QoS管理框架及实现

流媒体服务是Internet上一类高带宽需求和高实时性约束的应用，对服务质量（Quality of Service，QoS）有较高的要求。流媒体服务的发展导致传统的QoS管理框架难以适应平台的异构性和复杂性。本文提出了一种基于网格的流媒体服务QoS管理框架，为由异构的系统构成的流媒体服务提供集成的、平台无关的QoS管理机制。在谈框架的基础上，我们设计了一个基于网格的流媒体服务QoS管理系统。     

QoS guarantee for multimedia traffic in smart homes

With the advent of home networking and widespread deployment of broadband connectivity to homes, a wealth of new services with real-time Quality of Service (QoS) requirements have emerged, e.g., Video on Demand (VoD), IP Telephony, which have to co-exist with traditional non-real-time services such as Web browsing and file downloading over the Transmission Control Protocol (TCP). The co-existence of such real-time and non-real-time services demands the residential gateway (RG) to employ bandwidth management algorithms to control the amount of non-real-time TCP traffic on the broadband access link from the Internet Service Provider (ISP) to the RG so that the bandwidth requirements of the real-time traffic are satisfied. In this paper we propose an algorithm to control the aggregate bandwidth of the incoming non-real-time TCP traffic at the RG so that QoS requirements of the real-time traffic can be guaranteed. The idea is to limit the maximum data rates of active TCP connections by dynamically manipulating their flow control window sizes based on the total available bandwidth for the non-real-time traffic. We show by simulation results that our algorithm limits the aggregate bandwidth of the non-real-time TCP traffic thus granting the real-time traffic the required bandwidth.     

基于马尔科夫链的联合呼叫接入控制算法

异构网络的接入策略与网络资源管理效率紧密相关;同时,网络复杂性与网络资源竞争性直接影响到用户服务质量。针对异构网络接入控制存在的切换掉话率和呼叫阻塞率高、资源利用率低等问题,提出了基于马尔科夫链的联合呼叫接入控制算法。接入控制算法为切换呼叫业务、实时业务动态地预留了一定的带宽资源,根据不同业务设置带宽降级因子来决定是否释放带宽;同时,根据用户偏好和不同业务的QoS要求,构建了呼叫接入控制效用函数,利用马尔科夫链进行了建模分析。仿真表明,算法提高了网络资源利用率,降低了系统复杂度,满足了各类业务的QoS要求。     

无线多媒体网络中一种基于测量网络状态的动态呼叫接纳控制算法

提出了一种无线多媒体网络中基于测量网络状态的动态呼叫接纳控制算法.它区分了实时和非实时业务,在网络带宽资源不足时可通过降低非实时业务带宽确保实时业务呼叫连接的可靠性;还可根据当前网络状况调整预留带宽大小,使小区实时业务切换呼叫掉线率低于设定的门限值.大量仿真结果显示该算法具有低实时业务切换呼叫掉线率和与固定预留方案相当的带宽利用率,而只以略高的新呼叫阻塞率为代价,适合各种不同概率发生时实际应用的情况.     

LGRR: A new packet scheduling algorithm for differentiated services packet-switched networks

Since Quality of Service (QoS) support is a mandatory requirement in the next-generation networking, each router in a packet-switched network must provide a better service to higher-priority packets under any situation such as congestion. We propose in this paper the loan-grant based Round Robin (LGRR) packet scheduler for use in each output port of a router in a DiffServ network. LGRR is a frame-based scheduler to pass traffic streams according to their class types and to their immediate upstream source routers. It uses a loan-grant scheme so that a higher priority traffic stream can be processed quickly by requesting a bandwidth loan from the scheduler. To control the amount of transmitted bits from each stream and to prevent malicious abuse, the bandwidth loan must be paid back from the quantum values acquired in future. LGRR gives a fair opportunity to different traffic streams to access to the network bandwidth. It performs better than MDRR+, MDRR++, and OCGRR in handling traffic under both normal and bursty traffic, but it also gives a better loss and delay performance to the higher-priority traffic when traffic load is very high.     

流媒体传输中的QoS研究及其实现

文中提出的对视频会议的服务质量控制策略是为解决以下问题:视频会议对涉及到的实时数据流传输对网络带宽、时延和丢包率有较高要求,但是,目前得到广泛应用的IP网络是一种尽力而为的网络,它并没有对实时多媒体提供任何服务质量保证。该策略从两个方面对服务质量加以控制:在发送端控制发送流量;在数据再现端通过缓冲机制控制媒体同步。     

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.     

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.     

Resource management for bursty streams on multi-tenancy cloud environments

The number of applications that need to process data continuously over long periods of time has increased significantly over recent years. The emerging Internet of Things and Smart Cities scenarios also confirm the requirement for real time, large scale data processing. When data from multiple sources are processed over a shared distributed computing infrastructure, it is necessary to provide some Quality of Service (QoS) guarantees for each data stream, specified in a Service Level Agreement (SLA). SLAs identify the price that a user must pay to achieve the required QoS, and the penalty that the provider will pay the user in case of QoS violation. Assuming maximization of revenue as a Cloud provider’s objective, then it must decide which streams to accept for storage and analysis; and how many resources to allocate for each stream. When the real-time requirements demand a rapid reaction, dynamic resource provisioning policies and mechanisms may not be useful, since the delays and overheads incurred might be too high. Alternatively, idle resources that were initially allocated for other streams could be re-allocated, avoiding subsequent penalties. In this paper, we propose a system architecture for supporting QoS for concurrent data streams to be composed of self-regulating nodes. Each node features an envelope process for regulating and controlling data access and a resource manager to enable resource allocation, and selective SLA violations, while maximizing revenue. Our resource manager, based on a shared token bucket, enables: (i) the re-distribution of unused resources amongst data streams; and (ii) a dynamic re-allocation of resources to streams likely to generate greater profit for the provider. We extend previous work by providing a Petri-net based model of system components, and we evaluate our approach on an OpenNebula-based Cloud infrastructure.     

Network-adaptive autonomic transcoding algorithm for seamless streaming media service of mobile clients

As a result of improvements in wireless communication technologies, a multimedia data streaming service can now be provided for mobile clients. Since mobile devices have low computing power and work on a low network bandwidth, a transcoding technology is needed to adapt the original streaming media for mobile environments. However, wireless networks have variable bandwidths depending on the movement of clients and the communication distance from Access Point (AP). These characteristics make it hard to support stable Quality of Service (QoS) streams for mobile clients. In this paper, a target transcoding bit-rate decision algorithm is proposed to provide stable QoS streams for mobile clients. In our experiments, the proposed algorithm provides seamless streaming media services based on the network adaptive bit rate control and reduces transmission failure.     

A new approach to dynamic bandwidth allocation in Quality of Service networks: Performance and bounds

Efficient dynamic resource provisioning algorithms are necessary to the development and automation of Quality of Service (QoS) networks. The main goal of these algorithms is to offer services that satisfy the QoS requirements of individual users while guaranteeing at the same time an efficient utilization of network resources.In this paper we introduce a new service model that provides per-flow bandwidth guarantees, where users subscribe for a guaranteed rate; moreover, the network periodically individuates unused bandwidth and proposes short-term contracts where extra-bandwidth is allocated and guaranteed exclusively to users who can exploit it to transmit at a rate higher than their subscribed rate.To implement this service model we propose a dynamic provisioning architecture for intra-domain Quality of Service networks. We develop a set of dynamic on-line bandwidth allocation algorithms that take explicitly into account traffic statistics and users’ utility functions to increase users’ benefit and network revenue.Further, we propose a mathematical formulation of the extra-bandwidth allocation problem that maximizes network revenue. The solution of this model allows to obtain an upper bound on the performance achievable by any on-line bandwidth allocation algorithm.We demonstrate through simulation in realistic network scenarios that the proposed dynamic allocation algorithms are superior to static provisioning in providing resource allocation both in terms of total accepted load and network revenue, and they approach, in several network scenarios, the ideal performance provided by the mathematical model.     

Real-time applications in a CRMA network: a performance analysis

This paper analyses bandwidth allocation schemes for managing real-time applications (Variable Bit Rate video and voice) in a CRMA network. A methodology to compute the Quality of Service (QoS) experienced by variable bit rate (VBR) video and voice sources is proposed. As VBR video applications only tolerate extremely low packet loss rates (< 10⁻⁸), we need a computational approach to estimate very low tail probabilities. Studying the QoS with a simulation technique is not feasible, because computational costs make it almost impossible to estimate tail distribution probabilities lower than 10⁻²−10⁻³. Therefore, to achieve this target, we propose a model which represents a CRMA network's worst case behaviour (i.e. a scenario with maximum network congestion), and which can be solved analytically. By solving this model for different bandwidth allocation schemes, we can obtain the corresponding bounds on the QoS experienced by VBR video users. Finally, for those bandwidth allocation schemes which provide an acceptable QoS for VBR video traffic, we also estimate (via a trace-driven simulation) the QoS achieved by voice users.     

Service Adaptability in Multimedia Wireless Networks

Next-generation wireless communication systems aim at supporting wireless multimedia services with different quality-of-service (QoS) and bandwidth requirements. Therefore, effective management of the limited radio resources is important to enhance the network performance. In this paper, we propose a QoS adaptive multimedia service framework for controlling the traffic in multimedia wireless networks (MWN) that enhances the current methods used in cellular environments. The proposed framework is designed to take advantage of the adaptive bandwidth allocation (ABA) algorithm with new calls in order to enhance the system utilization and blocking probability of new calls. The performance of our framework is compared to existing framework in the literature. Simulation results show that our QoS adaptive multimedia service framework outperforms the existing framework in terms of new call blocking probability, handoff call dropping probability, and bandwidth utilization.