A Study on Differentiated Service Queuing Scheme with an Overflow Buffer Allocation Within a UMTS Core Network

According to the QoS features of the four types of UMTS traffic, this study proposes a priority-based queuing scheme to support differentiated services among all UMTS traffic; it bases on packet transmission priorities of four types of UMTS traffic to handle packets forwarding in a gateway within a UMTS core network. In the proposed queuing scheme, a static dedicated logical queuing buffer allocation and a dynamic overflow buffer allocation are used to support packet enqueuing/dequeuing jobs. In this study, the ns2 network simulator is used as a simulation platform and different categories of simulation scenarios are executed. Moreover, the IETF DiffServ scheme is used as a comparison to understand the packet forwarding performance of the proposed scheme. By analyzing simulation data, two important points can be drawn. The proposed queuing scheme supports a differentiated packet forwarding behavior among four types of UMTS traffic. And, the differentiated packet forwarding behaviors with the proposed queuing scheme are similar to the packet forwarding behavior with the IETF DiffServ scheme.     

A Study on Adaptive Time Token Priority-Based Queuing Scheme

Several famous priority-based queuing schemes operated in a gateway to support differentiated services among internet traffic. Examining packet forwarding operations in these queueing schemes, they only support a priority-based service either in a packet enqueuing process or in a packet dequeuing process. If a queuing scheme can support priority-based services in both packet enqueuing/dequeuing processes; it would enhance differentiated service performance for internet traffic. This study proposes a priority-based queuing scheme with an adaptive time token allotment measure to support a differentiated packet forwarding process for different types of IP traffic both in packet enqueuing/dequeuing processes. Depending on packet sizes and packet forwarding priorities of IP traffic, the proposed queuing scheme assigns fix and adaptive time token thresholds dynamically to logical queuing buffers separately. With assigned time tokens, logical queuing buffers allow arrival IP packets to be enqueued in a differentiated way. Moreover, the proposed queuing scheme uses a transferred WRR dequeuing measure to enhance a differentiated packet forwarding process. The simulation results show that the proposed queuing scheme supports a differentiated packet forwarding process for different types of IP traffic. The differentiated packet forwarding performance supported by the proposed scheme is close to the IETF DiffServ scheme; this result shows that the proposed scheme can support differentiated packet forwarding performance for different types of IP traffic with a lower operation cost.     

Improving mobile network performance with two queueing buffer allocations of priority-based queueing scheme

As current mobile core network systems are expected to evolve into all-IP networks, packet switching will be a prerequisite for all mobile applications. Next-generation mobile networks, as envisioned by ITU-T, are packet-based networks capable of providing consistent and ubiquitous service to end users, independent of the network, access technology, and device used. This study discusses the differentiated packet forwarding performance of four major types of mobile network traffic under the proposed mobile network priority-based queueing (MPQ) scheme with two queueing buffer allocations, namely dynamic queueing buffer (DQB) allocation and overflow queueing buffer (OQB) allocation. As different queueing buffer allocations are adopted to store arriving packets in DQB and OQB, the MPQ scheme shows different packet forwarding performance under these two methods. In this study, we use ns2 (Network Simulator version 2) as the simulation platform to simulate several scenarios. The simulation results show that the MPQ scheme is able to support differentiated packet forwarding behavior for mobile traffic with both DQB and OQB allocations in a mobile core network. Some issues were identified in the MPQ scheme with both DQB and OQB allocation, which will need to be addressed.     

Novel Radio Link Buffer Management Schemes for End-User Multi-class Traffic in High Speed Packet Access Networks

The requirement to provide multimedia services with QoS support in mobile networks has led to standardization and deployment of high speed data access technologies such as the High Speed Downlink Packet Access (HSDPA) system. HSDPA improves downlink packet data and multimedia services support in WCDMA-based cellular networks. As is the trend in emerging wireless access technologies, HSDPA supports end-user multi-class sessions comprising parallel flows with diverse Quality of Service (QoS) requirements, such as real-time (RT) voice or video streaming concurrent with non real-time (NRT) data service being transmitted to the same user, with differentiated queuing at the radio link interface. Hence, in this paper we present and evaluate novel radio link buffer management schemes for QoS control of multimedia traffic comprising concurrent RT and NRT flows in the same HSDPA end-user session. The new buffer management schemes—Enhanced Time Space Priority (E-TSP) and Dynamic Time Space Priority (D-TSP)—are designed to improve radio link and network resource utilization as well as optimize end-to-end QoS performance of both RT and NRT flows in the end-user session. Both schemes are based on a Time-Space Priority (TSP) queuing system, which provides joint delay and loss differentiation between the flows by queuing (partially) loss tolerant RT flow packets for higher transmission priority but with restricted access to the buffer space, whilst allowing unlimited access to the buffer space for delay-tolerant NRT flow but with queuing for lower transmission priority. Experiments by means of extensive system-level HSDPA simulations demonstrates that with the proposed TSP-based radio link buffer management schemes, significant end-to-end QoS performance gains accrue to end-user traffic with simultaneous RT and NRT flows, in addition to improved resource utilization in the radio access network.     

UMTS-to-IP QoS mapping for voice and video telephony services

Quality of service mapping between UMTS services and IP transport is crucial for maintaining a suitable end-to-end delay for emerging UMTS multimedia telephony. However, due to incompatibilities in QoS classifications within these two technologies, straightforward mapping is impossible and current proposals within the 3GPP could lead to unpredictable and undesirable behavior for certain services. In this article we focus on two very important UMTS services, voice and video telephony, and establish the QoS issues that exist for these services. We then propose a refined QoS mapping that differentiates between the transmission of voice and video-telephony and a weighted fair queuing scheduler to schedule the transmissions. Through a simulation study, we show the effect on the queuing delays of both traffic types when their WFQ weights vary and then derive an optimal weight that provides the best overall delays for multimedia telephony services.     

基于CTMC和状态空间模型的宽带无线接入网的QoS性能研究

为了实现同一地域范围内的众多用户在有限带宽条件下提出的高QoS要求,本文对基于IEEE 802.16标准的宽带无线接入网中数据包级QoS（Quality of Service）性能进行了研究.具体做法是,首先采用批马尔可夫到达过程（BMAP,Batch Markov Arrival Process）和连续时间马尔科夫链（CTMC,Continuous Time Markov Chain）对到达过程和流量源进行建模,得到更符合实际和更准确的排队模型;然后基于状态空间,对一个无线接入网络系统进行建模,通过对得到的系统模型并结合前面得到的排队模型的深入分析,从而获得该模型下的各项QoS性能指标,如平均队列长度、丢包率、队列吞吐量和平均包时延.仿真实验结果表明,本文提出的算法模型相比于其他典型的算法模型,能够使得各项QoS性能指标有较大的改善和提高.     

Cross-layer quality-driven adaptation for scheduling heterogeneous multimedia over 3G satellite networks

Wireless networks are experiencing a paradigm shift from focusing on the traditional data transfer to accommodating the rapidly increasing multimedia traffic. Hence, their scheduling algorithms have to concern not only network-oriented quality-of-service (QoS) profiles, but also application-oriented QoS targets. This is particularly challenging for satellite multimedia networks that lack fast closed-loop power control and reliable feedbacks. In this paper, we present a cross-layer packet scheduling scheme, namely Hybrid Queuing and Reception Adaptation (HQRA), which performs joint adaptations by considering the traffic information and QoS targets from the applications, the queuing dynamics induced from the network, as well as the end-to-end performance and channel variations from respective users. By jointly optimizing multiple performance criteria at different layers, the scheme enjoys quality-driven, channel-dependant, and network-aware features. HQRA can well accommodate return link diversity and the imperfect feedbacks, whilst ensuring robustness in highly heterogeneous and dynamic satellite environments. We evaluate its performance over diverse network and media configurations in comparison with the state-of-the-art solutions. We observe noticeable performance gains on application-oriented QoS, bandwidth utilization, and objective video quality, together with favorable fairness and scalability measures.     

一种UMTS核心网中CAC算法

在通用移动通信系统(UMTS)中，无论在无线接入部分还是在核心网部分，QOS的保证都是关键性问题。呼叫准入控制(CAC)是QOS机制的重要组成部分，本文根据3GPP对UMTS业务种类的定义及相应的QOS分类，对UMTS核心网络的QoS准入控制问题进行了研究。基于排队论的思想，提出了一种基于综合排队机制的CAC算法，并对其进行了相应的性能分析和仿真实验，结果表明了本算法的有效性。     

QoS-, Queue-and Channel-Aware Packet Scheduling for Multimedia Services in Multiuser SDMA/TDMA Wireless Systems

With the growing demand for wireless multimedia services and continuing emergence of new multimedia applications, it is necessary for the network to provide various levels of quality of service (QoS) while maximizing the utilization of channel resources. This paper presents an adaptive queuing model and a novel cross-layer packet scheduling algorithm for providing differentiated QoS and effective channel utilization in a space-division-multiple-access/time-division-multiple-access (SDMA/TDMA) system. At the medium access control (MAC) layer, we take into consideration the heterogeneous and bursty nature of multimedia traffic and provide for QoS requirements. At the physical (PHY) layer, we exploit the randomness of the physical channel by incorporating opportunistic scheduling and adopting adaptive modulation and coding (AMC). Performance results obtained by simulations show that by employing the proposed queuing model and packet scheduling algorithm, the system is able to provide for diverse QoS and achieve high throughput.     

Admission control in time-slotted multihop mobile networks

The emergence of nomadic applications have generated a lot of interest in next-generation wireless network infrastructures which provide differentiated service classes. So it is important to study how the quality of service (QoS), such as packet loss and bandwidth, should be guaranteed. To accomplish this, we develop am admission control scheme which can guarantee bandwidth for real-time applications in multihop mobile networks. In our scheme, a host need not discover and maintain any information of the network resources status on the routes to another host until a connection request is generated for the communication between the two hosts, unless the former host is offering its services as an intermediate forwarding station to maintain connectivity between two other hosts. This bandwidth guarantee feature is important for a mobile network to interconnect wired networks with QoS support. Our connection admission control scheme can also work in a stand-alone mobile ad hoc network for real-time applications. This control scheme contains end-to-end bandwidth calculation and bandwidth allocation. Under such a scheme, the source is informed of the bandwidth and QoS available to any destination in the mobile network. This knowledge enables the establishment of QoS connections within the mobile network and the efficient support of real time applications. In the case of ATM interconnection, the bandwidth information can be used to carry out an intelligent handoff between ATM gateways and/or to extend the ATM virtual circuit service to the mobile network with possible renegotiation of QoS parameters at the gateway. We examine via simulation the system performance in various QoS traffic flows and mobility environments     

一种基于无线多媒体业务的可升降级QoS动态带宽分配与优化策略

无线多媒体网络中的业务包括话音、流媒体、交互类和背景类业务4种,除话音业务外其余3种业务都是可变比特速率业务。对该网络用户资源分配(主要是带宽的分配)若采用传统的固定分配方法,必定陷入资源利用率低下和用户QoS得不到保障的两难境地。该文提出了一种基于无线多媒体业务的动态带宽分配与优化策略,在保证用户QoS的前提下,尽可能提高资源利用率。该文分别从网络和用户两个角度考虑,通过系统容量、业务阻塞率、数据延迟、流媒体的实际传输比和VBR业务综合服务等级等参数,对可升降级QoS无线多媒体网络进行了仿真分析,结果表明,对比传统的网络资源管理策略,该策略大大改善了系统的性能,提高了系统资源利用率。     

数据业务端到端QoS参数映射与优化

在移动通信网内采用业务质量(QoS,Quality of Service)机制实现差异化的业务传输是近年的研究热点。端到端QoS包括非接入层用户和业务的识别、不同承载间QoS的一致性传递和接入层QoS的正确执行。这里以EDGE网络中数据业务传送机制为基础,全面研究了3GPP的QoS框架在UMTS网络的GERAN A/Gb模式和UTRAN Iu模式中的QoS映射,给出一个经过优化的端到端QoS参数映射表,并基于该表,在真实的EDGE网络中验证了QoS参数优化的可行性和参数映射的一致性。实验结果表明,根据不同业务的特点采用差异化的传输策略可以提升业务的感知。     

Downlink queuing model and packet scheduling for providing lossless handoff and QoS in AG mobile networks

Next generation mobile networks are expected to provide seamless personal mobile communication and quality-of-service (QoS) guaranteed IP-based multimedia services. Providing seamless communication in mobile networks means that the networks have to be able to provide not only fast but also lossless handoff. This paper presents a two-layer downlink queuing model and a scheduling mechanism for providing lossless handoff and QoS in mobile networks, which exploit IP as a transport technology for transferring datagrams between base stations and the high-speed downlink packet access (HSDPA) at the radio layer. In order to reduce handoff packet dropping rate at the radio layer and packet forwarding rate at the IP layer and provide high system performance, e.g., downlink throughput, scheduling algorithms are performed at both IP and radio layers, which exploit handoff priority scheduling principles and take into account buffer occupancy and channel conditions. Performance results obtained by computer simulation show that, by exploiting the downlink queuing model and scheduling algorithms, the system is able to provide low handoff packet dropping rate, low packet forwarding rate, and high downlink throughput.     

Transport of TCP/IP traffic over assured forwarding IP-differentiated services

The Internet is facing a twofold challenge: to increase network capacity in order to accommodate a steadily increasing number of users; to guarantee the quality of service for existing applications and for new multimedia applications requiring real-time network response. In order to meet these requirements, IETF is currently defining the differentiated service (DiffServ) architecture, which should offer a simple and scalable platform to guarantee differentiated QoS in the Internet. In the DiffServ domain, the assured forwarding service is designed to provide data applications with acceptable performance, overcoming the limits of the Internet's current best-effort service. Since data applications mostly rely on the TCP transport protocol, it is important to examine the interaction between the congestion avoidance and control mechanisms of TCP and assured forwarding. Our main purpose is to shed light on this interaction, and to show that, in the current DiffServ framework, poor performance of TCP traffic flows can result from the existing mismatch between the assured forwarding traffic conditioning procedures and the TCP congestion management. We propose a new adaptive packet marking policy to deal with congestion situations that may occur. We show that, with this policy, the provisioned rate for TCP flows can be achieved.     

Cross-Layer Design of 2D Queuing Model for Multi-hop Wireless Networks

Based on cross-layer design, a modified 2-dimensional queuing model (2DQM) is proposed in this paper to tackle the problem of end-to-end quality of service (QoS) metric calculation. This model exploits the traffic arrival process, multi-rate transmission in the physical layer and error recovery technology with the protocol of truncated automatic repeat request in the data link layer. Based on this model, QoS metrics of wireless links can be evaluated hop by hop. The model can be used in more realistic scenarios of multi-hop wireless networks, although the computational complexity of 2DQM is slightly higher compared with existing 1-dimensional queuing model. Simulation results indicate that the proposed model can estimate the end-to-end packet loss-rate and average delay more accurately than existing models, and a model based QoS routing algorithm can find routes with better QoS performance (with lower end-to-end packet loss-rate and delay).     

Fair Class-Based Downlink Scheduling with Revenue Considerations in Next Generation Broadband Wireless Access Systems

The success of emerging Broadband Wireless Access Systems (BWASs) will depend, among other factors, on their ability to manage their shared wireless resources in the most efficient way. This is a complex task due to the heterogeneous nature, and hence, diverse Quality of Service (QoS) requirements of different applications that these systems support. Therefore, QoS provisioning is crucial for the success of such wireless access systems. In this paper, we propose a novel downlink packet scheduling scheme for QoS provisioning in BWASs. The proposed scheme employs practical economic models through the use of novel utility and opportunity cost functions to simultaneously satisfy the diverse QoS requirements of mobile users and maximize the revenues of network operators. Unlike existing schemes, the proposed scheme is general and can support multiple QoS classes with users having different QoS and traffic demands. To demonstrate its generality, we show how the utility function can be used to support three different types of traffic, namely best-effort traffic, traffic with minimum data rate requirements, and traffic with maximum packet delay requirements. Extensive performance analysis is carried out to show the effectiveness and strengths of the proposed packet scheduling scheme.     

Analysis of Distributed DDQ for QoS Router

In a packet switching network, congestion is unavoidable and affects the quality of real‐time traffic with such problems as delay and packet loss. Packet fair queuing (PFQ) algorithms are well‐known solutions for quality‐of‐service (QoS) guarantee by packet scheduling. Our approach is different from previous algorithms in that it uses hardware time achieved by sampling a counter triggered by a periodic clock signal. This clock signal can be provided to all the modules of a routing system to get synchronization. In this architecture, a variant of the PFQ algorithm, called digitized delay queuing (DDQ), can be distributed on many line interface modules. We derive the delay bounds in a single processor system and in a distributed architecture. The definition of traffic contribution improves the simplicity of the mathematical models. The effect of different time between modules in a distributed architecture is the key idea for understanding the delay behavior of a routing system. The number of bins required for the DDQ algorithm is also derived to make the system configuration clear. The analytical models developed in this paper form the basis of improvement and application to a combined input and output queuing (CIOQ) router architecture for a higher speed QoS network.     

Performance evaluation of the RQMA-CDMA scheme for multimedia traffic with QoS guarantees

The rapid growth of cellular mobile technology in recent years, coupled with the explosive growth of the Internet, has significantly increased the demand for wireless data services. Traffic on mobile devices is expected to be a mix of real-time traffic such as video, voice, and data, with users requiring diverse quality of service (QoS) guarantees for different types of traffic (video, voice and data). One of the primary challenges of providing QoS is how to prioritize and allocate network resources among contending applications. In order to achieve these goals, a scheduling scheme that can provide equitable and effective packet routing is required. This paper proposes a scheduling scheme called remote queuing multiple access-code division multiple access (RQMA-CDMA), whose purpose is to equitably assign bandwidth resources with QoS guarantees to different mobile devices. RQMA-CDMA is a rate scheduling scheme that can be used to assign bandwidth resources in conjunction with GPS (generalized processor sharing). Additionally, we analyze an admission control that is based on signal to interference plus noise ratio (SINR) for multimedia traffic. Finally, the performance of RQMA-CDMA is evaluated and compared to schemes based on CDMA-GPS according to dropped packets, delay, and throughput.     

无线网络中基于优先级队列延迟限定的研究

由于无线网络环境下网络节点的增加,网络延时成为一个亟待解决的问题。为了提高服务质量(QoS),提高吞吐量,文中提出了一种基于优先级的队列延迟模型,通过将每一个包预设置优先级来区分其重要性和实时性,同时将每一个AP设备中的队列根据优先级划分为3种类型,并将预设优先级的包放入其中进行传输,从而有效减少发送端的队列延迟。通过分析和仿真可以发现,与未划分优先级队列的节点网络相比,这种方案不仅使单个节点的延迟大幅减少,也使整个网络的平均延迟明显降低,网络整体性能显著提高。     

Radio resource management for cellular CDMA systems supporting heterogeneous services

A novel radio resource management (RRM) scheme for the support of packet-switched transmission in cellular CDMA systems is proposed by jointly considering the physical, link, and network layer characteristics. The proposed resource management scheme is comprised of a combination of power distribution, rate allocation, service scheduling, and connection admission control. Power distribution allows individual connections to achieve their required signal-to-interference-plus-noise ratio, while rate allocation guarantees the required delay/jitter for real-time traffic and the minimum transmission rate requirement for non-real-time traffic. Efficient rate allocation is achieved by making use of the randomness and burstiness; of the packet generation process. At the link layer, a packet scheduling scheme is developed based on information derived from power distribution and rate allocation to achieve quality of service (QoS) guarantee. Packet scheduling efficiently utilizes the system resources in every time slot and improves the packet throughput for non-real-time traffic. At the network layer, a connection admission control (CAC) scheme based on the lower layer resource allocation information is proposed. The CAC scheme makes use of user mobility information to reduce handoff connection dropping probability (HCDP). Theoretical analysis of the grade of service performance, in terms of new connection blocking probability, HCDP, and resource utilization, is given. Numerical results show that the proposed RRM scheme can achieve both effective QoS guarantee and efficient resource utilization.