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 a Differentiated Service Queuing Scheme Based on Transmitted Bytes Within a UMTS Core Network Gateway

Universal mobile telecommunications system (UMTS) is a popular 3G system to support network applications full of variety. According to the quality of service (QoS) features of four types of UMTS traffic, this study proposes a BBQ (byte-based queuing) scheme to handle a UMTS packet forwarding process in differentiation. With the BBQ scheme, four types of UMTS traffic depend on their QoS features to receive differentiated packet forwarding performance within a UMTS core network. Several scenarios are simulated to realize the packet transmission performance of the BBQ scheme. Moreover, a packet transmission performance comparison between the IETF RIO scheme and the BBQ scheme is discussed in this study. The simulation results show that a differentiated packet forwarding behavior among UMTS traffic can be provided by the BBQ scheme with low cost operation/architecture; this demonstrate the effectiveness of the proposed queuing scheme.     

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.     

A cross-layer optimized adaptive modulation and coding scheme for transmission of streaming media over wireless links

In this paper, a novel cross-layer Adaptive Modulation and Coding scheme that optimizes the overall packet loss (by both transmission errors and excessive delays) probability under a given arrival process is developed. To this end, an improved Large Deviations approximation for the fraction of packets that suffer from excessive queuing delay is proposed. This approximation is valid for G/G/1 queues with infinite buffers that are driven by stationary arrival and service processes which satisfy certain conditions. Such models can capture the time correlations in the amount of traffic generated by streaming media sources and the time varying service capacity of a wireless link. Through numerical examples, the proposed AMC policy is shown to achieve a significant reduction in the overall packet loss rate compared to previously proposed schemes. This algorithmic performance gain can be translated into a sizeable decrease in the required transmit power or an analogous increase in the rate of the arrival process, subject to a given maximum packet loss rate Quality of Service constraint. Furthermore, the proposed AMC policy can be combined with ARQ in order to achieve an even lower overall packet loss probability.     

SEAL2: An SDN‐enabled all‐Layer2 packet forwarding network architecture for multitenant datacenter networks

This paper presents the design and development of a new network virtualization scheme to support multitenant datacenter networking (MT‐DCN) based on software‐defined networking (SDN) technologies. Effective multitenancy supports are essential and challenging for datacenter networking designs. In this study, we propose a new network virtualization architecture framework for efficient packet forwarding in MT‐DCN. Traditionally, an internet host uses IP addresses for both host identification and location information, which causes mobile IP problems whenever the host is moved from one IP subnet to another. Unfortunately, virtual machine (VM) mobility is inevitable for cloud computing in datacenters for reasons such as server consolidation and network traffic flow optimization. To solve the problems, we decouple VM identification and location information with two independent values neither by IP addresses. We redefine the semantics of Ethernet MAC address to embed tenant ID information to the MAC address field without violating its original functionality. We also replace traditional Layer2/Layer3 two‐stage routing schemes (MAC/IP) with an all‐Layer2 packet forwarding mechanism that combines MAC addresses (for VM identification and forwarding in local server groups under an edge switch gateway) and multiprotocol label switching (MPLS) labels (for packet transportation between edge switch gateways across the core label switching network connecting all the edge gateways). To accommodate conventional IP packet architecture in a multitenant environment, SDN (OpenFlow) technology is used to handle all this complex network traffics. We verified the design concepts by a simple system prototype in which all the major system components were implemented. Based on the prototype system, we evaluated packet forwarding efficiency under the proposed network architecture and compared it with conventional IP subnet routing approaches. We also evaluated the incurred packet processing overhead caused by each of the packet routing components.     

CCN中基于业务类型的多样化内容分发机制

针对内容中心网络(Content Centric Networking,CCN)如何提供差异化的业务需求服务的问题,采用区分服务的思想,从内容传输和缓存决策的角度出发,提出了一种基于业务类型的多样化内容分发机制.该机制依据不同的业务请求特征,分别设计了持久推送、并行预测和逐包请求的数据分发模式,对应提出了透明转发、边缘概率缓存和渐进式推进的沿途存储策略,实现了内容传递对于业务类型的感知和匹配.仿真结果表明,该机制减小了内容请求时延,提高了缓存命中率,以少量额外的控制开销提升了CCN网络整体的内容分发性能.     

Seamless Handoff Scheme Based on Pre-registration and Pre-authentication for UMTS–WLAN Interworking

Interworking issues between Universal Mobile Telecommunication System (UMTS) and Wireless Local Area Network (WLAN) have become a great matter of interest as an increasing number of mobile internet users require broadband wireless access to IP services in the wide area. In this paper, we present a practical UMTS–WLAN interworking architecture based on 3GPP standards and propose a seamless handoff scheme that guarantees low delay and low packet loss during UMTS–WLAN handoff. For low handoff delay, the proposed handoff scheme performs pre-registration and pre-authentication processes before layer 2 handoff. Moreover, it uses packet buffering and forwarding functions in order to reduce packet loss during the handoff period. On the above basis, detailed signaling procedures are presented, together with system requirements when a mobile terminal moves from UMTS to WLAN and vice-versa. Numerical results show that the proposed scheme performs well with respect to signaling cost, handoff delay, and packet loss compared with conventional schemes.     

Towards a better-than-best-effort forwarding service for multimediaflows

A salient requirement of interactive multimedia applications is that they transmit data continuously at uniform rates with minimum possible end-to-end delay. The majority of these applications do not require hard and fast guarantees of network performance, but the current best-effort forwarding model of the Internet is frequently insufficient for realizing these requirements. Worse still, the requirement of uniform-rate transmission puts many multimedia applications at odds with current and proposed Internet network management practices that assume or require TCP-like reactions to packet loss. We are investigating router-based active queue management, specifically the use of queue occupancy thresholds to isolate TCP flows and to provide a better-than-best-effort forwarding service for flows in need of uniform-rate transmissions. Our current scheme, class-based thresholds (CBT), relies on a packet marking mechanism such as those proposed for realizing differentiated services on the Internet. CBT, when combined with existing active router queue management schemes such as random early detection (RED), provides a performance for TCP that approximates that achievable under a packet scheduling scheme and acceptable performance for multimedia flows. CBT is a simple and efficient mechanism with implementation complexity and run-time overhead comparable to that of RED     

Performance analysis of wireless multi-user VoIP system with adaptive modulation and coding

In this paper, we propose a new framework to analyze performance considering finite-length queuing and adaptive modulation and coding for multi-user Voice over IP (VoIP) services in wireless communication systems. We formulate an uplink VoIP system as a two-dimensional discrete-time Markov chain (DTMC) based on a Markov modulated Poisson process traffic model for VoIP services and modulation and coding scheme (MCS)-level set transition reflecting users’ channel variations. We extend the transition modeling of the MCS-level for a single-user to the transition modeling of the MCS-level set for multiple users. Since the users can have various MCS combinations in the case of a multi-user system, the MCS-level set transitions are more complicated than the MCS-level transitions of the single-user case. Throughout our DTMC formulation, we present various performance metrics, such as average queue-length, average throughput, packet dropping probability, packet loss probability, and so on. By using the results of the packet loss probability, we can find an optimum packet error rate value that minimizes the total packet loss probability.     

Optimized Packet Scheduling Management: Maximizing Bandwidth Utilization for Next-Generation Mobile Multimedia Communications

Base stations in next-generation broadband mobile networks (NGBMNs) must efficiently schedule different kinds of multimedia packets providing different quality of service (QoS) classes. During the past 10?years, many researchers have experimented with various packet scheduling schemes. In this paper we will propose a batch-arrival queuing model for evaluating NGBMN multimedia packet scheduling systems, and for obtaining three performance measures: packet loss rate (PLR), queuing delay (QD), and bandwidth utilization (BU). The three measures can be used to solve utilization optimization problems with QoS constraints. Specifically, a combination of a traffic statistic plus maximum PLR and QD constraints can be used to maximize BU for a multimedia packet scheduling management architecture. According to results from mathematical tests of the proposed model, it offers an efficient approach to managing scheduling buffers. The model and optimized parameters can be applied to flexible bandwidth deployment and classified buffer size control, thus enhancing profitability.     

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.     

Load Balancing for Parallel Forwarding

Workload distribution is critical to the performance of network processor based parallel forwarding systems. Scheduling schemes that operate at the packet level, e.g., round-robin, cannot preserve packet-ordering within individual TCP connections. Moreover, these schemes create duplicate information in processor caches and therefore are inefficient in resource utilization. Hashing operates at the flow level and is naturally able to maintain per-connection packet ordering; besides, it does not pollute caches. A pure hash-based system, however, cannot balance processor load in the face of highly skewed flow-size distributions in the Internet; usually, adaptive methods are needed. In this paper, based on measurements of Internet traffic, we examine the sources of load imbalance in hash-based scheduling schemes. We prove that under certain Zipf-like flow-size distributions, hashing alone is not able to balance workload. We introduce a new metric to quantify the effects of adaptive load balancing on overall forwarding performance. To achieve both load balancing and efficient system resource utilization, we propose a scheduling scheme that classifies Internet flows into two categories: the aggressive and the normal, and applies different scheduling policies to the two classes of flows. Compared with most state-of-the-art parallel forwarding schemes, our work exploits flow-level Internet traffic characteristics.     

A packet scheduling scheme for 4G wireless access systems aiming to maximize revenue for the telecom carriers

Efficient utilization of network resources is a key goal for emerging BWAS. This is a complex goal to achieve due to the heterogeneous service nature and diverse QoS requirements of various applications that BWAS support. Packet scheduling is an important activity that affects BWAS QoS outcomes. This paper proposes a new packet scheduling mechanism that improves QoS in mobile wireless networks which exploit IP as a transport technology for data transfer between BWAS base stations and mobile users at the radio transmission layer. In order to improve BWAS QoS the new packet algorithm makes changes at both the IP and the radio layers. The new algorithm exploits handoff priority scheduling principles and takes into account buffer occupancy and channel conditions. The packet scheduling mechanism also incorporates the concept of fairness. The algorithm offers an opportunity to maximize the carriers’ revenue at various traffic situations. Simulation results were compared to well-known algorithms which demonstrated the new packet scheduling algorithm is able to provide a low handoff packet drop rate, low packet forwarding rate, low packet delay, ensure fairness amongst the users of different services and generates higher revenue. Furthermore this research proposes a new and novel measure named “Satisfaction Factor” to measure the efficacy of various scheduling schemes and finally proposes four performance metrics for NodeB’s of in Next Generation Wireless Networks.     

Quality‐of‐Service Mechanisms for Flow‐Based Routers

In this paper, we propose quality of service mechanisms for flow‐based routers which have to handle several million flows at wire speed in high‐speed networks. Traffic management mechanisms are proposed for guaranteed traffic and non‐guaranteed traffic separately, and then the effective harmonization of the two mechanisms is introduced for real networks in which both traffic types are mixed together. A simple non‐work‐conserving fair queuing algorithm is proposed for guaranteed traffic, and an adaptive flow‐based random early drop algorithm is proposed for non‐guaranteed traffic. Based on that basic architecture, we propose a dynamic traffic identification method to dynamically prioritize traffic according to the traffic characteristics of applications. In a high‐speed router system, the dynamic traffic identification method could be a good alternative to deep packet inspection, which requires handling of the IP packet header and payload. Through numerical analysis, simulation, and a real system experiment, we demonstrate the performance of the proposed mechanisms.     

Adaptive Fair Subcarrier/Rate Allocation in Multirate OFDMA Networks: Radio Link Level Queuing Performance Analysis

This paper presents a semi-analytical methodology for radio link level performance analysis in a multirate "orthogonal frequency-division multiple-access" (OFDMA) network with adaptive fair rate allocation. Multirate transmission is assumed to be achieved through adaptive modulation, and fair rate allocation is based on the principle of generalized processor sharing to allocate the subcarriers adaptively among the users. The fair rate allocation problem is formulated as an optimization problem with the objective of maximizing system throughput while maintaining fairness (in terms of transmission rate) among the users. The "optimal" fair rate allocation is obtained by using the "Hungarian method." A heuristic-based approach, namely the "iterative approach," that is more implementation friendly is also presented. The throughput performance of the iterative fair rate allocation is observed to be as good as that of optimal fair rate allocation and is better than that of the static subcarrier allocation scheme. Also, the iterative fair allocation provides better fairness compared to that for each of the optimal and the static subcarrier allocation schemes. To this end, a queuing model is formulated to analyze radio link level performance measures such as packet dropping probability and packet transmission delay under the above rate allocation schemes. In this formulation, packet arrivals are modeled by the discrete Markov modulated Poisson process, which is flexible to model different types of traffic arrival patterns. The proposed framework for radio link level performance analysis of multirate OFDMA networks is validated by extensive simulations. Also, examples on the application of the proposed model for connection admission control and quality-of-service provisioning are illustrated     

Core‐stateless fair rate estimation fair queuing

Core‐stateless mechanisms, such as core‐stateless fair queuing (CSFQ), reduce the complexity of fair queuing, which usually need to maintain states, manage buffers, and perform flow scheduling on a per‐flow basis. However, they require executing label rewriting and dropping decision on a per‐packet basis, thus preventing them from being widely deployed. In this paper, we propose a novel architecture based on CSFQ without per‐packet labelling. Similarly, we distinguish between edge routers and core routers. Edge routers maintain the per‐flow state by employing a fair queuing mechanism to allocate each flow a fair bandwidth share locally and a token bucket mechanism to regulate those flows with feedback packets sent from egress edge routers. Core routers do not maintain per‐flow state; they use FIFO packet scheduling extended by a fare rate alarm mechanism by estimating the arrival rate and the number of flows using a matching–mismatching algorithm. The novel scheme is called core‐stateless fair rate estimation fair queuing (CSFREFQ). CSFREFQ is proven to be capable of achieving max–min fairness. Furthermore, we present and discuss simulations and experiments on the performance under different traffic scenarios. Copyright © 2005 John Wiley & Sons, Ltd.     

Soft QoS provisioning using the token bank fair queuing scheduling algorithm

Future-generation wireless packet networks will support multimedia applications with diverse QoS requirements. Much of the research on scheduling algorithms has been focused on hard QoS provisioning of integrated services. Although these algorithms give hard delay bounds, their stringent requirements sacrifice the potential statistical multiplexing performance and flexibility of the packet-switched network. Furthermore, the complexities of the algorithms often make them impractical for wireless networks. There is a need to develop a packet scheduling scheme for wireless packet-switched networks that provides soft QoS guarantees for heterogeneous traffic, and is also simple to implement and manage. This article proposes token bank fair queuing (TBFQ), a soft scheduling algorithm that possesses these qualities. This algorithm is work-conserving and has a complexity of O(1). We focus on packet scheduling on a reservation-based TDMA/TDD wireless channel to service integrated real-time traffic. The TBFQ scheduling mechanism integrates the policing and servicing functions, and keeps track of the usage of each connection. We address the impact of TBFQ on mean packet delay, violation probability, and bandwidth utilization. We also demonstrate that due to its soft provisioning capabilities, the TBFQ performs rather well even when traffic conditions deviate from the established contracts.     

Quality of experience (QoE) driven adaptation scheme for?voice/video over IP

Network quality of service (NQoS) of IP networks is unpredictable and impacts the quality of networked multimedia services. Adaptive voice and video schemes are therefore vital for the provision of voice over IP (VoIP) services for optimised quality of experience (QoE). Traditional adaptation schemes based on NQoS do not take perceived quality into consideration even though the user is the best judge of quality. Additionally, uncertainties inherent in NQoS parameter measurements make the design of adaptation schemes difficult and their performance suboptimal. This paper presents a QoE-driven adaptation scheme for voice and video over IP to solve the optimisation problem to provide optimal QoE for networked voice and video applications. The adaptive VoIP architecture was implemented and tested both in NS2 and in an Open IMS Core network to allow extensive simulation and test-bed evaluation. Results show that the scheme was optimally responsive to available network bandwidth and congestion for both voice and video and optimised delivered QoE for different network conditions, and is friendly to TCP traffic.     

Cost models for QoS-differentiated interconnecting and wholesale access services in?future generation networks

Currently, network operators and Internet service providers are offering ??Triple Play?? products integrating services with different Quality of Service (QoS) requirements. It is leading to Internet traffic with strong service integration under an all-IP-based broadband network platform. However, new multimedia service offers require individual QoS guarantees for each type of services. The interconnection between different providers necessitates the reconsideration of the actual cost schemes. Interconnection and wholesale access services (It is an extension of ??wholesale network?? definition, where Telco??s physical network and equipment are ??shared?? to many independent Service Providers. If the incumbent offers broadband access services, the rest of the alternative providers have recourse to the incumbent??s ??wholesale access service??. Bitstream service is the most important service of this type, actually regulated over DSL and cable networks.) appear to be a simple solution, but the consideration of QoS parameters requires an extension of the current network dimensioning methods based mainly on the average bandwidth demand from each user. This paper proposes a cost model which considers QoS parameters and, based on the ??Total Element based Long Run Incremental Cost?? (TELRIC) model, is applied to the wholesale access and interconnection paradigm. Three traffic engineering methods are considered and studied for network dimensioning. Hereby the aim is to guarantee the QoS of the different services: complete traffic segregation under virtual tunnels, complete traffic integration by over-engineering and partial traffic integration using a priority queuing scheme. The proposed method enables the development of a specific cost scheme based on a complete scenario considering different types of users. The variety of used IP applications suppose direct implications over different levels of interconnection, mainly at the low-level Metro access and the high-level edge node.     

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.