To simultaneously support guaranteed real-time services and best-effort service, a Priority-based Scheduling Architecture (PSA) designed for high-speed switches is proposed. PSA divides packet scheduling into high-priority phase and low-priority phase. In the high-priority phase, an improved sorted-priority algorithm is presented. It introduces a new constraint into the scheduling discipline to overcome bandwidth preemption. Meanwhile, the virtual time function with a control factor α is employed. Both computer simulation results and theoretic analysis show that the PSA mechanism has excellent performance in terms of the implementation complexity, fairness and delay properties.

There are many system proposals for satellite-based broadband communications that promise high capacity and ease of access. Many of these proposals require advanced switching technology and signal processing on-board the satellite(s). One solution is based on a geo-synchronous (GEO) satellite system equipped with on-board processing and on-board switching. An important feature of this system is allowing for a maximum number of simultaneous users, hence, requiring effective medium access control (MAC) layer protocols for connection admission control (CAC) and bandwidth on demand (BoD) algorithms. In this paper, an integrated CAC and BoD algorithm is proposed for a broadband satellite communication system with heterogeneous traffic. A detailed modeling and simulation approach is presented for performance evaluation of the integrated CAC and BoD algorithm based on heterogeneous traffic types. The proposed CAC and BoD scheme is shown to be able to efficiently utilize available bandwidth and to gain high throughput, and also to maintain good Grade of Service (GoS) for all the traffic types. The end-to-end delay for real-time traffic in the system falls well within ITU's Quality of Service (QoS) specification for GEO-based satellite systems.

Time synchronization is indispensable for wide area distributed systems including sensor networks, automation systems, and measurement/control systems. Another application is clock distribution, which is indispensable to support continuous information transfer. Because of the increasing demand for more sophisticated applications, it is essential to establish a time synchronization technique that offers higher accuracy and reliability. Particularly, the accuracy of time synchronization for Ethernet must be enhanced since Ethernet is becoming more important in telecommunication networks. This paper investigates a precise time synchronization technique that supports Gb/s Ethernet. To obtain accurate time synchronization, delay variation in message transfer and processing must be minimized. For this purpose, the paper first describes the implementation of preemptive priority queuing, which decreases the message delay variation of Ethernet. Through experiments, it is shown that preemptive priority queuing effectively achieves very low delay variation. The paper then proposes a method to synchronize the time signal of a slave node to that of the master node. The proposed time synchronization method is performed in the lower protocol layer and implemented on FPGA-based hardware. The method achieves superior time accuracy through the low message transfer/processing delay variation provided by preemptive priority, lower layer execution, and hardware implementation. The effectiveness of the method is confirmed through experiments. The experiments show that the time variation achieved by the method is smaller than 0.1 µsec. This performance is better than those obtained by existing synchronization methods.

Supporting diversified rates for real-time communications will become possible and essential with the rapidly increasing transmission rates provided by the 4th generation (4G) mobile communication systems. In this paper, a novel wireless Quality of Service (QoS) scheme suitable for broadband CDMA packet cellular systems with adaptive modulation coding is proposed and its characteristics are described. The proposed QoS scheme comprises several control factors laid on the MAC and RRC layers, and can be harmonized with IP-QoS. Two important control factors are proposed: radio-condition-aware admission control and resource allocation reflected multistage scheduling. Computer simulations and testbed experiments indicate that by using the radio-condition-aware admission control, stable and guaranteed service can be provided to real-time users regardless of the interference and the variation in the location of the mobile station. Moreover, resource allocation reflected multistage scheduling maintains guaranteed rates for real-time users and provides high resource utilization efficiency for best-effort users. Consequently, by using the proposed wireless QoS scheme, it is possible to provide users with high quality and diversified real-time services, on a packet based radio network for enhanced 3G and beyond.

This paper describes an overview of overall carrier-grade Ethernet technologies for next generation wide area Ethernet. In recent years, from access network to metro and core network, we can find many areas where communication services are provided by Ethernet technologies. This comes from the fact that operational efficiency and economical efficiency of Ethernet are far better than that of conventional wide area communication technologies such as SONET and ATM. On the other hand, carrier-grade reliability, operations-administration-maintenance (OAM) and quality of service (QoS) are inferior to SONET and ATM. Various standard schemes in IEEE 802 and ITU-T and vendors' proprietary schemes can leave various approaches to solve these problems. In this paper, the author explains a basic architecture of wide area Ethernet service (Q-in-Q tagging for metro network and Mac-in-Mac encapsulation for core network) at first. Various switch control technologies are then discussed which are deployed or are under evaluation in order to improve (i) reliability (i.e., resiliency) to protect subscribers against network failures, (ii) OAM for providers to perform fault and performance management, and (iii) QoS to guarantee subscriber's service level agreement between a carrier and a subscriber. Finally, a new switching architecture, Global Open Ethernet (GOE), is also introduced as one of promising approaches to realize a next generation carrier-grade Ethernet.

This letter studies the effect of node mobility on application-level QoS of audio-video multipath streams in wireless ad hoc networks. The audio-video streams are transmitted with the MultiPath streaming scheme with Media Synchronization control (MPMS), which was previously proposed by the authors. We perform computer simulation with a grid topology network of IEEE 802.11b including two mobile nodes. The simulation results show that MPMS is effective in achieving high application-level QoS in mobile networks as well.

This paper describes the author's perspective on multimedia quality prediction methodologies for multimedia communications in advanced mobile and internet protocol (IP)-based telephony, and reports related experiments and trials. First, the paper describes the need for perceptual QoS (Quality of Service) assessment in which various quality factors in multimedia communications for advanced mobile and IP-based telephony are analyzed. Then an objective quality prediction scheme is proposed from the viewpoints of quality measurement tools for each quality factor and an opinion model for compound quality factors in mobile and IP-based communications networks. Finally, the author's current trials of measurement tools and opinion models are described.

This paper proposes a framework for session-level SLA (Service Level Agreement) and network-level SLA management to provide QoS-oriented application services over DiffServ/MPLS networks. DiffServ and MPLS technologies enhance the capability of QoS guarantee on IP network, and application service provider can provide QoS-oriented application services to their customers based on the transport networks. The example of using our approaches in the VoIP service involving the network service provider, the VoIP service provider, and the VoIP customers are examined. The session-level SLA between VoIP service provider and VoIP customer and the network-level SLA (N-SLA) between network provider and VoIP service provider are constructed in this paper. Besides, the VoIP service provider faces the QoS-mapping issue and the balance between revenue and cost, i.e., how to contract the N-SLA. Therefore, we introduce an off-line SLA evaluation scheme, a heuristic optimization algorithm and an on-line SLA process method to provide VoIP service policy, and then the optimal QoS-mapping can be resolved. The concept of this framework of session-level SLA and network-level SLA management can be extended easily into other real-time multimedia and non-real time data services.

In this paper, we propose a novel dynamic bandwidth allocation scheme for the downlink real-time video streaming in the wireless cellular networks. Our scheme is able to maximize the bandwidth utilization, while satisfying the required packet loss probability, a QoS constraint, by dynamically determining the amount of bandwidth to be allocated at each unit time interval by measuring the queue length and the packet loss probability. The simulation results show that, without the need of a priori knowledge about the traffic traces, our scheme is able to achieve the same level of performance as what can be accomplished with the pre-calculated effective bandwidth in terms of the bandwidth utilization and the packet loss rate.

As an alternative solution to provide the quality of services (QoS) for broadband access over Ethernet Passive Optical Network (EPON), we present the usage of MAC control message for plural class queues and a traffic-class burst-polling based delta dynamic bandwidth allocation (DBA), referred to as TCBP-DDBA, scheme. For better QoS support, the TCBP-DDBA minimizes packet delays and delay variations for expedited forwarding packet and maximizes throughput for assured forwarding and best effort packets. The network resources are efficiently utilized and adaptively allocated to the three traffic classes for the given unbalanced traffic conditions by guaranteeing the requested QoS. Simulation results using OPNET show that the TCBP-DDBA scheme performs well in comparison to the conventional unit-based allocation scheme over the measurement parameters such as: packet delay, packet delay variation, and channel utilization.

We propose a probing-based channel adaptive video streaming method to provide seamless video transmission over the wireless 3G network. In the proposed method, the available bandwidth (AB) of the end-to-end path is estimated by analyzing RTP packets at the streaming client and the network/client buffer status (NCBS) is predicted by examining the RTCP receiver report (RR) and the application defined packet (APP) including the AB information. Using the estimated AB, the NCBS, and the stored multirate bitstream information, the proposed streaming method determines the next transmission bitrate precisely. Experimental results indicate that the proposed method can be effectively used for video streaming.

Admission control is becoming an essential technique for IP networks to provide full-fledged multimedia streaming services. Although signaling-based schemes are utilized to achieve this, these are difficult to deploy and can hardly achieve strict admission control taking the properties of packet arrival into consideration. In this paper, we propose a novel admission control strategy called the Tentative Accommodating and Congestion Confirming Strategy (TACCS). The main idea is to accommodate incoming flows tentatively and confirm congestion after a certain period. Since tentative accommodating enables us to generate the same situation as where incoming flows have been accommodated, TACCS makes it possible to control admission considering the properties of packet arrival after they have been accommodated, without collecting resource information in advance. From the results of mathematical analysis, we confirmed that TACCS enabled a domain to control admission without a centralized management agent and we provided guidelines for configuring parameters of TACCS.

This paper proposes a subjective model for estimating the quality of video streaming services with dynamic bit-rate control. In a subjective quality assessment test, we clarify users' perceptions of distributed video signals whose quality is time-variant due to dynamic bit-rate control. Using this result, we constructed an estimation model considering the following three characteristics: 1) the influence of the video section where quality degradation is large will strongly affect the overall quality, 2) the impression of a past quality weakens with the passage of time, and 3) the range of evaluation scores becomes wider when the time duration of an evaluation is longer. We found that the proposed model enables the accuracy of estimating overall subjective quality to be dramatically improved compared with that of a model that averages segmental quality. The estimation error of the proposed model is less than the statistical reliability of the subjective score even for verification data. We also show that our findings are applicable to QoS design/management issues for video streaming services with dynamic bit-rate control.

Seamless interconnection with wireless LAN and 3G technologies is essential for the future wireless environment. The Wireless Local Area Network (WLAN) integrated mobile device is designed to extend the reach of enterprise applications and to create new collaboration environments. Also rapid new service development has started to change traffic mixes in the cellular networks towards IP dominating carriers like GPRS (General Packet Radio Service) and WLAN. These drastic changes require new research in the network as well as system inter-working areas for both the cellular and WLAN technology areas. This is the main challenge our research is trying to solve giving answers to rising inter-working and interoperability questions.

This paper analyzes the user-perceived QoS (Quality of Service) of interactive data application, the most notable being web browsing, in a shared wireless packet access network. We present elaborate analytical results regarding the mean page delay, the Equivalent Circuit Rate (ECR), and the system saturation point for interactive data users with different channel conditions. The ECR for a user in a shared access network is the channel rate that a dedicated connection would need in order to offer an equivalent user experience. The system saturation point is the maximum number of users that can be served with a given tolerable service delay in a shared packet access system. These analytical results are useful in the performance prediction of a real wireless environment with multiple user groups that are served with different service rates due to multiple channel conditions. Therefore, our analytical results can be used in the network planning, admission control, and service differentiation for different user groups. Our analysis is demonstrated to be in good accord with computer simulations performed on model systems based on EDGE TDMA and IEEE 802.16 OFDMA.

This paper presents a way of using a linear regression model to produce a single-valued criterion that indicates the perceived importance of each block in a stream of speech blocks. This method is superior to the conventional approach, voice activity detection (VAD), in that it provides a dynamically changing priority value for speech segments with finer granularity. The approach can be used in conjunction with scalable speech coding techniques in the context of IP QoS services to achieve a flexible form of quality control for speech transmission. A simple linear regression model is used to estimate a mean opinion score (MOS) of the various cases of missing speech segments. The estimated MOS is a continuous value that can be mapped to priority levels with arbitrary granularity. Through subjective evaluation, we show the validity of the calculated priority values.

This investigation proposes a virtual-FIFO (VFIFO) back-off algorithm for wireless networks. The proposed scheme takes advantage of the central unit (CU) in a wireless network to broadcast a common back-off window size to all the users, significantly alleviating the unfairness of bandwidth utilization in conventional binary exponential back-off (BEB) algorithms. The proposed scheme exploits the CU's capability for collision detection to estimate the number of simultaneously competing users. Additionally, packets generated in a given cycle are split into groups according to their times of arrivals and are guaranteed to be serviced one after another within the next cycle. Although the proposed algorithm is not strictly first come fist served, the FIFO principle is virtually accomplished. Simulation results demonstrate that the standard deviation of delay can be improved by more than two orders and the throughput can be maintained at 0.42 when the number of users approaches infinity. The capture effect even further improves system performance.

We present an efficient method to optimize network resource allocations under nonlinear Quality of Service (QoS) constraints. We first propose a suite of generalized proportional allocation schemes that can be obtained by minimizing the information-theoretic function of relative entropy. We then optimize over the allocation parameters, which are usually design variables an engineer can directly vary, either for a particular user or for the worst-case user, under constraints that lower bound the allocated resources for all other users. Despite the nonlinearity in the objective and constraints, we show this suite of resource allocation optimization can be efficiently solved for global optimality through a convex optimization technique called geometric programming. This general method and its extensions are applicable to a wide array of resource allocation problems, including processor sharing, congestion control, admission control, and wireless network power control.

Grid computing is a state-of-the-art parallel computing technology which enables worldwide computers to dynamically share their computing powers and resource to each other. The grid takes advantage of Internet as a universal communication platform to carry messages. Basically, Internet doesn't guarantee loss-free and ordered transmission, hence, the grid should keep the cause and effect of events by itself to ensure the correct ordering of command invocations at the remote hosts. The ordering issue arises when the messages travel across the networks with unpredictable delay. Recent research has studied the security and resource control issues, but failed to address the requirements of transport layer on the grid communication platform. In this paper, we propose the Causal Ordered Grid (COG) architecture and implement it to study the transport performance issues when the grid is built over worldwide networks. The COG provides a novel service model to the applications with time-sensitive and causal-ordered transportation. From our experiments, the design of the grid middleware should use a causal-ordered, time-sensitive transportation rather than TCP. Our research will be beneficial to the improvement of the grid computing and can provide wealthy empirical results for the designer.

In this paper we present a network layer based admission control and simple class based service differentiation model to support QoS in mobile ad hoc network. Our distributed admission control procedure works along with the route finding phase of reactive routing protocols for mobile ad hoc network (AODV, DSR etc). We also propose a simple class based distributed service differentiation system to support QoS once a traffic is admitted by our admission control mechanism. The proposed service differentiation is based on DiffServ model and includes modifications like configuration of each node with edge and core functionality, dynamic selection of edge/core functionality, use of minimal and simple classes. Simulation results show that our system allows seven times more real time traffic in the network than the proposed QoS for AODV model while satisfying the demanded end-to-end delay and providing low jitter.