Class-based delay differentiation model has been recently proposed as a part of relative differentiated services frameworks, and it is shown that the model can provide delay differentiation without admission control and end-to-end resource reservation. In this paper, however, we observe that there can be inconsistent delay differentiation caused by different size of packets. We propose packet size-based delay differentiation model and show that packet size-based queueing is effective to achieve equal delay within a class and provide consistent delay differentiation between classes through simulations. Simulation results also show that the proposed model improves jitter characteristics of CBR flows.

In this paper, we propose an efficient quality-providing scheme to satisfy delay bound and loss ratio requirements for real-time video applications. To utilize network resources more efficiently while meeting service requirements, the network resources are dynamically allocated to each video connection based on the predicted traffic and currently provided quality of service degree. With the proposed bandwidth allocation method, a fair quality of service support in terms of packet loss ratio and maximum packet transfer delay to each video source can be achieved. To avoid possible quality violation by incoming new video connections, we present a connection admission control based on the provided QoS for existing connections and the measured traffic statistics. Simulation results show that our proposed dynamic method is able to provide accurate quality control.

This paper presents the performance modeling, analysis, and simulation of SIP-T (Session Initiation Protocol for Telephones) signaling system in carrier class packet telephony network for NGN (Next Generation Networks). Until recently, fone of the greatest challenges in the migration from existing PSTN (Public Switched Telephone Network) toward NGN is to build a carrier class packet telephony network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new packet telephony technology. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) draft is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of PSTN with carrier class packet telephony network. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP (Voice over IP) calls in carrier class packet telephony network. In this paper, we derive the buffer size, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class packet telephony network focused on toll by-pass or tandem by-pass of PSTN. First, we assume a mathematical model of the M/G/1 queue with non-preemptive priority assignment to represent SIP-T signaling system. Second, we derive the formulas of buffer size, queueing delay, and delay variation for the non-preemptive priority queue by queueing theory respectively. Besides, some numerical examples of buffer size, queueing delay, and delay variation are presented as well. Finally, the theoretical estimates are shown to be in excellent consistence with simulation results.

Future wireless communication systems will provide mobile terminals with high bit-rate transmissions for accessing broadband wired networks. In this paper, we envisage a Time Division Multiple Access - Time Division Duplexing (TDMA-TDD) air interface and we propose a Medium Access Control (MAC) protocol, named Dynamic Scheduling - TDD (DS-TDD), that allows guaranteeing the QoS of different traffic classes and efficiently supports uplink/downlink traffic asymmetries. The DS-TDD performance is theoretically analyzed. Moreover, the DS-TDD protocol is compared with another scheme proposed in the literature. Finally, the impact of packet errors on the DS-TDD performance is evaluated.

This paper presents an efficient scheme for access bandwidth control for VBR (Variable Bit Rate) traffic between radio mobile terminals and their base stations in a WATM (Wireless ATM) network. After introducing the wireless ATM system model, we describe a new algorithm that enables dynamic slot allocation under TDMA/TDD (Time Division Multiple Access/Time Division Duplex) Media Access Control, making use of UPC (Usage Parameter Control) parameters and traffic characteristics. We show more efficient bandwidth utilization with our proposed algorithm, compared to other conventional algorithms. Moreover, we reveal that our algorithm improves cell transmission delays.

In this paper, we propose a new fair queueing algorithm to improve cell delay variation (CDV) for real-time service categories and to make efficient use of system resources for multimedia traffic in high speed ATM networks. The proposed algorithm is called the delay variation-based fair queueing (DVFQ) algorithm, which is based on per-VC queueing to improve CDV and fairness for each VC of real-time services such as CBR and rt-VBR. In DVFQ algorithm, we define two fairness indexes, which indicate the degree of the fairness of CDV at the rate of each VC, and the degree of impartially sharing the bandwidth between the scheduled cells for each VC. The simulation results for both heavily and lightly loaded conditions show that DVFQ algorithm provides better performances in terms of the CDV, the CDV fairness, and the service fairness than those of FCFS for real-time service.

The problem of finding a path with two additive constraints, in particular finding a path that satisfies both the cost and the delay constraints, is called multi-constrained path (MCP) problem in the literature. In this paper, we explore the MCP problem based on the idea of single mixed weight --a mixed weight for each link is first obtained by combining its delay and cost, and then Dijkstra's algorithm is used to find the corresponding shortest path. Given two infeasible paths, it can be theoretically proved that a better path can possibly be found if we choose an appropriate parameter to construct the mixed weight. An approximate algorithm is thus proposed to solve the MCP problem. Theoretical analysis demonstrates that this algorithm can make a correct judgment whether there is a feasible path or not with a very high probability even in the strict case where the delay bound is between the delays of the least delay path and the least cost path, while the cost bound is between the costs of the two paths. On the other hand, the time complexity of this algorithm is very small since it only needs to execute Dijkstra's algorithm a limited number of times. The excellent performance of the proposed algorithm is verified by a large number of experiments on networks of different sizes.

In order for a service discipline to be used for guaranteed service networks at very high speed, its overall implementation must be scalable while it provides as wide a network schedulability region as possible. From this point of view, GPS-based service disciplines provide a narrow network schedulability region while EDF-based disciplines suffer from the implementation complexities of rate-controllers and admission control. Alternatively, although service disciplines based on service-curves can provide a wider network schedulability region than GPS-based and EDF-based disciplines, they may have even worse implementation complexities than EDF-based disciplines. In this paper, we propose to employ a service discipline based on our specific service-curves. We show that our service discipline has comparable implementation complexity to GPS-based disciplines while providing the same wide network schedulability region that EDF-based disciplines can provide. In fact, this service discipline is an SCED service discipline proposed in [14]. However, our specific service-curves provide the SCED service discipline with the same network schedulability region that EDF-based disciplines can provide, O(1) complexity for deadline calculation, and O(N) complexity for admission control where N is the number of sessions.

This paper describes the concept of an adaptive network, that is, a network environment that can rapidly and autonomously adapt its behavior according to network conditions and traffic status. The user interface of the adaptive network can access any resource in the network as a memory-mapped I/O device, as if it were attached to the local bus of the user's PC. This network concept has several benefits. From the application development viewpoint, no network related programming is required, and applications do not have to be modified even if the network topologies and protocols are changed. Network maintenance and upgrading can be done anytime without having to worry about the application users, because the network itself is concealed from the applications. In addition, the reconfigurable hardware technology functions as an autonomous network control through the use of a lower-layer protocol. We developed a testbed that makes heterogeneous resources available to users and used it to demonstrate the feasibility of our concept by implementing and running some applications over it.

Recently, as multimedia and high-speed traffic become more popular on the Internet, the various traffic requiring different qualities of service (QoS) will co-exist. In addition, classified service based on Diff-Serv (Differentiated Service), MPLS (Multi-Protocol Label Switching),etc., have come into wide use. Today's Internet environment requires routers to perform control mechanisms in order to guarantee various QoSs. In this paper, we propose a buffer management scheme for the Internet router that uses class-based priority control. This paper focuses on per-flow queueing, and evaluates the performance of the proposed buffer management scheme. Realization of differentiated services and dissolution of buffer occupation by specific flow is expected by the proposed control.

A quality-of-service based link control scheme to counteract correlated channel errors for wireless multimedia communications is proposed in this paper. Both the medium access (MAC) and data link control (DLC) layers are treated. The performance of the proposed scheme is evaluated using both analysis and simulation. The delay and jitter behaviors are examined for both the constant bit rate (CBR) traffic and variable bit rate (VBR) traffic. The throughput performance is also obtained for the available bit rate (ABR) traffic. Through numerical experiments, the proposed scheme is demonstrated to be not only robust against channel impairments but also capable of providing the desired QoS for wireless multimedia communications.

The IETF Differentiated Services (DiffServ) framework achieves scalability by (1) aggregating traffic flows with coarse grain QoS on the data plane, and (2) allocating network resources with a bandwidth broker (BB) on the control plane. However, there are many issues that need to be addressed under such framework. First, it has been shown that the concatenation of strict priority (SP) scheduler of class-based queues (CBQ) can cause delay jitter unbounded under certain utilization, which is not acceptable to support the premium service (PS). Furthermore, it is not clear how such a DiffServ network can support traffic flows requiring the guaranteed service (GS), which is a desirable feature of the future Internet. This paper presents architecture and mechanisms to support multiple QoS under the DiffServ paradigm. On the data plane, we present a node architecture based on the virtual time reference system (VTRS). The key building block of our node architecture is the core-stateless virtual clock (CSVC) scheduling algorithm, which, in terms of providing delay guarantee, has the same expressive power as a stateful weighted fair queueing (WFQ) scheduler. With the CSVC scheduler as our building block, we design a node architecture that is capable of supporting integrated transport of the GS, the PS, the assured service (AS), and the traditional best effort (BE) service. On the control plane, we present a BB architecture to provide flexible resource allocation and QoS provisioning. Simulation results demonstrate that our architecture and mechanisms can provide scalable and flexible transport of integrated traffic of the GS, the PS, the AS, and the BE services.

In the past, a number of scheduling algorithms that approximate GPS, such as WFQ, have been proposed and have received much attention. This class of algorithms provides per-flow QoS guarantees in terms of the bounded delay and minimum bandwidth guarantee. However, with O(log N) computational cost for each new arrival scheduling, where N is the number of backlogged flows, these algorithms are expensive to implement (e.g., in terms of scalability). Moreover, none of them addresses the issues of delay distribution and jitter. In this paper, we propose a new traffic scheduling discipline called Jitter Control Frame-based Queueing (JCFQ) that provides an upper bound for delay jitter in the case of rate-controlled connections, such as packet video streams and IP telephony, while guaranteeing bounded delay and worst-case fair weighted fairness, such as in the WF2Q algorithm, but with O(1) complexity in selecting the next packet to serve, assuming that the number of flows is fixed. Three different algorithms for slot or service order assignment between flows are proposed: Earliest Jitter Deadline First (EJDF), Rate Monotonic (RM) and Maximum Jitter First (MJF). In these algorithms, delay jitter is formulated into the virtual finish time calculation. We compare the fairness, delay and jitter performance of the JCFQ with that of the MJF algorithm with WF2Q via simulation. The results show that with proper choice of the slot size, JCFQ can achieve better flow isolation in delay distribution than can WF2Q.

In this paper we study the scalability issue in the design of a centralized bandwidth broker model for dynamic control and management of QoS provisioning. We propose and develop a path-oriented, quota-based dynamic bandwidth allocation mechanism for efficient admission control operations under the centralized bandwidth broker model. We demonstrate that this dynamic bandwidth allocation mechanism can significantly reduce the overall number of QoS state accesses/updates, thereby increasing the overall call processing capability of the bandwidth broker. Based on the proposed dynamic bandwidth allocation mechanism, we also extend the centralized architecture with a single bandwidth broker to a hierarchically distributed architecture with multiple bandwidth brokers to further improve its scalability. Our study demonstrates that the bandwidth broker architecture can be designed in such a manner that it scales with the increase in the network capacity.

Emerging multimedia technologies introduce the prevalent multicast transmission, and the multicast tree is determined using the time-invariant network parameters. This paper addresses the time-varying multicast tree problem and presents path selection heuristics for multicast routing to determine an alternative path for real-time applications. A network is partitioned into the optimal region, the disjoint region, and the edge cutset if a branch of the multicast tree meets the un-guaranteed QoS condition. The path selection heuristics operate during the multicast session phase to efficiently select an alternative routing path containing an edge in the edge cutset to connect the multicast tree again. The source-based heuristics PS-SPT finds the path for minimal source-to-destination delay and the sharing-based heuristics PS-DDMC for minimal total cost. These path selection heuristics can efficiently provide solutions to keep the multicast transmission reliable. Simulation results also show that the proposed heuristics can provide effective good solutions for real-time multicast transmission. PS-SPT can select a path with optimal source-to-destination delay and PS-DDMC can select a path with optimal total cost.

This paper suggests the way to perform the handover by predicting the movement route of the mobile terminal by considering the movement pattern of the user. By considering the fact that the most users has the constant movement pattern, the channels needed for the handover can be reserved, and the required quality of service (QoS) is maintained during handover. The suggested algorithm makes the channel allocation schemes more efficient.

We have developed a new scheme to provide Diffserv-based QoS over ATM access networks. Well-known Diffserv over ATM scheme requires some extension for conventional routers with ATM interfaces. The routers must map their Diffserv classes of services into ATM QoS classes and forward IP packets into prioritized VCs based on DSCP (DiffServ Code Point). The purpose of this work is to provide Diffserv-based QoS over ATM network using conventional IP over ATM interfaces on routers. We propose DSCP snooping at ATM edge nodes, which differentiates services over a single VC between two IP domains. A prototype circuit was used to evaluate this scheme.

The differentiated services (diffserv) architecture has been proposed for implementing scalable service differentiation in the Internet. Expedited forwarding and assured forwarding have been standardized as Per-Hop Behaviors (PHB) in diffserv. Assured forwarding can be utilized to realize the service, which provides each user with a minimum guaranteed rate and a fair share of the residual bandwidth. We call it guaranteed rate (GR) service. With GR service, each packet for flow i is marked in or out based on comparison between the sending rate and the minimum guaranteed rate. When congestion occurs in networks, out packets are dropped more aggressively than in packets. Recently, several fair queuing schemes have been proposed for core stateless networks. They can achieve fairer bandwidth allocation than random early detection (RED). However, there have not been any studies that consider in/out bit usage to support GR service. This paper proposes how to extend the schemes that have been proposed for core stateless networks to allow the support of in/out bit usage. We present the performance of one of the extended schemes and compare the scheme to RED with in/out bit (RIO) in terms of fair bandwidth allocation.

One drawback of Integrated service architecture is the scaling problem. Therefore, flow aggregation is an important solution for supporting quality of service in large-scale network. In advance reservation, priori information of advance-reserved requests can be used for flow aggregation before their initiation time. However, an impolitic aggregation can lead to violate admission control. In this paper, we propose an effective algorithm to aggregate advance-reserved requests with guaranteed delay. The proposed algorithm not only can reduce the amount of state in core network but also minimize the bandwidth consumption. The simulation result indicates that the state in the core network can be reduced as low as 17.3% even in the worst case.

Call admission control (CAC) is becoming vital for multimedia services in the ability of wireless/mobile networks to guarantee Quality of Service (QoS) partially due to the network's limited capacity. In this paper, we propose an optimal call admission control scheme with bandwidth reallocation algorithm (multi-class-CAC-BRA) for multi-classes of adaptive multimedia services in wireless/mobile networks. The multi-class-CAC-BRA approach optimizes revenue for service providers and satisfies QoS requirements for service users. The proposed approach adopts semi-Markov Decision Process to model both call admission control and bandwidth reallocation algorithm. In other words, whenever decisions are made, decisions are made for both call admission control and bandwidth reallocation. Since the non-adaptive multimedia traffic is a special case of the adaptive multimedia traffic, the non-adaptive optimal CAC scheme is a special case of our optimal multi-class-CAC-BRA scheme. Furthermore, the Interior-point Method in linear programming is used to solve the optimal decision problem. Simulation results reveal that the proposed multi-class-CAC-BRA scheme adapts itself well to adaptive multi-class multimedia traffic, achieves optimal revenue, and satisfies QoS requirements that are the upper bounds of handoff dropping probabilities. Our approach solves the optimal adaptive multimedia CAC problem. We believe that this work has both theoretical and practical significance.