Supporting quality of service (QoS) capabilities for multi-media applications is one of the major issues in medium access control (MAC) research. In distributed contention-based MAC algorithms, it is a challenging task to support the desired QoS because of the inherent random access characteristics. In this paper, we propose an efficient prioritized fast collision resolution algorithm. The MAC protocol with this new algorithm attempts to provide significantly high throughput performance for data services and support QoS for real-time services. We incorporate the priority algorithm based on service differentiations with the fast collision resolution algorithm, and show that this algorithm can simultaneously achieve high throughput and good QoS support for real-time and data services.

In this letter, we analyze blocking probabilities for prioritized multi-classes in optical burst switching (OBS) networks. The blocking probability of each traffic class can be analytically evaluated by means of class aggregation and iteration method. The analytic results are validated with results garnered from simulation tests.

Call Admission Control (CAC) is a very important issue in CDMA systems to guarantee a required quality of service (QoS) and to increase system capacity. In this paper, we proposed and analyzed the CAC scheme using multiple criterions (MCAC), which can provide a quicker processing time and better performance. One is based on the number of active users with the minimum/maximum threshold by considering the spillover ratio, and the other is based on the signal to interference ratio (SIR). If active users are lower/higher than the minimum/maximum number of users threshold (N_min )/(N_max ), we accept/reject the new call without any other considerations based on the first criterion. And if the number of active users is between the N_min and N_max, we consider the current SIR to guarantee QoS based on the second criterion. Then the system accepts the new call when the SIR satisfies the system requirements, otherwise, the call is rejected. The multiple criterions scheme is investigated and its performance is compared with the number of user based CAC and power based CAC.

The issue of guaranteeing Quality of Services (QoS) in a network has emerged in recent years. The Proportional Delay Differentiated Model has been presented to provide the predictable and controllable queueing delay differentiation for different classes of connections. However, most related works have focused on providing this model for a wired network. This study proposes a novel scheduler to provide proportional delay differentiation in a wireless network that includes a multi-state link. This scheduler, Look-ahead Waiting-Time Priority (LWTP), offers proportional delay differentiation and a low queueing delay, by adapting to the location-dependent capacity of the wireless link and solving the head-of-line (HOL) blocking problem. The simulation results demonstrate that the LWTP scheduler actually achieves delay ratios much closer to the target delay proportion between classes and yields smaller queueing delays than past schedulers.

A principal attraction of ATM networks, in both wired and wireless realizations, is that the key quality of service (QoS) parameters of every call, including end-to-end delay, jitter, and loss are guaranteed by the network when appropriate cell-level traffic controls are imposed at the user network interface (UNI) on a per call basis, utilizing the peak cell rate (PCR) and the sustainable cell rate (SCR) values for the multimedia--voice, video, and data, traffic sources. There are three practical difficulties with these guarantees. First, while PCR and SCR values are, in general, difficult to obtain for traffic sources, the typical user-provided parameter is a combination of the PCR, SCR, and the maximum burstiness over the entire duration of the traffic. Second, the difficulty in accurately defining PCR arises from the requirement that the smallest time interval must be specified over which the PCR is computed which, in the limit, will approach zero or the network's resolution of time. Third, the literature does not contain any reference to a scientific principle underlying these guarantees. Under these circumstances, the issue of providing QoS guarantees in the real world, through traffic controls applied on a per call basis, is rendered uncertain. This paper adopts a radically different, high level approach to the issue of QoS guarantees. It aims at uncovering through systematic experimentation a relationship, if any exists, between the key high level user traffic characteristics and the resulting QoS measures in a realistic operational environment. It may be observed that while each user is solely interested in the QoS of his/her own traffic, the network provider cares for two factors: (1) Maximize the link utilization in the network since links constitute a significant investment, and (2) ensure the QoS guarantees for every user traffic, thereby maintaining customer satisfaction. Based on the observations, this paper proposes a two-phase strategy. Under the first phase, the average "link utilization" computed over all the links in a network is maintained within a range, specified by the underlying network provider, through high level call admission control, i.e. by limiting the volume of the incident traffic on the network, at any time. The second phase is based on the hypothesis that the number of traffic sources, their nature--audio, video, or data, and the bandwidth distribution of the source traffic, admitted subject to a specific chosen value of "link utilization" in the network, will exert a unique influence on the cumulative delay distribution at the buffers of the representative nodes and, hence, on the QoS guarantees of each call. The underlying thinking is as follows. The cumulative buffer delay distribution, at any given node and at any time instant, will clearly reflect the cumulative effect of the traffic distributions of the multiple connections that are currently active on the input links. Any bounds imposed on the cumulative buffer delay distribution at the nodes of the network will also dominate the QoS bounds of each of the constituent user traffic. Thus, for each individual traffic source, the buffer delay distributions at the nodes of the network, obtained for different traffic distributions, may serve as its QoS measure. If the hypothesis is proven true, in essence, the number of traffic sources and their bandwidth distribution will serve asa practically realizable high level traffic control in providing realistic QoS guarantees for every call. To verify the correctness of the hypothesis, an experiment is designed that consists of a representative ATM network, traffic sources that are characterized through representative and realistic user-provided parameters, and a given set of input traffic volumes appropriate for a network provider approved link utilization measure. The key source traffic parameters include the number of sources that are incident on the network and the constituent links at any given time, the bandwidth requirement of the sources, and their nature. For each call, the constituent cells are generated stochastically, utilizing the typical user-provided parameter as an estimate of the bandwidth requirement. Extensive simulations reveal that, for a given link utilization level held uniform throughout the network, while the QoS metrics--end-to-end cell delay, jitter, and loss, are superior in the presence of many calls each with low bandwidth requirement, they are significantly worse when the network carries fewer calls of very high bandwidths. The findings demonstrate the feasibility of guaranteeing QoS for each and every call through high level traffic controls. As for practicality, call durations are relatively long, ranging from ms to even minutes, thereby enabling network management to exercise realistic controls over them, even in a geographically widely dispersed ATM network. In contrast, current traffic controls that act on ATM cells at the UNI face formidable challenge from high bandwidth traffic where cell lifetimes may be extremely short, in the range of µs. The findings also underscore two additional important contributions of this paper. First, the network provider may collect data on the high level user traffic characteristics, compute the corresponding average link utilization in the network, and measure the cumulative buffer delay distributions at the nodes, in an operational network. The provider may then determine, based on all relevant criteria, a range of input and system parameters over which the network may be permitted to operate, the intersection of all of which may yield a realistic network operating point (NOP). During subsequent operation of the network, the network provider may guide and maintain the network at a desired NOP by exercising control over the input and system parameters including link utilization, call admittance based on the requested bandwidth, etc. Second, the finding constitutes a vulnerability of ATM networks which a perpetrator may exploit to launch a performance attack.

This paper considers the problem of providing relative service differentiation in IEEE 802.11 Wireless LAN by using different Medium Access Control (MAC) parameters for different service classes. We present an analytical model which predicts the saturation throughput of IEEE 802.11 Distributed Coordination Function with multiple classes of service. This model allows us to show that relative service differentiation can be achieved by varying the initial contention window alone. In this case, the saturation throughput of a station can be shown to be approximately inversely proportional to the initial contention window size being used by that station. The simulation results validate our analytical model.

Future mobile communication systems are expected to support multimedia applications (audio phone, video on demand, video conference, file transfer, etc.). Multimedia applications make a great demand for bandwidth and impose stringent quality of service requirements on the mobile wireless networks. In order to provide mobile hosts with high quality of service in the next generation mobile multimedia wireless networks, efficient and better bandwidth reservation schemes must be developed. A novel traffic-based bandwidth reservation scheme is proposed in this paper as a solution to support quality of service guarantees in the mobile multimedia wireless networks. Based on the existing network conditions, the proposed scheme makes an adaptive decision for bandwidth reservation and call admission by employing fuzzy inference mechanism, timing based reservation strategy, and round-borrowing strategy in each base station. The amount of reserved bandwidth for each base station is dynamically adjusted, according to the on-line traffic information of each base station. We use the dynamically adaptive approach to reduce the connection-blocking probability and connection-dropping probability, while increasing the bandwidth utilization for quality of service sensitive mobile multimedia wireless networks. Simulation results show that our traffic-based bandwidth reservation scheme outperforms the previously known schemes in terms of connection-blocking probability, connection-dropping probability, and bandwidth utilization.

Future IP networks will provide multi-class-services that have multiple levels of Quality of Services (QoS) at different prices. One of the issues for the network service provider (NSP) will be how to profit by providing them. This paper proposes a scheme that maximizes the profit of the NSP by controlling the service-list under the constraint of the available network resources. We introduce a model in which the users' selection from among the multiple classes is influenced not just by the price and QoS of one class, but the prices and QoS levels of all classes. In short, the user's selection involves a balance between the price and QoS levels of all classes. To model the users' class choice, we adopt discrete choice analysis; it can estimate the model parameters such that the model fits actual choice data. This paper proposes a functional framework that consists of User Choice Model Function, Original Demand Forecast Function, and Service-list Determination Function. The proposed model has the advantage of following actual changes adaptively. Effectiveness of the proposed scheme is evaluated by computer simulation for a multiple class service; even if the real parameters are changed, the proposal can follow the change and provide the optimal service-list that maximizes profit adaptively.

Quality of Service (QoS) provisioning is a new but challenging research area in the field of Mobile Ad hoc Network (MANET) to support multimedia data communication. However, the existing QoS routing protocols in ad hoc network did not consider a major aspect of wireless environment, i.e., mutual interference. Interference between nodes belonging to two or more routes within the proximity of one another causes Route Coupling. This can be avoided by using zone-disjoint routes. Two routes are said to be zone disjoint if data communication over one path does not interfere with the data communication along the other path. In this paper, we have proposed a scheme for supporting priority-based QoS in MANET by classifying the traffic flows in the network into different priority classes and giving different treatment to the flows belonging to different classes during routing so that the high priority flows will achieve best possible throughput. Our objective is to reduce the effect of coupling between routes used by high and low priority traffic by reserving zone of communication. The part of the network, used for high priority data communication, i.e, high priority zone, will be avoided by low priority data through the selection of a different route that is maximally zone-disjoint with respect to high priority zones and which consequently allows contention-free transmission of high priority traffic. The suggested protocol in our paper selects shortest path for high priority traffic and diverse routes for low priority traffic that will minimally interfere with high priority flows, thus reducing the effect of coupling between high and low priority routes. This adaptive, priority-based routing protocol is implemented on Qualnet Simulator using directional antenna to prove the effectiveness of our proposal. The use of directional antenna in our protocol largely reduces the probability of radio interference between communicating hosts compared to omni-directional antenna and improves the overall utilization of the wireless medium in the context of ad hoc wireless network through Space Division Multiple Access (SDMA).

Multiuser diversity, identified by recent information theoretic results, is a form of diversity inherent in a wireless network. The diversity gain is obtained from independent time-varying fading channels across different users. The main practical issue in multiuser diversity is lack of Quality of Service (QoS) guarantees. This study proposes a wireless scheduling algorithm named MUDSEQ for downlink channels exploiting multiuser diversity under explicit QoS constraints. The numerical results demonstrate that the novel algorithm can yield non-negligible diversity gain even under tight QoS constraints and little scattering or slow fading environments. Additionally, a system framework for dynamic resource allocation based on the proposed algorithm is developed.

The issue of scalable Differentiated Services (DiffServ) admission control now is still an open research problem. We propose a new admission control model that can not only provide coarse grain Quality of Services (QoS), but also guarantee end-to-end QoS for assured service without per-flow state management at core routers within DiffServ domain. Associated with flow aggregation model, a hybrid signaling protocol is proposed to select the route satisfying the end-to-end QoS requirements. Simulation result shows that the proposed model can accurately manage resource, leading to much better performance when compared to other schemes.

Recently, wireless networks have become a major sector in the telecommunication industry. More and more applications seek to become wireless. However, a major obstacle in adapting wired applications to wireless is the quality of service problem. Although the wireless bandwidth is improving at a fast pace, it still is not enough for modern multimedia applications. Even if we solve the bandwidth problem, the mobility of users also poses challenges for QoS provision. If the user moves randomly, how and where can resources be reserved in advance for roaming users to move smoothly and seamlessly? In this paper, we propose a method for predictive resource reservation in wireless networks. Resources reserved but not used will seriously affect the system performance. Therefore, we also have mechanisms to release the reserved resources when it is not used within a time limit and allow resources to be used temporarily by another mobile user. We compare the performance of our method with those of fixed allocation scheme and shadow cluster scheme. The results indicate its effectiveness and feasibility.

Recently routing protocols for QoS aware multicast are actively studied, but there are few studies focusing on the scalability of link state advertisement when the available bandwidth of a link is updated along with the QoS aware multicast tree construction. This paper proposes a new QoS aware multicast routing protocol that is scalable in terms of the link state advertisement exchange. Our protocol has the following features; (1) A multicast network is divided into domains, and the advertisement of information on links within a domain is limited within the domain. (2) Among the border multicast routers, only the link state information of inter-domain links is advertised. As a result, the number of link state advertisement messages will be drastically reduced. (3) When a multicast tree spreads over multiple domains, the tree construction needs to be performed without information on links in other domains, and it is possible that the construction may fail. In order to cope with this problem, the crank back mechanism of a tree construction is introduced. This paper describes the detailed procedures and the message formats of our protocol. It also describes the evaluation of the number of exchanged link state advertisement messages and shows that our protocol can reduce the number comparing with the conventional protocols.

This paper presents the performance modeling application of SIP-T (Session Initiation Protocol for Telephones) signaling system based on two-class priority queueing process in carrier class VoIP (Voice over IP) network. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of existing PSTN (Public Switched Telephone Network) with carrier class VoIP network. One of the greatest challenges in the migration from PSTN toward NGN (Next Generation Networks) is to build a carrier class VoIP network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new VoIP technology. 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 calls in carrier class VoIP network. This paper presents the two class priority queueing model, performance analysis, and simulation of SIP-T signaling system in carrier class VoIP network focused on toll by-pass or tandem by-pass of PSTN. In this paper, we analyze the average queueing length, 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 VoIP network. A mathematical model of the M/G/1 queue with two-class non-preemptive priority assignment is proposed to represent SIP-T signaling system. Then, the formulae of average queueing length, queueing delay, and delay variation for the non-preemptive priority queue are expressed respectively. Several significant numerical examples of average queueing length, queueing delay, and delay variation are presented as well. Finally, the two-class priority queueing model and performance analysis of SIP-T signaling system are shown the accuracy and robustness after the comparison between theoretical estimates and simulation results.

This paper presents measurement and analysis of various networks and applications using a high-speed IP meter. The authors have developed a high-speed IP meter with a GPS timestamp component, which enables precise measurement of packet delay and jitter in various networks. Measurement of the following networks was performed, 1) Measurement of traffic in a commercial IPv6 access service on ADSL, as a typical broadband access service network. 2) Measurement of traffic in the 54th IETF meeting in Yokohama, as a typical high-speed Internet backbone network. This paper reports the characteristics identified in these networks, e.g. asymmetricities of one-way packet delay over an ADSL access network and the difference in TCP/UDP packet delay over a high-speed backbone network. It also presents the analysis results of some multimedia applications in the Internet, and discusses the quality of service on Internet access service networks.

In this paper, we propose a new predictive location management strategy that reduces the update cost while restricting the paging load optimized for mobiles roaming with traceable patterns. Enhanced with directional predictive capabilities offered by Kalman filtering, new update boundaries are assigned to better reflect the movement patterns of individual mobiles upon location registration. Thus, while complying with the required delay constraints, QoS measures such as throughput will not need to be sacrificed as a result of increasing update threshold. The contribution of this paper is two-fold: (1) to propose a distribution model that is capable of describing a wide range of movement patterns with varying correlation between traveling directions and (2) to show the capabilities (in terms of reliable performances) of the Kalman filter in predicting future movement patterns. Simulation results have successfully demonstrated the ability of the Kalman filter in assigning update boundaries capable of reflecting a mobile's roaming characteristics. The performance gains achieved mainly through a significant reduction in the number of updates indicate its potential for promoting better bandwidth conservation.

The emerging multimedia applications for future mobile communication systems typically require highly diversified Quality of Service (QoS). However, due to the time and location dependent fluctuating nature of radio resources in the radio link, it is very difficult to maintain a constant level of QoS with the current end-to-end QoS control only. Therefore, wireless-aware QoS is the key issue for achieving better end-end QoS. In this paper, a new wireless QoS scheme for a joint CDMA/NC-PRMA cellular system are proposed considering QoS prioritization mechanism, users' diversified requirements and the harmonization with IP-QoS. Two wireless QoS-aware resource allocation algorithms are proposed to support QoS prioritization while achieving high radio resource utilization. By introducing a set of new QoS resource request parameters (minimum, average and maximum requirements), the algorithms can assign radio resource in a more flexible way than the conventional fixed resource allocation. Computer simulations indicate that the proposed QoS algorithms exhibit superior performance with respect to packet dropping probability for realtime application users, and improve transmission rate for non-realtime application users, which evince the effectiveness of the proposed wireless QoS algorithms.

End-to-end delay and loss measurement is an efficient way for a host to examine the network performance. Unnoticed clock errors that influence the accuracy of the timestamp may result in fatal system errors. In this paper, we discuss the characteristics and defects of the existing clock distortion adjustment algorithms. Those algorithms are not applicable to process a long-term delay trace, which contains periodical NTP clock adjustment. Therefore, we propose a relatively robust algorithm to resolve the problem. The algorithm employs window function to partition the long-term trace into short segments, improves the precision of the estimation of the time and amount of NTP clock adjustment To evaluate the performance of our proposed algorithm, we practice it in adjusting the clock distortion of the real delay traces collected from Internet. The results indicate that our proposed algorithm has excellent effect on the removal of the clock distortion from the long-term delay traces.

There is a growing demand that mobile networks should provide quality-of-service (QoS) to mobile users since portable devices become popular and more and more applications require real-time services. Providing QoS to mobile hosts is very difficult due to mobility of hosts. The resource ReSerVation Protocol (RSVP) establishes and maintains a reservation state to ensure a given QoS level between the sender and receiver. However, RSVP is designed for fixed networks and thus it is inadequate in wireless mobile networking environments. In this paper, we propose a resource reservation protocol for mobile hosts in mobile networks. The proposed protocol extends the RSVP by introducing RSVP agents in local networks to manage the reservations. The proposed protocol reduces packet delay, bandwidth overhead, and the number of RSVP messages to maintain reservation states. We examined the performance of the proposed protocol by simulation and we got an improved performance over the existing protocols.

As multimedia and high-speed traffic become more popular on the Internet, the various traffic requiring different qualities of service (QoS) must co-exist. In addition, classified services 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 smart buffer management scheme for the Internet router that uses hierarchical priority control with port class and flow level. Furthermore, since the proposed scheme must operate at very high speed, we first propose several design policy for high speed operation and the hardware implementation is performed in VHDL code. Implementation results show that the proposed scheme can scale with high-speed link, achieving the maximum rate of 4.0 Gbps by using the 3.5 µm CMOS technology.