IEEE 802.15.4 is a new standard, uniquely designed for low rate wireless personal area networks (LR-WPANs). It targets ultra-low complexity, cost, and power, for low-data-rate wireless connectivity. However, one of the main problems of this new standard is its insufficient, and inefficient, media access control (MAC) for priority data. This paper introduces an extended contention access period (XCAP) concept for priority packets, also an traffic adaptive contention differentiation utilizing the XCAP (TACDX). The TACDX determines appropriate transmission policy alternatively according to the traffic conditions and type of packet. TACDX achieves not only enhanced transmission for priority packets but it also has a high energy efficiency for the overall network. The proposed TACDX is verified with simulations to measure the performances.

Mobility-aware quality of service (QoS) signaling is crucial to provide seamless multimedia services in the ambient environment where mobile nodes may move frequently between different wireless access networks. The mobility of an IP-based node in ambient networks affects routing paths, and as a result, can have a significant impact on the operation and state management of QoS signaling protocols. In this paper, we first analyze the impact of mobility on QoS signaling protocols and how the protocols operate in mobility scenarios. We then propose an efficient mobility-aware QoS signaling protocol which can operate adaptively in ambient networks. The key features of the protocol include the fast discovery of a crossover node where the old and new paths converge or diverge due to handover and the localized state management for seamless services. Our analytical and simulation/experimental results show that the proposed/implemented protocol works better than existing protocols in the IP-based mobile environment.

In OLSR (Optimized Link State Routing Protocol), the multipoint relay mechanism has been introduced to minimize routing overhead for flooding control traffic. In order to achieve as low a routing overhead as possible, the selection of MPRs (multipoint relays) is designed to limit the overall number of such relays in the network. OLSR provides the shortest paths in terms of hops; however, it does not take into account the QoS (quality of service) requirements. Therefore, QOLSR (Quality OLSR), which adds a QoS extension to the OLSR, has been proposed. Although QOLSR provides the best QoS path, its selection process does not consider the number of MPRs, which causes an increase in the routing overhead. In this paper, we focus on the selection process of MPRs in link state QoS routing protocol. We propose three heuristics for high-efficiency selections: MIMS, MQES, and MCIS. The basic approach of these selections is to determine a smaller set of MPRs that provide better QoS paths between any two nodes. The main objective in doing so is to maximize the QoS effect while limiting the routing overhead. In addition, we evaluate the performance of the routing protocols with the proposed selections by simulation. The results indicate that MIMS and MCIS achieve high-efficiency selection; as compared to QOLSR, MIMS reduces the maintenance cost by 30%, while the throughput of the resultant path is decreased by 13%, and MCIS reduces the cost by 21% without any decrease in the throughput.

We propose two vertical handoff schemes for cellular network and wireless local area network (WLAN) integration: integrated service-based handoff (ISH) and integrated service-based handoff with queue capabilities (ISHQ). Compared with existing handoff schemes in integrated cellular/WLAN networks, the proposed schemes consider a more comprehensive set of system characteristics such as different features of voice and data services, dynamic information about the admitted calls, user mobility and vertical handoffs in two directions. The code division multiple access (CDMA) cellular network and IEEE 802.11e WLAN are taken into account in the proposed schemes. We model the integrated networks by using multi-dimensional Markov chains and the major performance measures are derived for voice and data services. The important system parameters such as thresholds to prioritize handoff voice calls and queue sizes are optimized. Numerical results demonstrate that the proposed ISHQ scheme can maximize the utilization of overall bandwidth resources with the best quality of service (QoS) provisioning for voice and data services.

Channel scanning is an important aspect of seamless handovers since it is required in order to find a target point of attachment (PoA). However, channel scanning in single radio devices may cause severe service disruptions with the current PoA so that the provided QoS will be further degraded during a handover. In this letter, we propose a dynamic channel scanning algorithm that supports QoS. Simulation results show that the proposed mechanism reduces the service disruptions and provides the desired QoS to users during the scanning period.

In this paper, we analyze the extended real-time Polling Service (ertPS) algorithm in IEEE 802.16e systems, which is designed to support Voice-over-Internet-Protocol (VoIP) services with data packets of various sizes and silence suppression. The analysis uses a two-dimensional Markov Chain, where the grant size and the voice packet state are considered, and an approximation formula for the total throughput in the ertPS algorithm is derived. Next, to improve the performance of the ertPS algorithm, we propose an enhanced uplink resource allocation algorithm, called the e 2rtPS algorithm, for VoIP services in IEEE 802.16e systems. The e 2rtPS algorithm considers the queue status information and tries to alleviate the queue congestion as soon as possible by using remaining network resources. Numerical results are provided to show the accuracy of the approximation analysis for the ertPS algorithm and to verify the effectiveness of the e 2rtPS algorithm.

A call admission control (CAC) scheme based on statistical information is proposed, called the statistical CAC scheme. A conventional scheme needs to manage session information for each link to update the residual bandwidth of a network in real time. This scheme has a scalability problem in terms of network size. The statistical CAC rejects session setup requests in accordance to a pre-computed ratio, called the rejection ratio. The rejection ratio is computed by using statistical information about the bandwidth requested for each link so that the congestion probability is less than an upper bound specified by a network operator. The statistical CAC is more scalable in terms of network size than the conventional scheme because it does not need to keep accommodated session state information. Numerical results show that the statistical CAC, even without exact session state information, only slightly degrades network utilization compared with the conventional scheme.

Slotted wireless ad hoc networks are drawing more and more attention because of their advantage of QoS (Quality of Service) support for multimedia applications owing to their collision-free packet transmission. Time slot assignment is an unavoidable and important problem in such networks. The existing time slot assignment methods have in general a drawback of limited available bandwidth due to their local assignment optimization without the consideration of directions of the radio wave transmission of wireless links along the routes in such networks. A new time slot assignment is proposed in this paper in order to overcome this drawback. The proposed assignment is different from the existing methods in the following aspects: a) consideration of link directions during time slot assignment; b) largest bandwidth to be achieved; c) feasibility in resource limited ad hoc networks because of its fast assignment. Moreover, the effectiveness of the proposal is confirmed by some simulation results.

MPLS-based path technology shows promise as a means of realizing reliable IP networks. Real-time services such as VoIP and video-conference supplied through a multi-domain MPLS network must be able to guarantee end-to-end QoS of the inter-domain paths. Thus, it is important to allocate an appropriate QoS class to the inter-domain paths in each domain traversed by the inter-domain paths. Because each domain has its own policy for QoS class allocation, it is necessary to adaptively allocate the optimum QoS class based on estimation of the QoS class allocation policies in other domains. This paper proposes two kinds of adaptive QoS class allocation schemes, assuming that the arriving inter-domain path requests include the number of downstream domains traversed by the inter-domain paths and the remaining QoS value toward the destination nodes. First, a measurement-based scheme, based on measurement of the loss rates of inter-domain paths in the downstream domains, is proposed. This scheme estimates the QoS class allocation policies in the downstream domains, using the measured loss rates of path requests. Second, a state-dependent type scheme, based on measurement of the arrival rates of path requests in addition to the loss rates of paths in the downstream domains, is also proposed. This scheme allows an appropriate QoS class to be allocated according to the domain state. This paper proposes an application of the Markov decision theory to the modeling of state-dependent type scheme. The performances of the proposed schemes are evaluated and compared with those of the other less complicated non-adaptive schemes using a computer simulation. The results of the comparison reveal that the proposed schemes can adaptively increase the number of inter-domain paths accommodated in the considered domain, even when the QoS class allocation policies change in the other domains and the arrival pattern of path requests varies in the considered domain.

Jitter is the variation of latencies, when real-time Intellectual Properties (IPs) are accessing data from the data storages. It is a critical factor for such IPs from the Quality-of-Service (QoS) perspective. Jitter of a real-time IP can be measured by how frequently it experiences the underflows and overflows from its data queue in read mode and write mode, respectively. Such failures critically depend on the bus arbitration scheme which determines the bus acquisition order of IPs. The proposed idea allows IPs to inform the bus arbiter of the status of their data buffers when they assert bus requests. Such information helps the bus arbiter to determine the bus acquisition order while greatly reducing the jitter. The experimental results show that our method effectively eliminates the overflows and underflows of real-time IPs by dynamically preempting the jitter-critical bus requests.

QoS (Quality of Service) control in WLAN (IEEE802.11 Wireless LAN) is becoming increasingly important because WLAN is widely deployed as an access network and also plays a key role in providing seamless QoS communication between wired networks and wireless terminals. Although previous research has attempted to increase total throughput or available bandwidth in WLAN, few studies have treated individual TCP/UDP flow QoS. EDCA in IEEE802.11e might provide prioritized QoS functions that would partially address this problem. However, in uplink flow, which is defined as data moving from a terminal toward an Access Point, EDCA has limitations. These manifest themselves both across classes and in differentiated QoS control between terminals in the same class. Furthermore, 802.11e requires modification of terminals as well as other alterations proposed by other researchers. Instead of 802.11e or other modifications of 802.11, we propose an approach to controlling QoS that requires no terminal modifications or installation of additional software/hardware. The proposed idea is MAC-frame Receiving-Opportunity Control (ROC), in which a MAC (Media Access Control) frame is completely delivered only if it has sufficiently high priority; otherwise either the MAC frame is discarded or an Acknowledge (ACK) to the frame is not sent. The frame that was discarded is forced to accept a back-off waiting time for retransmission, consistent with 802.11DCF. This results in QoS degradation for low priority flows and eventually results in QoS improvements for high priority flows. Performance evaluation shows that the ROC causes some performance degradation in total WLAN throughput but can achieve not only QoS priority control but also arbitrary throughput performance. In particular, the ROC (in the MAC layer) can also permit different throughputs for high priority and low priority flows, conditioned on control processes in other layers. These may include rate adaptation (in the MAC layer) and TCP congestion control (in the TCP layer).

When an IP Multicast network is constructed on a switch-based network, many IP Multicast packet broadcasts are generated, and these broadcasts cause trouble for all of the other kinds of communication. To solve this problem, implementing IGMP Snooping on various switches has been proposed. However, some switches have insufficient IP Multicast packet-handling capability. This problem is also mentioned in RFC4541. In this paper, we propose the IGMP Snooping Activator (ISA) mechanism as a way to solve the IP Multicast packet-handling problem. The ISA transmits dummy IGMP Queries to maintain the IP Multicast network, and it joins the flooding IP Multicast group to activate IGMP Snooping in switches that are unable to handle IP Multicast packets. The experimental evaluation shows the effectiveness of our proposed method: the network load decreases because of the method's effective restraint of IP Multicast packet flooding.

Orthogonal Frequency Division Multiple Access (OFDMA) is the technique for the next generation wireless networks, whose enhanced capacity is to serve a combination of traffic with diverse QoS requirements. To realize this, the resource allocation scheme has to be carefully designed so that the instantaneous channel condition, QoS provision, and the network utilization are integrated. In this paper, we propose the resource allocation scheme for downlink traffic of 2 classes; guaranteed and non-guaranteed, having different traffic contracts. We provide guaranteed throughput for the guaranteed class by considering the cost incurred from serving this class. Then, we formulate the assignment problem with the objective of minimizing this cost. For the non-guaranteed class, we aim to maximize network utilization and to maintain throughput fairness, by employing Proportional Fairness (PF) utility function and emphasizing on the portion of network resource that the user received and the individual user's queue length. We use a heuristic approach to schedule users' data into the downlink subframe by exploiting multi-user multi-channel diversity to utilize system's bandwidth efficiently. Intensive simulation shows that our scheme differentiates classes of traffic and provides satisfied throughput, lower packet drop rate, and lower queuing delay to the guaranteed class, comparing with those of the non-guaranteed class. Furthermore, the results also show that the scheme is fair to users in the same class in both throughput and service time.

We propose Adaptive Resource Allocation for the Partial Block MC-CDMA (ARA-PB/MC-CDMA) system. The ARA-PB/MC-CDMA system aims to improve total throughput performance and frequency efficiency across various channel conditions. It adaptively changes the number of blocks to improve the throughput performance and frequency efficiency according to the Signal to Interference Ratio (SIR). Therefore, the proposed system supports various Quality of Service (QoS) requirements for various SIR values.

The basic playout scheme (BAS) is designed not to take into account network impairment information during silence periods. We propose a jitter-robust playout mechanism (RST), which uses silence description (SID) packets. The lateness loss percentages are compared between the BAS and the RST algorithms. We report that the accuracy of the playout schedule calculation in the BAS is getting worse as the previous silence interval increases and our proposed RST algorithm is more effective in removing high jitter than the BAS. Under high jitter Internet conditions, the accuracy of the estimates and therefore the resulting of VoIP playout quality can be significantly improved by using the SID packets in the playout schedule recalculation.

A feature of services provided in a Next Generation Network (NGN) is that the end-to-end quality is guaranteed. This is quite a challenging issue, given the considerable fluctuation in network conditions within a Fixed Mobile Convergence (FMC) network. Therefore, a novel approach, whereby a network node and a mobile terminal such as a cellular phone cooperate with each other to control service quality is essential. In order to achieve such cooperation, the mobile terminal needs to become more intelligent so it can estimate the service quality, including the user's perceptual quality, and notify the measurement result to the network node. Subsequently, the network node implements some kind of service control function, such as a resource and admission control function, based on the notification from the mobile terminal. In this paper, the role of the mobile terminal in such collaborative system is focused on. As a part of a QoS/QoE measurement system, we describe an objective speech quality assessment with payload discrimination of lost packets to measure the user's perceptual quality of VoIP. The proposed assessment is so simple that it can be implemented on a cellular phone. We therefore did this as part of the QoS/QoE measurement system. By using the implemented system, we can measure the user's perceptual quality of VoIP as well as the network QoS metrics, in terms of criteria such as packet loss rate, jitter and burstiness in real time.

A rate-6/4 full-diversity orthogonal space-time block code (STBC) is constructed for QPSK and 2 transmit antennas by applying constellation scaling and rotation to the set of quaternions used in Alamouti code. Also given is a rate-9/8 full-diversity quasi-orthogonal space-time block code (QOSTBC) for 4 transmit antennas. Lastly, a rate-10/8 code is presented for 4 transmit antennas. Simulation results indicate that these high-rate codes achieve better throughputs in the high signal-to-noise ratio region.

Recent rapid development of high-speed wireless access technologies has created mixed WLAN (Wireless LAN) environments where QoS capable APs coexist with legacy APs. To provide QoS guarantee in this mixed WLAN environment, this paper proposes a new AP selection algorithm. The proposed algorithm assigns an STA (Station) to an AP in the overall WLAN service area. Simulation results show improvement in the VoIP performance in terms of an eMOS (estimated Mean Opinion Score) value and the FTP throughput compared to conventional algorithms.

In next generation mobile multimedia communications, different wireless access networks are expected to cooperate. However, it is a challenging task to choose an optimal transmission path in this scenario. This paper focuses on the problem of selecting the optimal access network for multicast services in the cooperative mobile and broadcasting networks. An algorithm is proposed, which considers multiple decision factors and multiple optimization objectives. An analytic hierarchy process (AHP) method is applied to schedule the service queue and an artificial neural network (ANN) is used to improve the flexibility of the algorithm. Simulation results show that by applying the AHP method, a group of weight ratios can be obtained to improve the performance of multiple objectives. And ANN method is effective to adaptively adjust weight ratios when users' new waiting threshold is generated.

This paper performs application-level QoS and user-level QoS assessment of audio-video streaming in cross-layer designed wireless ad hoc networks. In order to achieve high QoS at the user-level, we employ link quality-based routing in the network layer and media synchronization control in the application layer. We adopt three link quality-based routing protocols: OLSR-SS (Signal Strength), AODV-SS, and LQHR (Link Quality-Based Hybrid Routing). OLSR-SS is a proactive routing protocol, while AODV-SS is a reactive one. LQHR is a hybrid protocol, which is a combination of proactive and reactive routing protocols. For application-level QoS assessment, we performed computer simulation with ns-2 where an IEEE 802.11b mesh topology network with 24 nodes was assumed. We also assessed user-level QoS by a subjective experiment with 30 assessors. From the assessment results, we find AODV-SS the best for networks with long inter-node distances, while LQHR outperforms AODV-SS for short inter-node distances. In addition, we also examine characteristics of the three schemes with respect to the application-level QoS in random topology networks.