We propose a new resource management scheme, Distributed Channel Allocation Protocol (DCAP), for multimedia Ad hoc Wireless LANs (AWLANs). This scheme implements a Quality-of-Service (QoS) providing distributed resource management on the Orthogonal Frequency Division Multiplexing-Code Division Multiple Access (OFDM-CDMA) channel architecture. According to the performance evaluation results for MPEG traffic sources, DCAP can be a good choice of resource management scheme for AWLANs supporting multimedia services on the Broadband Wireless Access (BWA)-type physical layer.

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.

In this paper, the capacity is analyzed for a CDMA system supporting heterogeneous traffic with on/off activity. The capacity is analyzed by modeling the on/off traffic activity as a binomial random variable and compared to the conventional analysis with the simplified traffic activity factor which is a constant. It has been shown that the capacity with the conventional analysis is more optimistic than the capacity analyzed with the binomial modeling of traffic activity. The effect of traffic activity on the capacity is further investigated for two different cases. One is the case under the same transmission rate where the average rate changes according to the traffic activity. The other is the case under the same average rate where the transmission rate changes according to the traffic activity. As the traffic activity factor becomes larger, the capacity increases for the case under the same average rate, whereas it decreases for the case under the same transmission rate.

To recover the lost packets in VoIP systems, we propose a dynamic redundant audio transmission mechanism that responds to network status. A receiver computes the QoS information using timestamps and sequence numbers in RTP headers and sends the information to a sender using the RTCP protocol. The sender transmits redundant audio packets to the receiver to recover the lost packets. Experimental results show that the rate of recovering the lost packets is 63% in case of consecutive packet loss and is 42% in case of non-consecutive packet loss.

The IMT-2000 service launched in 2001 in Japan is expected to popularize multimedia services such as videophone, visual mail, video, and music distribution. With the rapidly increasing demand for high-speed mobile multimedia and the need to support diversified requirements of users, wireless Quality of Service (QoS) resource management has become an important and challenging issue. In order to improve the system capacity and rate of satisfied users, in this paper, a novel wireless QoS resource management scheme is proposed to carry out radio resource cooperative control among base stations. Computer simulations indicate that the proposed QoS resource cooperative control exhibits superior performance over conventional ones, and that a higher rate of satisfied users is achieved.

This letter addresses the QoS issue of a RSVP flow during handoff events. For the QoS guarantee of real-time applications with RSVP reservation in All-IP wireless networks, mechanisms are required to minimize the resource reservation path changes and the packet loss resulting from handoff events. If the new RSVP reservation is made along the path via a new base station (BS) in advance for soft handoff, on-going RSVP flow session can be kept with the guaranteed QoS. Therefore, we propose a seamless switching scheme of RSVP branch path for soft handoff. We also show that this scheme could provide the QoS guarantee by adaptively adjusting the pilot signal threshold values for soft handoff.

Optical Burst Switching (OBS) is one of the most important switching technologies in future optical Internet. One of critical design issues in OBS is how to reduce burst dropping resulting from resource contention. Especially when traffic load is high, there should be frequent deflection routing as well as more contentions in an optical burst-switched network. The burst loss performance can be improved by implementing a proper deflection routing scheme. In this paper, we propose a limited deflection routing scheme to prevent injudicious deflection routing. The proposed scheme reduces unnecessary contentions resulting from deflection routing itself, increasing the utilization of network resource such as channels. Simulation tests were performed to evaluate the performance of the proposed scheme.

This letter addresses the issue of RSVP path management in IP micro-mobility networks. We describe efficient RSVP QoS paths with a minimal impact to the existing protocol and underlying routing infrastructure. The goal of this letter is to reduce RSVP path reservation restoration latency and unnecessary control traffic caused by mobility events. We thus propose a RSVP branch-path rerouting scheme at a crossover router (CR) under IP micro-mobility networks. We show that this scheme could give a good tradeoff between the resource reservation cost and the link usage during the lifetime of a RSVP connection.

Many congestion control mechanisms have been proposed to solve the problems of a high loss rate and inefficient utilization of network resources in the present Internet. This problem is caused by competition between traffic flows while the network is congested. Differentiated Services (DiffServ) architecture permits the allocation of various levels of traffic resource requirements needed for Quality of Service (QoS). Random Early Detection (RED) is an efficient mechanism to pre-drop packets before actual congestion occurs, and it is capable of introducing a random early packet dropping scheme, and based on the queue length in reaching a certain degree of fairness for resource utilization. However, it still suffers from a lack of robustness among light traffic load, or in heavy traffic load using fixed RED parameters. In this paper, we modified the RED scheme and proposed a novel adaptive RED model, which we named the OURED model, to enhance the robustness of resource utilization so that it could be utilized in the DiffServ edge router. The OURED model introduces two additional packet dropping traces, one is Over Random Early Detection (ORED), which is used to speed up the dropping of packets when the actual rate is higher than the target rate, and the other one is the Under Random Early Detection (URED), used to slow down the packet dropping rate in the reverse situation. The simulation results show that OURED is not only more robust than MRED in resource utilization, but that it also can be implement efficiently in the DiffServ edge router.

Different quality of service (QoS) requirements must be guaranteed in multimedia code division multiple access (CDMA) mobile communication system to support various applications such as voice, video, file transfer, e-mail, and Internet access. In this paper, we analyze the system in a mixed-traffic environment consisting of voice, stream data, and packet data where preemptive priority is granted to delay-intolerant voice service and a buffer is offered to delay-tolerant stream data services. In the case of best-effort packet data service, the probability of access control by transmission permission is applied to obtain throughput improvement. To analyze the multimedia CDMA mobile communication system, we built a two-dimensional Markov chain model on high-priority voice and stream data services, and then performed numerical analyses in conjunction with packet data services based on a residual capacity equation. We investigate the performance of an integrated voice/stream-data/packet-data CDMA mobile system with the finite buffer size of stream data in terms of voice service blocking probability, average stream data service delay, average packet data service delay, and throughput.

This paper proposes a wireless scheduling algorithm that can provide the Internet with delay proportional differentiated services in wireless networks. When considering wireless network environments that include burst and location-dependent channel errors, the proposed WDPS (Wireless Delay Proportional Service) scheduling algorithm can adaptively serve packets in class queues based on the delivered delay performance for each class. The significant characteristics of the WDPS scheduler include support for a fair relative delay service, the provision of graceful throughput and delay compensations, and the avoidance of class queue-blocking problems. Simulations show that the proposed algorithm can achieve the desirable properties for providing delay proportional services in wireless networks.

Appropriate test signals defined by formula or generated by algorithm are used for measuring objective QoS (Quality of Services) for voice operated telecommunication devices such as telephone and speech codec (coder-decoder). However, that for measuring residual echo characteristics in hands-free telecommunications equipped with acoustic echo canceller is under study in ITU-T Recommendation G.167. This paper describes comparative assessment of test signals for measurement of residual echo characteristics. In hands-free telecommunications, acoustical echo canceller has been developed to remove a room echo signal through the loudspeaker to the microphone in the receiving end. Performance of the echo canceller system is evaluated by residual echo characteristics expressed in echo return loss enhancement (ERLE). The ERLE can be conventionally measured by putting white noise into the echo canceller system. However, white noise is not adequate as the test signal for measuring the performance of the echo canceller, since the performance may depend on the characteristics of input test signal, and the characteristics of the white noise differ from those of real voice. Therefore, this paper discusses appropriate characteristics of real voice required for objective quality evaluation of echo canceller system. The test signals used for this verification tests were real voice (RV), white noise (WN), frequency weighted noise (FWN), artificial voice (AV), and composite source signal (CSS) depending on the approximation of real voice characteristics. As the comparative assessment results, the ERLE characteristics measured by artificial voice conforming to ITU-T Recommendation P.50 having average characteristics of real voices in time and frequency domains are almost equivalent to those of real voice and best among those test signals. It is concluded that artificial voice P.50 is satisfied with measurement of residual echo characteristics.

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.

The differentiated services (DiffServ) architecture is proposed to provide a service differentiation between traffic classes or behavior aggregates in a scalable manner. A key functional element to deploy DiffServ is traffic conditioning, more specifically traffic marker. This paper proposes an adaptive and aggregated traffic marker embodying the functions: (1) inter-connect two-rate three color markers (trTCMs), (2) estimate the aggregate rate of Assured Forwarding (AF) classes, and (3) re-mark the some parts of excessive portion of assigned link-rate per AF class queue with some down-grading probability to lower AF class. Both analysis and simulations are used to evaluate the performance of the proposed aggregate-traffic marker. The analysis of the proposed marker shows its clear service differentiation among behavior aggregates (BAs) under different traffic load conditions. Also a performance evaluation is performed through network simulation with more realistic traffics as unbalanced intensities among different BAs. Shortly, our proposed aggregate-traffic marker enables to keep the priority orders in terms of loss rate and delay/jitter among BAs in spites of varying and unbalanced traffic intensities.

As the Internet role changes from the experimental environment to the social infrastructure, end-to-end quality for data transfer of various types of traffic has been required as well as its connectivity. So we should precisely measure some performance such as packet loss probability and delay at routers on some source-destination path. By using so-called passive measurement technique, we can get the queue length distribution from some routers individually and estimate the end-to-end transmission delay. However, there may be some correlation between queue lengths of two or more routers packets go through in sequence, which would lead to inaccurate estimation of end-to-end delay performance. Thus in this paper, we model two tandem routers as queueing system, and analyze the queue length distributions and their correlation. Through some numerical results, we will investigate the impact of traffic parameters on the degree of correlation and how it affects the estimation of delay performance.

One purpose of Quality-of-Service (QoS) routing is to develop polynomial-time heuristic algorithms to tackle the MCP (multi-constrained-path) problem, which is NP-complete. In this paper, we introduce a new QoS routing heuristic framework, which focuses on how to increase the success ratio for finding a feasible path subject to multiple additive constraints. The key issue of this framework is to transform the single source single destination QoS routing problem to a single source multi-destination problem by expanding the destination vertex to its neighboring vertices. After that, the modified problem can be solved by existing source routing heuristic algorithms. The analysis and simulation results demonstrate that the framework can achieve a higher success ratio of finding a feasible path without increasing the computational complexity by setting the expansion operation properly.

In the past, enhancement techniques for the end-to-end quality of a networked application were studied by looking at each individual layer. Examples of such techniques include the error resilience/concealment methods in the application layer, the FEC/ARQ in the transport layer, and the Quality of Service (QoS) techniques in the network layers. However, an integrated approach that would look across all related layers had yet to be investigated. This paper proposes an approach that combines priority-aware video packetization, adaptively used layered FEC, and QoS controlled networks such as IntServ and DiffServ in order to provide an efficient end-to-end quality in layered streaming video. The combination is more efficient in terms of a simple network price mechanism, that is, in getting the best end-to-end quality under a given total cost constraint. Our proposed approach in DiffServ with video packet (VP) data-splitting and layered FEC guarantees minimal service quality in a scalable and cost effective manner without introducing resource reservation. For video, we also propose performance metrics such as corrupted/frozen frame ratio (CFR, FFR). These provide objective metrics like PSNR as well as a measurement for subjective perceptions. Our approach, which combines related layers such as video coding, transport, and network, has yielded results that have proven to be more cost-effective and practical than the supporting network QoS.

Existing research related to RSVP with mobility support has mainly focused on maintaining reservation state along the routing path, which changes continuously with the movements of mobile host (MH), without much overhead and delay. However, problems such as deepening RSVP's inherent scalability problem and requiring significant changes in the existing network infrastructure have not been adequately addressed. In this paper, we propose a new approach, known as Concatenation and Optimization for Reservation Path (CORP), which addresses these issues. In CORP, each BS pre-establishes pseudo reservations to its neighboring BSs in anticipation of the MH's movement. When the MH moves into another wireless cell, the associated pseudo reservation is activated and concatenated to the existing RSVP session to guarantee continuous QoS support. Because a pseudo reservation is recognized as a normal RSVP session by intermediate routers, little change is required in the current Internet environment to support both movements within a single routing domain and between two different routing domains. CORP also dynamically optimizes the extended reservation path to avoid the infinite path extension problem. Multicast addressing is used to further reduce resource consumption in the optimization process. The experimental results of the CORP implementation demonstrate that it significantly reduces the delay and overhead caused by handoffs compared to the case of establishing a new RSVP session. The improvement increases as the distance between the MH and its correspondent host (CH) grows.

In IP networks, QoS guarantee is becoming important, to allow real-time services such as voice over IP (VoIP) and video conferencing. To guarantee various QoS requirements from a variety of applications, call admission control (CAC) together with a service differentiation mechanism is useful. Service differentiation enables the network to provide different QoSs to various applications, by assigning priority classes to calls. Then, CAC limits the acceptance of new calls to prevent the QoSs of established calls from degrading. For this environment, we propose a CAC method in which priority classes assigned to already established calls are changed adaptively. Classes may be modified when the CAC function handles the acceptance of a new call if that can decrease the blocking probability. The effect of the method is shown by numerical examples.

This paper proposes an adaptive radio parameter control scheme that utilizes an optimum radio parameter set comprising the maximum number of retransmissions in hybrid automatic repeat request (HARQ) in addition to the data modulation and channel coding scheme (MCS) according to the Quality of Service (QoS) requirements (i.e., the required packet error rate and delay) and propagation conditions such as the delay spread in the forward link of Orthogonal Frequency and Code Division Multiplexing (OFCDM) broadband wireless access. We elucidate by simulation evaluation that most of the optimum MCSs are common regardless of the delay requirement of traffic data, i.e., common between non-real time (NRT) and real-time (RT) class data. Concretely, the three MCSs of QPSK with the coding rate of R=1/2, 16QAM with R=1/2 and 3/4 are optimum ones, although the additional MCS of QPSK with R=1/3 is effective only for the RT class data in the lower received average received signal energy per symbol-to-background noise power density ratio (Es/N0) region. Furthermore, application of a much higher MCS set, 16QAM with R=5/6 and 64QAM with R=3/4, in addition to the three common MCSs improves the throughput under much higher Es/N0 conditions in a small delay spread environment. The simulation results show that the delay requirement, i.e., the maximum number of retransmissions, in HARQ does not affect the key radio parameter such as MCS, because of informative results such as a smaller number of retransmissions associated with a less-efficient MCS achieves a higher throughput than does using a more highly-efficient MCS allowing a larger number of retransmissions. Consequently, it is concluded that the proposed adaptive radio parameter control according to the QoS requirements substantially results in the selection of the optimum MCS irrespective of the delay requirement except for the extreme case where no retransmissions are allowed and for special propagation channel conditions.