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1.
Voice service is very demanding in cognitive radio networks (CRNs). The available spectrum in a CRN for CR users varies owing to the presence of licensed users. On the other hand, voice packets are delay sensitive and can tolerate a limited amount of delay. This makes the support of voice traffic in a CRN a complicated task that can be achieved by devising necessary considerations regarding the various network functionalities. In this paper, the support of secondary voice users in a CRN is investigated. First, a novel packet scheduling scheme that can provide the required quality of service (QoS) to voice users is proposed. The proposed scheme utilizes the maximum packet transmission rate for secondary voice users by assigning each secondary user the channel with the best level of quality. Furthermore, an analytical framework developed for a performance analysis of the system, is described in which the effect of erroneous spectrum sensing on the performance of secondary voice users is also taken into account. The QoS parameters of secondary voice users, which were obtained analytically, are also detailed. The analytical results were verified through the simulation, and will provide helpful insight in supporting voice services in a CRN. 相似文献
2.
Next generation high capacity wireless networks need to support various types of traffic, including voice, video and data, each of which have different Quality of Service (QoS) requirements for successful transmission. This paper presents an advanced reservation packet access protocol BRTDMA (Block Reservation Time Division Multiple Access) that can accommodate voice and data traffic with equal efficiency in a wireless network. The proposed BRTDMA protocol has been designed to operate in a dynamic fashion by allocating resources according to the QoS criteria of voice and data traffic. Most of the existing reservation protocols offers reservation to voice traffic while data packets are transmitted using contention mode. In this paper we propose a block reservation technique to reserve transmission slots for data traffic for a short duration, which minimizes the speech packet loss and reduce the end-to-end delay for wireless data traffic. The optimum block reservation length for data traffic has been studied in a cellular mobile radio environment using a simulation model. Simulation results show that the BRTDMA protocol offers higher traffic capacity than standard PRMA protocol for integrated voice and data traffic and offers flexibility in accommodating multimedia traffic. 相似文献
3.
Seamless SIP-based mobility for multimedia applications 总被引:4,自引:0,他引:4
Application-level protocol abstraction is required to support seamless mobility in next-generation heterogeneous wireless networks. Session initiation protocol (SIP) provides the required abstraction for mobility support for multimedia applications in such networks. However, the handoff procedure with SIP suffers from undesirable delay and hence packet loss in some cases, which is detrimental to applications like voice over IP (VoIP) or streaming video that demand stringent quality of service (QoS) requirements. In this article we present a SIP-based architecture that supports soft handoff for IP-centric wireless networks. Soft handoff ensures that there is no packet loss and that the end-to-end delay jitter is kept under control. 相似文献
4.
‘Anytime, anywhere’ communication, information access and processing are much cherished in modern societies because of their ability to bring flexibility, freedom and increased efficiency to individuals and organizations. Wireless communications, by providing ubiquitous and tetherless network connectivity to mobile users, are therefore bound to play a major role in the advancement of our society. Although initial proposals and implementations of wireless communications are generally focused on near‐term voice and electronic messaging applications, it is recognized that future wireless communications will have to evolve towards supporting a wider range of applications, including voice, video, data, images and connections to wired networks. This implies that future wireless networks must provide quality‐of‐service (QoS) guarantees to various multimedia applications in a wireless environment. Typical traffic in multimedia applications can be classified as either Constant‐Bit‐Rate (CBR) traffic or Variable‐Bit‐Rate (VBR) traffic. In particular, scheduling the transmission of VBR multimedia traffic streams in a wireless environment is very challenging and is still an open problem. In general, there are two ways to guarantee the QoS of VBR multimedia streams, either deterministically or statistically. In particular, most connection admission control (CAC) algorithms and medium access control (MAC) protocols that have been proposed for multimedia wireless networks only provide statistical, or soft, QoS guarantees. In this paper, we consider deterministic QoS guarantees in multimedia wireless networks. We propose a method for constructing a packet‐dropping mechanism that is based on a mathematical framework that determines how many packets can be dropped while the required QoS can still be preserved. This is achieved by employing: (1) An accurate traffic characterization of the VBR multimedia traffic streams; (2) A traffic regulator that can provide bounded packet loss and (3) A traffic scheduler that can provide bounded packet delay. The combination of traffic characterization, regulation and scheduling can provide bounded loss and delay deterministically. This is a distinction from traditional deterministic QoS schemes in which a 0% packet loss are always assumed with deterministically bounding the delay. We performed a set of performance evaluation experiments. The results will demonstrate that our proposed QoS guarantee schemes can significantly support more connections than a system, which does not allow any loss, at the same required QoS. Moreover, from our evaluation experiments, we found that the proposed algorithms are able to out‐perform scheduling algorithms adopted in state‐of‐the‐art wireless MAC protocols, for example Mobile Access Scheme Based on Contention and Reservation for ATM (MASCARA) when the worst‐case traffic is being considered. Copyright © 2002 John Wiley & Sons, Ltd. 相似文献
5.
Donald F. Gingras J. Bibb Cain Stephan K. Lapic 《International Journal of Wireless Information Networks》2000,7(3):149-165
A medium access control (MAC) protocol for wireless mobile networks that supports integrated services and provides quality of service (QoS) support is presented and evaluated via simulation. A controlled random access protocol which allows all terminals to dynamically share a group of spread spectrum spreading codes is used. The protocol provides mobile terminals the access control required for efficient transfer of integrated traffic with QoS guarantees. Two service classes are provided; "best-effort" service, with priority queueing, and reserved bandwidth circuit service. The performance of the protocol is evaluated via simulation for traffic consisting of integrated voice, data and compressed video. The performance assessment measure is packet delay. 相似文献
6.
In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks. 相似文献
7.
Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance. 相似文献
8.
With the pervasive growth in the popularity of IEEE 802.11‐based wireless local area networks (WLANs) worldwide, the demand to support delay‐sensitive services such as voice has increased very rapidly. This paper provides a comprehensive survey on the medium access control (MAC) architectures and quality of service (QoS) provisioning issues for WLANs. The major challenges in providing QoS to voice services through WLAN MAC protocols are outlined and the solution approaches proposed in the literature are reviewed. To this end, a novel QoS‐aware wireless MAC protocol, called hybrid contention‐free access (H‐CFA) protocol and a call admission control technique, called traffic stream admission control (TS‐AC) algorithm, are presented. The H‐CFA protocol is based on a novel idea that combines two contention‐free wireless medium access approaches, that is, round‐robin polling and time‐division multiple access (TDMA)‐like time slot assignment, and it increases the capacity of WLANs through efficient silence suppression. The TS‐AC algorithm ensures efficient admission control for consistent delay‐bound guarantees and further maximizes the capacity through exploiting the voice characteristic that it can tolerate some level of inconsecutive packet loss. The benefits of the proposed schemes are demonstrated in the simulations results. Copyright © 2008 John Wiley & Sons, Ltd. 相似文献
9.
Efficient utilization of network resources is a key goal for emerging broadband wireless access systems (BWAS). This is a complex goal to achieve due to the heterogeneous service nature and diverse quality of service (QoS) requirements of various applications that BWAS support. Packet scheduling is an important activity that affects BWAS QoS outcomes. This paper proposes a novel packet scheduling mechanism that improves QoS in mobile wireless networks which exploit IP as a transport technology for data transfer between BWAS base stations and mobile users at the radio transmission layer. In order to improve BWAS QoS the new packet algorithm makes changes at both the IP and the radio layers. The new packet scheduling algorithm exploits handoff priority scheduling principles and takes into account buffer occupancy and channel conditions. The packet scheduling mechanism also incorporates the concept of fairness. Performance results were obtained by computer simulation and compared to the well known algorithms. Results show that by exploiting the new packet scheduling algorithm, the transport system is able to provide a low handoff packet drop rate, low packet forwarding rate, low packet delay and ensure fairness amongst the users of different services. 相似文献
10.
Jitter performance in ethernet passive optical networks 总被引:4,自引:0,他引:4
Ethernet passive optical networks (EPONs) have emerged as one of the most promising access network technologies. Propelled by rapid price declines in fiber optics and Ethernet components, these architectures combine the latest in optical and electronic advances and are poised to become the dominant means of delivering gigabit broadband connectivity to homes over a unified single platform. As this technology matures, related quality of service (QoS) issues are becoming a key concern. This paper proposes a novel dynamic scheduling algorithm, termed hybrid granting protocol (HGP), to support different QoS in EPON. Specifically, the proposed dynamic scheduling algorithm minimizes packet delay and jitter for delay and delay-variation sensitive traffic (e.g., voice transmissions) by allocating bandwidth in a grant-before-report (GBR) fashion. This considerably improves their performance without degrading QoS guarantees for other service types. Detailed simulation experiments are presented to validate the effectiveness of the proposed algorithm. 相似文献
11.
12.
Hai Jiang Ping Wang H. Vincent Poor Weihua Zhuang 《Wireless Communications and Mobile Computing》2010,10(4):547-558
Future mobile ad hoc networks are expected to support voice traffic. The requirement for small delay and jitter of voice traffic poses a significant challenge for medium access control (MAC) in such networks. User mobility presents unique difficulties in this context due to the associated dynamic path attenuation. In this paper, a MAC scheme for mobile ad hoc networks supporting voice traffic is proposed. With the aid of a low‐power probe prior to DATA transmissions, resource reservation is achieved in a distributed manner, thus leading to small packet transmission delay and jitter. The proposed scheme can automatically adapt to dynamic path attenuation in a mobile environment. Statistical multiplexing of on/off voice traffic can also be achieved by partial resource reservation for off voice flows. Simulation results demonstrate the effectiveness of the proposed scheme. Copyright © 2009 John Wiley & Sons, Ltd. 相似文献
13.
To guarantee the quality of service (QoS) of a wireless network, a new packet scheduling algorithm using cross-layer design technique is proposed in this article. First, the demand of packet scheduling for multimedia transmission in wireless networks and the deficiency of the existing packet scheduling algorithms are analyzed. Then the model of the QoS-guaranteed packet scheduling (QPS) algorithm of high speed downlink packet access (HSDPA) and the cost function of packet transmission are designed. The calculation method of packet delay time for wireless channels is expounded in detail, and complete steps to realize the QPS algorithm are also given. The simulation results show that the QPS algorithm that provides the scheduling sequence of packets with calculated values can effectively improve the performance of delay and throughput. 相似文献
14.
15.
The requirement to provide multimedia services with QoS support in mobile networks has led to standardization and deployment
of high speed data access technologies such as the High Speed Downlink Packet Access (HSDPA) system. HSDPA improves downlink
packet data and multimedia services support in WCDMA-based cellular networks. As is the trend in emerging wireless access
technologies, HSDPA supports end-user multi-class sessions comprising parallel flows with diverse Quality of Service (QoS)
requirements, such as real-time (RT) voice or video streaming concurrent with non real-time (NRT) data service being transmitted
to the same user, with differentiated queuing at the radio link interface. Hence, in this paper we present and evaluate novel
radio link buffer management schemes for QoS control of multimedia traffic comprising concurrent RT and NRT flows in the same
HSDPA end-user session. The new buffer management schemes—Enhanced Time Space Priority (E-TSP) and Dynamic Time Space Priority
(D-TSP)—are designed to improve radio link and network resource utilization as well as optimize end-to-end QoS performance
of both RT and NRT flows in the end-user session. Both schemes are based on a Time-Space Priority (TSP) queuing system, which
provides joint delay and loss differentiation between the flows by queuing (partially) loss tolerant RT flow packets for higher
transmission priority but with restricted access to the buffer space, whilst allowing unlimited access to the buffer space
for delay-tolerant NRT flow but with queuing for lower transmission priority. Experiments by means of extensive system-level
HSDPA simulations demonstrates that with the proposed TSP-based radio link buffer management schemes, significant end-to-end
QoS performance gains accrue to end-user traffic with simultaneous RT and NRT flows, in addition to improved resource utilization
in the radio access network. 相似文献
16.
As wireless networks become more widely used, there is a growing need to support advanced services, such as multimedia streaming and voice over IP. Traditional approaches to guarantee quality of service (QoS) work well only with predictable channel and network access. In wireless mobile networks, where conditions dynamically change as nodes move about the network, a stateless, high level approach is required. Since shared wireless resources are easily over-utilized, the load in the network must be controlled so that an acceptable QoS for real-time applications can be maintained. If minimum real-time requirements are not met, these unusable packets waste scarce bandwidth and hinder other traffic, compounding the problem. To enable high QoS for all admitted traffic, we propose the Perceptive Admission Control (PAC) protocol. PAC monitors the wireless channel and dynamically adapts admission control decisions to enable high network utilization while preventing congestion. Through discussion, simulations and testbed experiments, we demonstrate that PAC ensures low packet loss and delay for all admitted flows. 相似文献
17.
《Electronics letters》2002,38(15):789-790
A packet selection scheme, based on channel quality estimation, for Bluetooth is described. Link management based on this scheme can adaptively select a packet for maximising throughput while minimising transmission delay at any given channel quality, and can provide new degrees of freedom for higher layer issues such as buffer control, routing, QoS service and support of multimedia data 相似文献
18.
19.
Delay-on-Demand: A Signaling Protocol to Reduce Blocking Probability in Optical Burst-Switching Networks 总被引:2,自引:0,他引:2
Optical burst switching (OBS) is emerging as one promising switching paradigm for the next generation optical networks. To support multiple services in burst-switching networks, the OBS paradigm should support some quality-of-service (QoS) provisioning. A major design issue in such networks is to reduce the blocking probability of the bursts arising due to resource contention at the intermediate core router. In this paper, we propose a signaling protocol which we call ‘Delay-on-Demand’ (OBS-DoD), to reduce blocking probability and support QoS in optical burst-switching networks. The proposed scheme guarantees that at least one of the bursts succeeds depending on its priority, propagation delay from the ingress router, and the burst-size when contention occurs at the core router. For this, we use a control packet to delay, in case of a contention, the transmission of bursts at the ingress router. We compare the performance of our proposal, by simulation, with an earlier proposed scheme, and show that the proposed OBS-DoD outperforms the earlier scheme in reducing the blocking probability. For simulation, we generated bursty traffic using an M/Pareto distribution. 相似文献
20.
Zhu Han Xin Liu Z. Jane Wang K. J. R. Liu 《Wireless Communications, IEEE Transactions on》2007,6(2):423-428
Future wireless networks will support the growing demands of heterogeneous and delay sensitive applications. In this paper, a users' satisfaction factor (USF) is defined to quantify quality of service (QoS) for different types of services such as voice, data, and multimedia, as well as for different delay constraints. This USF not only predicts the final delivered QoS during transmission, but also take advantages of the fact that different packets can be decoded at different time in the receivers. Based on this USF, four types of scheduling schemes considering tradeoffs between system performance and individual fairness are proposed. These schemes explore the time, channel, and multi-user diversity to guarantee quality of service and enhance the network performance. From the simulation results, the proposed scheduling schemes achieve different tradeoffs between individual fairness and high system performance for the heterogeneous and delay sensitive applications, compared with the weighted round-robin and the modified proportional fairness scheduling schemes 相似文献