A call admission control framework for voice over WLANs

In this article a call admission control framework is presented for voice over wireless local area networks (WLANs). The framework, called WLAN voice manager, manages admission control for voice over IP (VoIP) calls with WLANs as the access networks. WLAN voice manager interacts with WLAN medium access control (MAC) layer protocols, soft-switches (VoIP call agents), routers, and other network devices to perform end-to-end (ETE) quality of service (QoS) provisioning and control for VoIP calls originated from WLANs. By implementing the proposed WLAN voice manager in the WLAN access network, a two-level ETE VoIP QoS control mechanism can be achieved: level 1 QoS for voice traffic over WLAN medium access and level 2 QoS for ETE VoIP services in the networks with WLANs as the local access. The implementation challenges of this framework are discussed for both level 1 and level 2. Possible solutions to the implementation issues are proposed and other remaining open issues are also addressed.     

Supporting multimedia applications through network redesign

Recent evolutions in high‐performance computing and high speed broadband Internet access have paved a way to enterprise‐wide multimedia applications, which require stern QoS from the underlying networks. In this paper, we have explored threefold studies on existing enterprise network, whereby we proposed an analytical approach to evaluate the performance of the existing network; we have examined the feasibility of existing enterprise networks to accommodate voice over Internet protocol (VoIP) services with acceptable QoS, and we have redesigned the enterprise network to accommodate VoIP services to comply with the user defined QoS. The network performance is evaluated by number of VoIP calls sustained by the network, bandwidth utilization, loss rate and latency through Network Simulation (NS‐2) tool. We have derived a cost model to show the cost‐effectiveness of VoIP services over telephonic network. For a medium‐size enterprise network of 200 clients and 9 servers, our simulation results show that the redesign improves the network performance by increasing the number of VoIP calls by 57% and decreasing bandwidth utilization and packet loss rate by 20% and 7%, respectively. Moreover, the proposed network redesign demonstrates that the network can be scalable and it can handle up to 4% increased voice calls in the future maintaining QoS standards. Copyright © 2012 John Wiley & Sons, Ltd.     

Call-Level and Packet-Level Quality of Service and User Utility in Rate-Adaptive Cellular CDMA Networks: A Queuing Analysis

A queuing analytical model is presented to evaluate call-level and packet-level quality of service (QoS) metrics in the uplink of a voice/data cellular code division multiple access (CDMA) network. In this model, a threshold-based call admission control (CAC) is used to limit the number of admitted calls in a cell and also to prioritize handoff calls over new calls. The transmission rates for data calls can be adjusted to accommodate more voice and/or data calls while satisfying the minimum signal-to-interference ratio (SIR)/ transmission rate requirement. Also, automatic repeat request (ARQ)-based error control is used for improved reliability of data packets. Call-level performance measures for both voice and data calls and packet-level performance measures specifically for data calls can be obtained from the analytical model. The interdependencies among call-level and packet-level QoS metrics are investigated under different CAC, rate adaptation, and error control parameter settings. To this end, the level of users' satisfaction (or user utility) is formulated as a function of the QoS metrics and an optimization formulation is presented to obtain the local-optimal system parameters     

Performance evaluation on IEEE 802.11e considering emergency calls in congested situation

To enhance the quality of service (QoS) support in IEE 802.11, IEEE 802.11e has been studied, which introduces the so-called hybrid coordination function (HCF). HCF includes two medium access mechanisms contention-based channel access (EDCA), and contention-free channel access (HCCA). Although IEEE 802.11e has provided differentiated channel access mechanism, when call demand rises for important festivals such as New Year's Day or large scale natural disasters such as earthquakes, the delay of voice will increase and the QoS of voice nodes will drop down rapidly. Through our simulation study, in order to guarantee the QoS of emergency voice calls in congested situation, a higher priority for these calls will be required.     

On the impact of mobility and soft vertical handoff on voice admission control in loosely coupled 3G/WLAN networks

Loose coupling between 3G and WLAN ensures flexibility and openness. However, providing an ubiquitousmobile voice service in a loosely coupled 3G/WLAN network requires both packet-level and call-level quality of service (QoS) guarantees using soft vertical handoff (SVHO) and call admission control (CAC). In this paper, we evaluate the impact of both SVHO and WLAN mobility on call blocking and dropping probabilities rederived for the integrated network. For this purpose, we propose a new multi-region mobility model that accurately estimate these probabilities under a resource-efficient dynamicthreshold SVHO compared to a standard static-threshold SVHO. Results show us that the resource-efficient SVHO blocks and drops much less voice calls than the static one when very low mean and high variability of multi-mode mobile station velocities are noticed. Therefore, resource-efficient SVHO implementations are highly recommended in these mobility environments.     

On the support of voice call continuity across UMTS and wireless LANs

In this paper, we address the architecture and the procedures that can enable voice call handover from UMTS to WLAN and we also study how efficiently the WLAN can support the voice calls transferred from UMTS. Our study is based on a practical simulation model that lets us quantify the maximum number of voice calls that can be handed over from UMTS to WLAN, subject to maintaining the same level of UMTS QoS and respecting some WLAN policies. In addition, several other voice call performance metrics are derived. Our results indicate that an IEEE 802.11e access point can support a limited number of voice calls handed over from UMTS, which depends primarily on the applied WLAN bandwidth sharing policy (i.e., how the bandwidth is shared between WLAN voice and data users) and the QoS requirements. The performance of the WLAN scheduling algorithm is also of paramount importance and in our study we consider the so‐called ARROW scheduler. Although the simulation results are derived for a specific bandwidth sharing policy, they can readily be scaled and provide practical upper bounds for the number of UMTS voice calls that can be seamlessly admitted to a WLAN access point. Copyright © 2007 John Wiley & Sons, Ltd.     

The Threats of Infrastructure Obsolescence to Smart Grid: A Case Study

Network convergence has vital importance due to limited network resources. Though it decreases network cost, it also reduces Quality of Service (QoS) as well. Due to the increase in real-time network applications, such as video conferencing and voice over IP calls, there is a need to achieve fairness in user's demand through QoS. QoS measurement tasks are more challenging and increased network losses in infrastructure less network. QoS is not yet implemented in Wireless Local Area Network (WLAN) using Software Defined Network (SDN). We have proposed a model to support QoS in WLAN using SDN architecture for real-time traffic. It is efficiently utilizing bandwidth and reducing flow starvation due to centralize control. We named it, an Adaptive QoS model. This model allocates queues dynamically to flows using real-time traffic analysis in SDN architecture. A specific queue is allocated to a specific type of traffic. Flows are dynamically switched based on the increase in traffic demand if other queues are under-use. We achieved handsome improvement in Adaptive QoS model rather than a standard QoS model in terms of throughput and losses.

     

Quality assurance of voice over WLANs (VoWLANs) with differentiated services

Several technical issues make commercial and large voice over wireless local area network (VoWLAN) services difficult to provide. The most challenging issue when voice over Internet Protocol (VoIP) services are ran over IEEE 802.11-based WLANs is the bandwidth inefficiency due to the considerable overhead associated with WLAN packet transmission. In this work, we propose a session-based quality-of-service management architecture (SQoSMA) to overcome the low number of VoIP calls in IEEE 802.11 Wireless LANs and the negative effect of new call addition when the WLAN reaches its capacity. The SQoSMA combines data and control planes to detect VoWLAN QoS degradations and performs either an adaptive audio codec switching or a call stopping to fix VoWLAN issues in a differentiated services manner. In addition, our solution deals with user sessions information, by considering user priority (from its agreement) to guarantee a certain level of its multimedia applications. Performance evaluation using a real test-bed shows that call codec change and call stopping techniques can easily assure high-priority calls with acceptable call blocking probability.     

Design and Analysis of a Channel Assignment Scheme for CDMA/TDMA Mobile Networks

In this paper, a channel assignment scheme is proposed for use in CDMA/TDMA mobile networks carrying voice and data traffic. In each cell, three types of calls are assumed to compete for access to the limited number of available channels by the cell: new voice calls, handoff voice calls, and data calls. The scheme uses the movable boundary concept in both the code and time domains in order to guarantee the quality of service (QoS) requirements of each type. A traditional Markov analysis method is employed to evaluate the performance of the proposed scheme. Measures, namely, the new call blocking probability, the handoff call forced termination probability, the data call loss probability, the expected number of handoff and the handoff link maintenance probability are obtained from the analysis. The numerical results, which are validated by simulation, indicate that the scheme helps meet the QoS requirements of the different call types.     

Adaptive quality of service handoff priority scheme for mobilemultimedia networks

For various advantages including better utilization of radio spectrum (through frequency reuse), lower mobile transmit power requirements, and smaller and cheaper base station equipment, future wireless mobile multimedia networks are likely to adopt micro/picocellular architectures. A consequence of using small cell sizes is the increased rate of call handoffs as mobiles move between cells during the holding times of calls. In a network supporting multimedia services, the increased rate of call handoffs not only increases the signaling load on the network, but makes it very difficult for the network to guarantee the quality of service (QoS) promised to a call at setup or admission time. This paper describes an adaptive QoS handoff priority scheme which reduces the probability of call handoff failures in a mobile multimedia network with a micro/picocellular architecture. The scheme exploits the ability of most multimedia traffic types to adapt and trade off QoS with changes in the amount of bandwidth used. In this way, calls can trade QoS received for fewer handoff failures. The call level and packet level performance of the handoff scheme are studied analytically for a homogeneous network supporting a mix of wide-band and narrow-band calls. Comparisons are made to the performance of the nonpriority handoff scheme and the well-known guard-channel handoff scheme     

Efficient channel borrowing strategy for real-time services inmultimedia wireless networks

An efficient resource sharing strategy is proposed for multimedia wireless networks. We assume the channel resource in a wireless system is partitioned into two sets: one for voice calls and one for video calls. In the proposed channel borrowing strategy, voice calls can borrow channels from those pre-allocated to video calls temporarily when all voice channels are busy. A threshold type decision policy is designed such that the channel borrowing request will be granted only if the quality of service (QoS) requirement on video call blocking will not be violated during the duration of channel lending. An analytical model is constructed for evaluating the performance of the channel borrowing strategy in a simplified wireless system and is verified by computer simulations. We found that the proposed channel borrowing scheme can significantly reduce the voice call blocking probability while the increase in video call blocking probability is insignificant     

Handover in LTE-advanced wireless networks: state of art and survey of decision algorithm

The increasing demand for mobile communication calls for improvements to network operating services in terms of capacity, coverage, and Quality of Services (QoS). Ensuring QoS is one of the challenges faced by wireless network operators, which include the provision of high mobility speeds, thus the implementation of a seamless and fast handover between network cells is a prominent issue that must be addressed, especially when fulfilling QoS prerequisites. Long Term Evolution (LTE)/LTE-Advance has met these demands of QoS through the use of a new Radio Access Network and distribution of Radio Resource Management including the handover decision technique to evolve NodeB instead of relying on centralized control. In this paper, we review the control plane structure of LTE/LTE-A and present a comprehensive discussion of handover procedures such as the phases, techniques, requirements, features, and challenges involved. According to the overview of the handover decision phase, we surveyed and classified the present handover decision algorithms for a LTE-A system-based technology in regard to the primary handover decision technique. For each class, we describe in detail the fundamental operations and decision parameters using representative algorithms. A summary of input parameters, techniques, and performance evaluation of the handover decision algorithms concludes this work.     

Soft QoS in Call Admission Control for Wireless Personal Communications

In wireless multimedia communication systems, call admission control (CAC) is critical for simultaneously achieving a high resource utilization efficiency and maintaining quality-of-service (QoS) to mobile users. User mobility, heterogeneous nature of multimedia traffic, and limited radio spectrum pose significant challenges to CAC. QoS provisioning to both new calls and handoff calls comes with a cost of low resource utilization. This paper proposes a CAC policy for a wireless communication system supporting integrated voice and dataservices. In particular, soft QoS (or relaxed target QoS) is incorporated in the CAC policy to make compromises among different objectives.Numerical results are presented to demonstrate that (a) in dealing with the dilemma between QoS satisfaction and high resource utilization, how the resource utilization efficiency can be increased by introducing soft QoS; and (b) in accommodating different types of traffic, how the QoS of low priority traffic can be improved by specifying soft QoS to high priority traffic.     

Voice and data integration at an ATM DSL access multiplexor

We analyze cell loss and call blocking at an ATM-based DSL access multiplexor which integrates 2 types of calls onto a channel through a common buffer. Type-1 calls have stringent delay, but relaxed cell loss QoS requirement, and are representative of voice. Type-2 calls have stringent cell loss but relaxed delay QoS requirement, and are representative of Internet data. We study an access strategy that allocates separate buffers to the 2 cell types and assigns priority to the voice buffer in accessing the channel. We apply a fluid flow method to analyze both cell loss and call blocking of the two types of calls. These results are then used to size the admission region at the access node under loss and blocking constraints. Numerical results are presented which quantify the interaction between cell loss and call blocking and the utility of the priority scheme compared to the FIFO scheme in handling the two traffic types.     

NGN中语音业务质量监测参数的研究

为了及时管控下一代网络（NGN）中语音业务的服务质量,需要对NGN中的语音业务进行监测,然而,目前业界缺乏对该监测指标的明确定义。在分析了NGN中语音业务质量监测需求的基础上,从会话控制性能、话音质量、编码统计三个方面定义和描述了一套NGN中语音业务质量的监测指标,即NGN语音业务质量检测参数。对这些参数进行有效监测,将及时反映NGN中的语音业务质量和语音业务运营情况,为保证语音业务质量提供有力数据和参考。     

Channel Allocation for GPRS with Buffering Mechanisms

General Packet Radio Service (GPRS) provides mobile users end-to-end packet-switched services by sharing the radio channels with voice and circuit-switched services. In such a system, radio resource allocation for circuit-switched and packet-switched services is an important issue, which may affect the QoS for both services significantly. In this paper, we propose two algorithms: Dynamic Resource Allocation with Voice and Packet queues (DRAVP) and Dynamic Resource Allocation with Packet and Voice queues (DRAPV) for channel allocation of the voice calls and packets. We propose analytic and simulation models to investigate the performance of DRAVP and DRAPV in terms of voice call incompletion probability, packet dropping probability, average voice call waiting time, and average packet waiting time. Our study indicates that the buffering mechanism for GPRS packets significantly increase the acceptance rate of GPRS packets at the cost of slightly degrading the performance of voice calls.     

Hybrid coupling scheme for UMTS and wireless LAN interworking

We propose a hybrid coupling scheme to support interworking between UMTS and WLAN networks. Under the Tight-coupled system, it is expected that WLAN users can also use UMTS services with guaranteed QoS and seamless mobility. However, the interworking is problematic. The capacity of UMTS core network nodes cannot accommodate the bulky data traffic from WLAN, since the core network nodes are designed to handle the small-sized data of circuit voice calls or short packets. The proposed coupling scheme differentiates the data paths according to the type of the traffic and can accommodate traffic from WLAN efficiently, with guaranteed QoS and seamless mobility. We compare the handover procedures of the proposed coupling strategy with those of the loose and tight coupled schemes. In addition, we analyze the delay based on signaling costs during vertical handover. It is shown that the handover latency decreases when the UMTS and WLAN are coupled in the proposed way.     

车用自组网媒体访问控制机制改进

为提升车用自组网传输音频、视频的服务质量,对基于IEEE802.11p的车用无线接入技术MAC机制进行改进,提出竞争窗口自适应EDCA机制。仿真实验表明,竞争窗口自适应EDCA机制有效地降低了车用自组网中音频、视频流的传输时延、时延抖动和丢包率,保证了车用自组网传输VoIP、视频会议、音视频流媒体等多媒体业务的服务质量。     

基于延迟快速调度算法的Vo LTE低丢包评估

为了改善VoLTE语音通话时的用户感知,提升VoLTE通话的语音质量,参考延迟快速调度算法,从承载VoLTE语音的相关网元入手,分析其主要功能和相关参数,提出VoLTE语音低丢包问题优化评估研究,建立参数模型,选出最优参数并推广。     

100+ VoIP Calls on 802.11b: The Power of Combining Voice Frame Aggregation and Uplink-Downlink Bandwidth Control in Wireless LANs

The bandwidth efficiency of voice over IP (VoIP) traffic on the IEEE 802.11 WLAN is notoriously low. VoIP over 802.11 incurs high bandwidth cost for voice frame packetization and MAC/PHY framing, which is aggravated by channel access overhead. For instance, 10 calls with the G.729 codec can barely be supported on 802.11b with acceptable QoS - less than 2% efficiency. As WLANs and VoIP services become increasingly widespread, this inefficiency must be overcome. This paper proposes a solution that boosts the efficiency high enough to support a significantly larger number of calls than existing schemes, with fair call quality. The solution comes in two parts: adaptive frame aggregation and uplink/downlink bandwidth equalization. The former reduces the absolute number of MAC frames according to the link congestion level, and the latter balances the bandwidth usage between the access point (AP) and wireless stations. When used in combination, they yield superior performance, for instance, supporting more than 100 VoIP calls over an IEEE 802.11b link. The authors demonstrate the performance of the proposed approach through extensive simulation, and validate the simulation through analysis.