IEEE 802.11e无线局域网中VoIP容量分析及改进

IEEE 802.11e提供了对不同的业务流进行优先级区分的机制,提高了无线局域网（WLAN）内VoIP等实时业务相对于其他业务流的优先性,从而在一定程度上保障其QoS要求,但并没有对WLAN中移动节点（MH）和接入节点（AP）进行优先级的区分,导致AP处上下行业务流量的不平衡,并造成瓶颈效应,限制了VoIP容量。针对这一问题,在对WLAN中VoIP进行容量分析的基础上,提出一种改进AP信道接入能力的策略。通过对AP中队列长度的检测来判断网络拥塞程度,从而动态调整AP的仲裁帧间间隔（AIFS）,以达到根据网络拥塞状况来动态调整AP的优先级,弱化AP的瓶颈效应对VoIP容量制约的目的。仿真结果表明,所提出的改进方案能够有效地提高WLAN中的VoIP容量。     

支持QoS的IEEE 802.11 EDCA性能研究

随着无线网络应用的不断发展,为了适应网络中不同类型业务的区分服务需要,IEEE 802.11工作组在IEEE 802.11 DCF(distributed coordination function)的基础上推出了支持QoS的IEEE 802.11 EDCA协议.针对EDCA(enhanced distributed channel access)协议的优先级区分信道接入特性,提出了一种基于Markov链的协议性能模型分析方法.与已有文献不同,该模型分析同时包含了EDCA主要的3个关键区分信道接入机制:Wmin/Wmax,AIFS(arbitration inter-frame space),TXOP(transmission opportunity).通过模型分析,获得了EDCA协议各优先级接入的传输吞吐量、信道接入延迟、数据丢失率等性能分析.不仅分析了饱和情形下的EDCA性能,而且还对非饱和情形下的EDCA性能进行了分析.仿真验证结果表明,模型分析结果和仿真结果取得了很好的一致性.利用给出的模型分析方法,提出了一种基于TXOP动态调整的D-TXOP(dynamic TXOP)接纳控制算法.算法根据网络中业务流的QoS要求,在动态调整各优先级的TXOP参数设置的基础上,对网络中新到业务流进行接纳控制,达到提高网络的业务承载能力的目的.     

提高WLAN语音传输容量的MAC协议研究

在传统的无线局域网(Wireless Local Area Network,WLAN)环境下,VoIP(Voice over Internet Protocol)业务的传输容量严重受限,大大限制了无线VoIP的发展.通过分析与仿真,发现接入点(Access Point,AP)为限制无线局域网中语音传输容量的瓶颈,从而有针对性地提出了动态调整AP竞争信道优先级的方法.该方法在AP队列过长时提高AP发送数据的优先级,简单易行,并可有效缓解AP的瓶颈效应.多场景下的仿真结果表明,无线局域网中语音传输容量可提高20%,而信道接入延时可降低60%左右.     

基于WLAN的VoIP嵌入式终端的设计

VOIP是一种新型的语音通信技术,由于其以Internet作为载体受到了广泛的关注.利用已有的WLAN网络实现无线的VoIP通话,可以提高网络资源的利用率并降低了通话成本.提出了一种基于SIP协议把VoIP和WLAN技术相融合的嵌入式无线终端方案,进行了硬件和软件设计,为系统开发提供了可行的实现途径.     

基于SIP的嵌入式VoWiFi电话终端设计

在802.11a/b/g无线局域网(WLAN)上采用VoIP技术实现VoWiFi电话具有较好的应用前景.本文在ARM嵌入式系统硬件平台上,采用WiFi无线模块,并应用开源的Linux操作系统移植开发SIP协议,以ARM9嵌入式系统开发板为核心扩展各功能模块设计实现了VoWiFi电话终端.完成VoWiFi电话终端语音通话功能.     

WinCE上SIP移动终端的设计与实现

VoIP(Voice Over IP)和WLAN(Wireless LAN)两种主导技术的结合,产生了一种新的应用,VoIP on WLAN(无线局域网上的VoIP).本文在系统分析WLAN和VoIP相关技术的基础上,设计实现了一个基于嵌入式平台WinCE的VoIP无线终端软件,并给出了测试结果和评价.     

基于CAPWAP协议的集中式无线局域网架构

在WLAN日益普及,组网规模不断扩大的趋势下,如何对数量较多的无线接入点（AP）进行管理,监测和控制的问题,对WLAN的管理提出了挑战。本文讨论一种新出现的WLAN架构：集中式WLAN,以及实现这一架构的协议：CAPWAP（无线接入点控制）。CAPWAP协议规定了接入控制器（AC）对无线终端（WTP）的管理,该协议使通过IP协议进行多个无线终端的网络配置成为可能。     

WLAN上跑语音——谈VoIP基于无线局域网的应用

虽然 WLAN(无线局域网)由于安全标准的不确定性,严重阻碍了其发展的步伐,但利用基于 WLAN 的 VoIP 来代替或补充目前现有的电话系统,如通过笔记本电脑、PDA 或者新型的 Wi-Fi 电话与 WLAN 语音服务器连接,将成为了促使 WLAN 广泛应用的亮点之一, 换占之,无线网络上跑语音即将成为现实。作为一种成熟的网络通信技术,VoIP 在有线网络已逐渐开始广泛应用,而基于 WLAN 的 VoIP 技术以与之相补充显示了其市场潜力。虽然目前在无线网络上实现 VoIP 还面临着无线吞吐量的波动、热点间漫游、Wi- Fi 电话终端缺乏等问题的困扰,但基于 WLAN 的 VoIP 应用代表了市场发展的趋势。可软硬件实现基于 WLAN 的 VoIP 实现可通过软硬来实现,其工作方式分为两个部分。一是基于 SpectraLink、Symbol Technologies Inc．Cisco 等公司的     

共享WiFi系统中MAC层快速切换方案的研究与实现

基于IEEE 802.11协议的无线局域网(WLAN)能够为无线多媒体应用提供足够的带宽,由于无线接入点AP(Access Point)发射功率受限,其服务覆盖范围也因此受限,这使得终端在移动过程中需要在不同AP之间频繁地切换,由此带来的时延严重影响了无线多媒体应用的服务质量。因此,提出一种基于信道掩码和优先级的选择扫描策略和选择性邻居图缓存算法,增大了命中缓存的概率,减少了扫描的信道数量和系统信令开销。仿真结果表明,该切换方案可以有效降低移动节点MN(Mobile Node)在共享WiFi(Wireless Fidelity)系统中不同AP间的切换时延,满足了无线多媒体业务(如VoIP)的传输要求。     

一种饱和状态下EDCA机制新的性能分析模型

针对支持QoS的IEEE 802.11e协议的优先级区分信道接入特性,提出了一种新的性能分析模型方法。考虑在饱和状态下,该模型分析包含了EDCA主要三个方面:AIFS(arbitribution inter-frame space)区分机制,信道和帧拥塞机制以及不同的竞争窗口大小。在不同的传输负荷,节点数和网络结构条件下。模型分析的值与仿真值在各个优先级接入的传输吞吐量,信道接入延迟和数据丢失率等性能有很好的一致性。     

一种提高无线网络语音通话数量的方法

语音通话(Voice over IP)已经成为无线局域网中的一种重要的应用,并且近年来得到了越来越广泛的关注.分析了无线局域网所能支持的语音通话的最大数量,并在网络仿真器NS2上建立了IEEE 802.11语音通话的仿真模型根据仿真分析结果,提出一种自适应传输间隔接入控制方法,提高了IEEE 802.11无线局域网支持语音通话的数量.在自适应传输间隔接入控制方法中,基站(Base Station)通过监测网络冲突概率,自适应的改变网络中语音通话的传输间隔,使网络始终工作在非饱和状态.仿真实验结果表明,提出的白适应传输间隔接入控制方法不仅能够保证IEEE 802.11无线局域网中语音通话的质量,还可以增加网络所能支持语音通话的数量.     

基于802.11E EDCA的VOIP性能分析

随着无线局城网(wLAN)的流行,在无线局城网上进行实时多媒体应用受到了广泛重视。传统无线局城网没有提供QoS保证,无法胜任实时流的传输。EDCA实现了区分服务,改善了多媒体流的传输质量。本文首先简要介绍了EDCA机制,然后通过理论和模拟分析了基于EDCA无线局域网的VoIP性能,并分析了协议参数设置对VoIP性能的影响。     

A comprehensive survey on scheduler for VoIP over WLAN

Voice over Internet Protocol (VoIP) technology has observed rapid growth in the world of telecommunications. VoIP offers high-rate voice services at low cost with good flexibility, typically in a Wireless Local Area Network (WLAN). In a voice conversation, each client works either as a sender or a receiver depending on the direction of traffic flow over the network. A VoIP technologically requires high throughput, less packet loss and a high fairness index over the network. The packets of VoIP streaming may experience drops because of competition among the different kinds of traffic flow over the network. A VoIP application is also sensitive to delays and requires voice packets to arrive on time from the sender to the receiver without any delay over a WLAN. To date, scheduling of VoIP traffic is still an unresolved problem. The objectives of this survey paper are to discuss fundamental principles of VoIP-related schedulers and identify current scheduler issues. This survey paper also identifies the importance of the scheduling techniques over WLANs. Related research work for real-time applications specifically for VoIP will also be highlighted.     

VoWLAN中基于TDMA的QoS机制的研究

针对无线局域网语音应用缺乏可靠服务质量的情况,提出了一种基于时分多址接入的服务质量新机制。该机制采用了IEEE802.15.3无线个域网的中心调度方式,使用时分多址接入机制控制节点接入信道。并在平均接入时延、时延抖动和分组丢失率三方面对IEEE802.11b、IEEE802.11e和新方案三者的性能进行了对比。仿真结果表明,IEEE802.11b性能最差,新机制的性能在平均接入时延和分组丢失率方面比IEEE802.11e分别改善了20%和50%左右。     

QoS-aware path switching for VoIP traffic using SCTP

Voice over Internet protocol (VoIP) has been a prevalent multimedia service nowadays. It allows us to transmit voice data over IP networks. However, quality of service (QoS) is a major challenge to VoIP services. It must provide similar quality to traditional public switched telephone network or cellular phone services. Therefore, QoS related protocols have become important for real-time applications. Multi-protocol label switch (MPLS) is one of the important techniques to improve the network performance from QoS point of view. It employs label swapping to speed up packet forwarding. However, when a large number of users utilize VoIP services, the network congestion issue still exists. It causes delay, jitter and packet loss that affect VoIP QoS. In this paper, we propose a QoS-aware path switching strategy by using stream control transmission protocol (SCTP) in MPLS network to improve the VoIP traffic. This was done by employing SCTP selective acknowledgment mechanism to report the transmission parameters of primary path and to determine the criteria to switch to backup path. Simulation results show significant improvement in VoIP QoS.     

SIP的VoWLAN通信平台设计与实现

如何将会话初始化协议（SIP）与现有的通信网络有机结合,提供安全可靠的数据及通信服务已成为当今的热点问题。VoIP应用也受到业界的持续关注。安全问题一直都是企业实施VoIP的一个阻碍。提出了一个基于SIP的VoWLAN通信平台,将各种语音服务构建于无线局域网之上。利用虚拟专用网(VPN)、数据加密技术、VLAN和防火墙等必要安全技术和策略,应对在系统中的安全威胁,实现了通话质量可靠、安全性高的企业级VoIP无线网络架构。描述了该系统的设计和实现过程,讨论了其中的关键技术。     

Enhancing VoIP service for ubiquitous communication in a campus WLAN with partial coverage

The IEEE 802.11 WLAN technology has become the de facto standard for wireless Internet access. The spotty coverage of WLAN access points, however, confines the applicability of many real-time services such as VoIP within the boundary of the WLAN service area. In this paper, we investigate the problem of enhancing VoIP service for ubiquitous communication in a WLAN with spotty service area. We consider a university campus that has an established infrastructure for supporting SIP-based VoIP service through either wired or wireless data networks. The campus WLAN service does not have 100% full coverage, and hence users cannot make untethered VoIP calls anywhere on campus. The goal of this paper is to overcome the limitations of such “dead spots” for motivating the use of campus IP telephony service. To proceed, we start with two approaches called one-hop extension and dual-mode communication. The first approach uses multi-hop relay to extend the WLAN coverage, while the second approach leverages the availability of dual-mode handsets for ubiquitous voice communication. We implement the two approaches, and evaluate their performance in the campus testbed environment. We find that while the two approaches can effectively allow voice communication in WLAN dead spots, they have one common problem as the potential lack of support for voice call continuity that can cause degradation of the speech quality to an active call. We adopt a cross-layer solution based on signal processing algorithms to address the problem, thus achieving seamless voice call continuity while enabling ubiquitous voice communication on campus. Testbed evaluation shows promising results for future research along the proposed direction.     

Design and implementation of QoS-provisioning system for voice over IP

In this paper, we address issues in implementing voice over IP (VoIP) services in packet switching networks. VoIP has been identified as a critical real-time application in the network QoS research community and has been implemented in commercial products. To provide competent quality of service for VoIP systems comparable to traditional PSTN systems, a call admission control (CAC) mechanism has to be introduced to prevent packet loss and over-queuing. Several well-designed CAC mechanisms, such as the site-utilization-based CAC-and the link-utilization-based CAC mechanisms have been in place. However, the existing commercial VoIP systems have not been able to adequately apply and support these CAC mechanisms and, hence, have been unable to provide QoS guarantees to voice over IP networks. We have designed and implemented a QoS-provisioning system that can be seamlessly integrated with the existing VoIP systems to overcome their weakness in offering QoS guarantees. A practical implementation of our QoS-provisioning system has been realized.