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.     

VoWLAN通信平台的设计与实现

VoWLAN也叫VoWiFi或者WiFi VoIP。它是基于无线网络技术和VoIP网络,是两者的有机结合。即是通过WLAN提供VoIP业务,使得终端用户通过WLAN拨打IP电话成为现实。本文提出了在基于Linux操作系统的SIP应用服务器及VoIP网关中,如何通过SIP信令和传统的PSTN数据通信线路与无线网络无缝连接方案,从而实现IP网络与传统电话间的实时语音通信、电话会议、语音信箱、视频通信、短消息、数据传输等业务。本设计已成功应用于某企业的实时语音通信平台,获得良好的效果。     

VoIP — The Access Dimension

For a given voice quality, VoIP does not offer a particularly efficient transport solution. This becomes a concern in any bandwidth-constrained part of a network, in particular, a copper-based, access network. VoIP access systems need to be carefully designed if they are to offer an acceptable combination of quality and efficiency. This paper identifies and quantifies the various issues which need to be addressed in undertaking such a design, and suggests possible solutions. The challenges of distributing VoIP within the customer premises are also addressed, with specific and different solutions being proposed for the business and residential environments.     

Seven myths about voice over IP

Voice over Internet Protocol (VoIP) is one of the fastest growing and most misunderstood technologies in the world at the moment. Confusion, outdated beliefs and urban mythology reign over such simple issues as how it works, the quality of the calls, and, of course, how much it costs. This paper examines seven existing myths about VoIP. These myths are that: (1) VoIP is free; (2) the only difference between VoIP and regular telephony is the price; (3) the quality of service isn't an issue nowadays because there's plenty of bandwidth in the network; (4) VoIP can't replace regular telephony because it still cannot guarantee quality of service; (5) VoIP is just another data application; (6) VoIP isn't secure; and (7) a phone is a phone is a phone.     

Designing VoIP session management over interworked WLAN-3G networks

In an all-IP internetworked heterogeneous environment, ongoing VoIP sessions from roaming users will be subject to frequent vertical handoffs across network boundaries. Ensuring uninterrupted service continuity for these handoff calls requires successful session management among the participating access networks. As such, a mobility-aware novel interworking network design (interconnecting UMTS and WLAN over an IP-based common platform) [1] is presented in this article that facilitates VoIP session management, including session establishment and seamless session handoff across different networks. For comparison purposes, VoIP session management is evaluated in terms of session establishment, handoff delays, transient packet loss, end-to-end traffic delays, and jitter value for different voice codecs, which demonstrate satisfactory and feasible results. In the event (e.g., network congestion, buffer overflow) that session continuity cannot be guaranteed (also known as outage) across network boundaries, this article proposes an algorithm that compensates the user by reducing the unit service charge of future sessions (governed by the outage period) through a noncooperative game-theory-based pricing mechanism.     

Performance Optimizations for Deploying VoIP Services in Mesh Networks

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients     

An energy‐efficient transmission mechanism for VoIP over IEEE 802.11 WLAN

Voice over IP (VoIP) over WLAN (VoWLAN) is an important application for public and private WLANs. However, VoWLAN systems suffer from several technical challenges such as power consumption of a WLAN station (STA) and service capacity of an access point (AP), making the commercial deployment of a large‐scale VoWLAN service problematic. This study presents a cross‐layer and energy‐efficient mechanism for transmitting VoIP packets over IEEE 802.11 WLAN. The proposed mechanism considers the characteristics of voice packets that can tolerate certain loss, and dynamically disables the medium access control (MAC) layer acknowledgement for voice packets. In doing so, the time and energy consumed to transmit and receive voice packets for an STA can be reduced. Simulation results demonstrate that the mechanism improves the energy efficiency of a VoWLAN STA and WLAN utilization without sacrificing voice qualities. Copyright © 2009 John Wiley & Sons, Ltd.     

An analysis of VoIP service using 1/spl times/EV-DO revision a system

While voice-over-Internet protocol (VoIP) on wireline network is maturing, VoIP on wireless mobile network is still in its infancy. This disparity is due to the fact that the wireline bandwidth is abundant and can be traded off for delay performance and overhead, whereas bandwidth in wireless mobile network is still a scarce resource. With the deployment of 1/spl times/EV-DO revision 0 (DOr0) worldwide, the spectrum efficiency has been significantly improved. However, DOr0 still lacks of features essential for VoIP. For this reason, 1/spl times/EV-DO revision A (DOrA) has been standardized in the 3GPP2 with many improvements favorable for VoIP implementation. In this paper, we identify challenges and explore the feasibility of implementing VoIP using DOrA. We develop both analytical and simulation models to evaluate the VoIP capacity and delay performance over the air interface.     

Assessing the quality of voice communications over Internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.     

面向固定移动融合环境的P2PSIP系统设计

全业务运营是电信市场继语音和宽带接入服务之后的下一个增长点,而基于IP的融合有线网络和无线网络的语音服务则是全业务的重点之一。本文通过分析现有VoIP网络存在的问题以及固定移动融合网络环境下VoIP的特点,提出一种新型双层重叠网架构的P2PSIP架构,并阐述了新型架构的优点及双层重叠网之间的通信机制。新型架构能有效提高系统的安全性、健壮性和用户节点资源利用效率,更好的满足固定移动融合网络环境下VoIP对带宽、网络质量和安全性的要求。     

SIP协议穿透NAT网关的研究

详细分析了SIP协议的IP电话穿透NAT的问题所在。提出了适合于SIP电话终端的穿透方案，通过在SIP电话终端中的具体部署和实现以及测试，验证了电话终端程序穿透NAT方案的可行性，解决了部分NAT穿透的问题。     

VoIP over Wi-Fi Networks: Performance Analysis and Acceleration Algorithms

The expected VoIP call capacity in a one hop IEEE 802.11b network with G.711 voice codec is about 85 simultaneous calls, but the actual observed capacity is only 5 calls even at the highest data rate and under zero loss conditions. In this paper we analyze the reasons behind this inferior performance of VoIP traffic. We also present algorithms at the medium access control layer to improve the observed call capacity. Using ns-2 based simulations, we evaluate the algorithms and show that performance improvements of more than 300% can be achieved. Finally, using a testbed implementation of one of the proposed algorithms, we show its feasibilty in real world VoIP implementations. An earlier version of this paper was published in the IEEE Conference on Broadband Communications, Networks, and Systems (BroadNets) 2007 [1]. This work was supported in part by the National Science Foundation under grants CNS-0519733 and CNS-0721296, and the Georgia Tech Broadband Institute. Yeonsik Jeong was also supported under the Korea Research Foundation Grant KRF-2005-214-D00362 funded by the Korean Government (MOEHRD). This work was performed when Y. Jeong was a research scientist in the GNAN research group at Georgia Institute of Technology.     

基于IPv6的大规模接入汇聚路由器的设想和实现

针对下一代互联网面临的新业务拓展和安全性服务问题,本文综述了对未来网络的可运营、可管理和可控制的综合要求,提出了基于IPv6的大规模接入汇聚路由器(converging router access,ACR)的新概念,并分析了ACR工程实现的若干关键问题.ACR的提出不仅打破了大容量高性能路由器仅在核心骨干网络中使用的传统观念,开辟了大容量高性能路由器新的应用环境,而且为在下一代互联网中开展大规模、高速稳定和安全可靠的宽窄带新业务提供了创新的技术途径.     

vGPRS: A Mechanism for Voice over GPRS

This paper proposes vGPRS, a voice over IP (VoIP) mechanism for general packet radio service (GPRS) network. In this approach, a new network element called VoIP mobile switching center (VMSC) is introduced to replace standard GSM MSC. Both standard GSM and GPRS mobile stations can be used to receive real-time VoIP service, which need not be equipped with the VoIP (i.e., H.323) terminal capabilities. The vGPRS approach is implemented using standard H.323, GPRS, and GSM protocols. Thus, existing GPRS and H.323 network elements are not modified. Furthermore, the message flows for vGPRS registration, call origination, call release and call termination procedures are described to show the feasibility of our vGPRS system.     

Implementing a secure VoIP communication over SIP-based networks

Recent years the Session Initiation Protocol (SIP) is commonly used in establishing Voice over IP (VoIP) calls and has become the centerpiece for most VoIP architecture. As wireless and mobile all-IP networks become prosperous, free VoIP applications are utilized in all places. Consequently, the security VoIP is a crucial requirements for its adoption. Many authentication and key agreement schemes are proposed to protect the SIP messages, however, lacking concrete implementations. The performance of VoIP is critical for users’ impressions. In view of this, this paper studies the performance impact of using key agreements, elliptic curve Diffie–Hellman and elliptic curve Menezes–Qu–Vanstone, for making a SIP-based VoIP call. We evaluate the key agreement cost using spongycastle.jce.provider package in Java running on android-based mobile phones, the effect of using different elliptic curves and analyze the security of both key agreements. Furthermore, we design a practical and efficient authentication mechanism to deploy our VoIP architecture and show that a VoIP call can be established in an acceptable interval. As a result, this paper provides a concrete and feasible architecture to secure a VoIP call.     

VoIP traffic in wireless mesh networks: a MOS‐based routing scheme

Support of Voice over Internet Protocol (VoIP) services in wireless mesh networks requires implementation of efficient policies to support low‐delay data delivery. Multipath routing is typically supported in wireless mesh networks at the network level to provide high fault tolerance and load balancing because links in the proximity of the wireless mesh gateways can be very stressed and overloaded, thus causing scarce performance. As a consequence of using multipath solutions, lower delay and higher throughput can be supported also when a given path is broken because of mobility or bad channel conditions, and alternative routes are available. This can be a relevant improvement especially when assuming that real‐time traffic, such as VoIP, travels into the network. In this paper, we address the problem of Quality of Service (QoS) support in wireless mesh networks and propose a multipath routing strategy that exploits the Mean Opinion Score (MOS) metric to select the most suitable paths for supporting VoIP applications and performing adaptive load balancing among the available paths to equalize network traffic. Performance results assess the effectiveness of the proposed approach when compared with other existing methodologies. Copyright © 2015 John Wiley & Sons, Ltd.     

iVoIP: an intelligent bandwidth management scheme for VoIP in WLANs

Voice over Internet Protocol (VoIP) has been widely used by many mobile consumer devices in IEEE 802.11 wireless local area networks (WLAN) due to its low cost and convenience. However, delays of all VoIP flows dramatically increase when network capacity is approached. Additionally, unfair traffic distribution between downlink and uplink flows in WLANs impacts the perceived VoIP quality. This paper proposes an intelligent bandwidth management scheme for VoIP services (iVoIP) that improves bandwidth utilization and provides fair downlink–uplink channel access. iVoIP is a cross-layer solution which includes two components: (1) iVoIP-Admission Control, which protects the quality of existing flows and increases the utilization of wireless network resources; (2) iVoIP-Fairness scheme, which balances the channel access opportunity between access point (AP) and wireless stations. iVoIP-Admission Control limits the number of VoIP flows based on an estimation of VoIP capacity. iVoIP-Fairness implements a contention window adaptation scheme at AP which uses stereotypes and considers several major quality of service parameters to balance the network access of downlink and uplink flows, respectively. Extensive simulations and real tests have been performed, demonstrating that iVoIP has both very good VoIP capacity estimation and admission control results. Additionally, iVoIP improves the downlink/uplink fairness level in terms of throughput, delay, loss, and VoIP quality.     

在MPLS网络中如何保证VoIP的QoS

在以IP为基础的网络中,IP网络上传送语音(VoIP)成为新一代语音系统的代表,但其业务质量因由IP网络承载而受到限制.多协议标签交换(MPLS)是当前被普遍看好的高速骨干网络技术,通过MPLS,第三层的路由可以得到第二层技术的很好补充,以保证端到端的服务质量(QoS).将MPLS的QoS特性与区分服务(DiffServ)相结合,可实现更优化的网络业务QoS.     

An Intelligent Vertical Handover Scheme for Audio and Video Streaming in Heterogeneous Vehicular Networks

In heterogeneous vehicular networks, the most challenging issue is obtaining an efficient vertical handover during the vehicle roaming process. Efficient network selection process can achieve satisfactory Quality-of-Service for ongoing applications. In this paper, we propose an Intelligent Network Selection (INS) scheme based on maximization scoring function to efficiently rank available wireless network candidates. Three input parameters were utilized to develop a maximization scoring function that collected data from each network candidate during the selection process. These parameters are: Faded Signal-to-Noise Ratio, Residual Channel Capacity, and Connection Life Time. The results show that the proposed INS scheme is more efficient at decreasing handover delays, End-to-End delays for VoIP and Video applications, packet loss ratios as well as increasing the efficiency of network selection processes in comparison with the state of the arts.     

基于VoIP的校园网络管理系统的设计

文章在分析和比较其他VoIP网络管理系统设计方案的基础上,重点论述了VoIP校园网络管理系统的设计方案,阐述了校园网络管理系统主要的实施过程.并提出了采用约定访问速度对流量进行控制,采用排队的技术对拥塞进行处理,并根据校园网络的实际情况给出网络路由的总体设计方案.