Voice over the internet: A tutorial discussing problems and solutions associated with alternative transport

This article provides a tutorial overview of voice over the Internet, examining the effects of moving voice traffic over the packet switched Internet and comparing this with the effects of moving voice over the more traditional circuit-switched telephone system. The emphasis of this document is on areas of concern to a backbone service provider implementing Voice over IP (VoIP). We begin by providing overviews of the Plain Old Telephone Service (POTS) and VoIP. We then discuss techniques service providers can use to help preserve service quality on their VoIP networks. Next, we briefly discuss Voice over ATM (VoATM) as an alternative to VoIP. Finally, we offer some conclusions.     

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.     

Approach for voice quality and throughput estimation in wireless convergent networks

A common communications convergence scenario which is being adopted in personal communications relates to the combination of wireless and cellular networks by the use of multimode terminals. Since most of the wireless networks were initially dimensioned only for data communications, this paper shows how voice over wireless LAN dimensioning could be addressed under the optimal network throughput and the perspective of voice quality, using a simple approach. The maximum number of simultaneous users resulting from throughput is limited by the collisions taking place in the shared medium with the statistical contention protocol. The voice quality is conditioned by the delay and the packet loss in the contention protocol. Both approaches are analyzed within the scope of the voice codecs commonly used in voice over wireless LANs, to conclude that voice dimensioning based on network throughput and voice quality show complementary results. Additionally the use of low rate codecs in voice over wireless LANs is advantageous for the network performance point of view but may produce poor voice quality results. Mid range codecs like G729 could represent a trade-off for quality throughput. For these reasons, voice quality and wireless network throughput have to be taken into account in the network admission control, design and deployment to ensure a satisfactory user experience. The impact of handoff interval of wireless convergent networks on the conversation quality need also be assessed for a proper network design.     

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。     

3G手机VoIP话音的QoS实现技术研究

在对3G手机VoIP话音QoS的主要实现技术进行分析的基础上,提出了3G手机VoIP话音QoS新的实现技术。文中通过对实时传输控制协议（RTCP协议）的详细研究,同时根据3G系统无线信道的具体特点,说明了实时传输控制协议运用于3G手机VoIP话音的QoS控制中的缺陷,并阐述了相应的控制解决方法。在基于Android的3G智能手机的VoIP客户端软件中,综合运用VoIP话音QoS的主要成熟实现技术,同时结合文中提出的VoIP话音QoS的解决思路,实现了对VoIP话音的QoS的控制。基于Android的3G智能手机的VolP客户端软件通过在不同的网络环境条件下的测试,VoIP话音质量良好,说明文中提出的3G手机VoIP话音QoS新的实现技术具有一定的实用价值。     

Voice over IP service and performance in satellite networks

Voice over IP services have emerged as a low-cost alternative to PSTN voice service, and an attractive solution for voice/data integration in public and private networks. Satellite systems, as an integral part of the global communications infrastructure, already have an increasing portion of their capacities used to carry data packets, and with their global coverage and reach to remote areas are well positioned to enable growth of VoIP services. For VoIP over satellite, several issues need to be addressed. These include transmission and quality of service issues, as well as service-related issues such as service creation and customization, support of IN and supplementary services, and seamless integration with the PSTN. While the service-related aspects of VoIP are common to both terrestrial and satellite networks, transport-related issues are different. This article reports performance results of laboratory experiments for evaluating VoIP over satellite under different link and traffic conditions     

Variable bit rate VOiP in IEEE 802.11e wireless LANs [medium access control protocols for wireless LANs]

WLANs have become a ubiquitous networking technology deployed everywhere. Meanwhile. VoIP is one popular application and a viable alternative to traditional telephony systems due to its cost efficiency. VoIP over WLAN (VoWLAN) has been emerging as an infrastructure to provide low-cost wireless voice services. However, VoWLAN poses significant challenges due to the characteristics of contention-based protocols and wireless networks. In this article we propose two mechanisms to provide quality of service for variable bit rate VoIP in IEEE 802.11e contention-based channel access WLANs: access time-based admission control and access point dynamic access. Simulation results are conducted to study these schemes.     

Header compressed VoIP in IEEE 802.11

Header compression techniques such as robust header compression can be used to reduce the overhead of IP-based traffic. Voice over IP may replace voice circuits in the next generations of wireless networks, and it is the type of traffic that benefits most from header compression because its packets have small payloads. IEEE 802.11 is a technology that will play an important role in the next generations of wireless networks. The study reported in this article shows that the maximum gain of the RoHC?s U-mode when applied to VoIP over IEEE 802.11 is about 23 percent for medium or better voice quality. Values for the RoHC Umode parameters over IEEE 802.11 are also suggested.     

Perceptual QoS assessment technologies for VoIP

Since quality is not generally guaranteed in an IP network, the proper design and management of networks and/or terminals for high-quality voice over IP services and maintenance of service levels is important. In terms of quality design and management, methodologies for appropriately and effectively evaluating the perceptual QoS of VoIP are indispensable. This article gives an overview of the state of the art of quality assessment technologies for VoIP, including recent work on improving their accuracy.     

Performance evaluation of packet aggregation scheme for VoIP service in wireless multi-hop network

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.     

IP communications in air traffic control (ATC)

Although established some 60 years ago voice communications is still the sole means to separate and guide aircrafts. Voice communication systems are therefore amongst the most critical installations in air traffic control (ATC). Frequentis was the first to introduce PCM (pulse code modulated) based equipment for ATC worldwide and is on the leading edge of steering into the world of packetized voice communications. VoIP is the emerging voice communication technology which has already proven to provide satisfactory service for commercial applications. However, in ATC a number of requirements exceeding commercial applications have to be met. The voice communication system for ATC integrates radio communication as a prime service. Delays generated by the system therefore directly affect the performance on the radio channel and need to be extremly low. Both the radio control (push-to-talk, PTT) and the voice content need to be processed and delivered in a timely manner. SIP (session initiated protocol) based signalling represents a promising approach to tackle the delay problem. In addition, voice communication systems are required to provide high availability figures. VoIP based systems strongly rely on the communications infrastructure as they are distributed by nature. Resilient packet ring structures allow for these high availablity figures for the communications infrastructure.     

Mobility management for VoIP service: Mobile IP vs. SIP

Wireless Internet access has gained significant attention as wireless/mobile communications and networking become widespread. The voice over IP service is likely to play a key role in the convergence of IP-based Internet and mobile cellular networks. We explore different mobility management schemes from the perspective of VoIP services, with a focus on Mobile IP and session initiation protocol. After illustrating the signaling message flows in these two protocols for diverse cases of mobility management, we propose a shadow registration concept to reduce the interdomain handoff (macro-mobility) delay in the VoIP service in mobile environments. We also analytically compute and compare the delay and disruption time for exchanging signaling messages associated with the Mobile IP and SIP-based solutions.     

Seamless SIP-based mobility for multimedia applications

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.     

Voice capacity analysis of WLANS with channel access prioritizing mechanisms

Voice over IP applications in wireless networks have gained increasing popularity in recent years. As a delay-sensitive real-time application, a VoIP flow is usually given higher priority in accessing the shared wireless channel than delay-insensitive non-real-time flows. In contention-based wireless networks two widely used prioritizing MAC mechanisms are class-dependent arbitration interframe space and class-dependent contention window. In this article we propose an analytical model to evaluate the effect of the two mechanisms on voice capacity (the maximum number of two-way voice flow pairs supportable) of ad hoc mode and infrastructure mode wireless LANs. We show that the AIFS mechanism has a relatively strong effect on WLAN voice capacity in the ad hoc mode, but not in the infrastructure mode; and the CW mechanism, when properly configured, has a mild effect on voice capacity in both modes.     

一种新型无线应急指挥通信车的设计方案

无线应急指挥通信车是城市应急指挥通信系统中的重要组成部分,所设计的无线应急指挥通信车不仅可以实现多个不同制式、不同频段的应急指挥通信网络的互连互通,而且还能实现应急指挥通车与远程指挥中心的通信。基于数字信号处理技术和VoIP技术的语音、信令信号的处理是设计方案中的关键技术。     

Implementing VoIP: a voice transmission performance progress report

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.     

一种新型无线应急指挥通信车的设计方案

无线应急指挥通信车是城市应急指挥通信系统中的重要组成部分，本方案所设计的无线应急指挥通信车不仅可以实现多个不同制式、不同频段的应急指挥通信网络的互连互通，还能实现应急指挥通信车与远程指挥中心的通信。基于数字信号处理技术和VoIP技术的语音、信令信号的处理是本设计方案中的关键技术。     

VoIP over DVB-RCS with QoS and bandwidth on demand

Motivated by the need for compliance/interoperability above the satellite-specific layers, this article proposes a consolidated approach for voice over IP over satellite networks based on the ETSI DVB-RCS standard. Voice communication is a real-time service that needs priority over other services in IP environments with limited bandwidth, such as IP satellite networks. Bandwidth utilization in such networks needs to be optimized in order to reduce service costs, and this requires the use of dynamic bandwidth allocation schemes. This article therefore addresses the role of bandwidth on demand in the optimization of bandwidth allocation for VoIP and assesses the impact of BoD mechanisms on voice quality. The trade-off between voice quality and bandwidth efficiency is investigated under different DVB-RCS-specific capacity request/allocation strategies, and it is demonstrated that DVB-RCS provides an efficient platform for integrated support for a variety of VoIP applications over satellite. The main contribution of this article consists of the identification of the mechanisms capable of responding to the key challenges raised by the VoIP application in the satellite environment.     

An enhanced SIP proxy server for wireless VoIP in wireless mesh networks

The wireless mesh network (WMN) has emerged recently as a promising technology for next-generation wireless networking. In WMNs, it is important to provide high quality multimedia service in a flexible and intelligent manner. To address this issue in this article, we study the Session Initiation Protocol (SIP) for wireless voice over IP (VoIP) applications. Especially, we investigate the technical challenges in WMN VoIP systems and propose a design of an enhanced SIP proxy server to overcome them. An analysis of the signaling process and a study of simulation results have shown the advantages of our proposed approach.     

Multicast Broadcast Services Support in OFDMA-Based WiMAX Systems [Advances in Mobile Multimedia]

Multimedia stream service provided by broadband wireless networks has emerged as an important technology and has attracted much attention. An all-IP network architecture with reliable high-throughput air interface makes orthogonal frequency division multiplexing access (OFDMA)-based mobile worldwide interoperability for microwave access (mobile WiMAX) a viable technology for wireless multimedia services, such as voice over IP (VoIP), mobile TV, and so on. One of the main features in a WiMAX MAC layer is that it can provide'differentiated services among different traffic categories with individual QoS requirements. In this article, we first give an overview of the key aspects of WiMAX and describe multimedia broadcast multicast service (MBMS) architecture of the 3GPP. Then, we propose a multicast and broadcast service (MBS) architecture for WiMAX that is based on MBMS. Moreover, we enhance the MBS architecture for mobile WiMAX to overcome the shortcoming of limited video broadcast performance over the baseline MBS model. We also give examples to demonstrate that the proposed architecture can support better mobility and offer higher power efficiency.