VoIP reliability: a service provider's perspective

Voice over IP services offer important revenue-generating opportunities, as well as many technical challenges in providing high-quality services. Users have come to expect highly available telecommunications services with high-quality voice. Service providers need reliable high-performance networks to meet user expectations, and must be able to guarantee performance and reliability to their customers. In converged voice and data networks, the network infrastructure must deliver very high quality and availability for some customer needs, while also providing low-cost high-capacity bandwidth for other needs. The use of quality of service mechanisms to provide prioritization for various traffic types is a key element needed for voice and data network convergence. However, it is not sufficient if the underlying networks are unreliable. The focus of this article is to address the reliability aspects of VoIP services, including the underlying IP networks.     

Critical issues of onboard switching in DVB-S/RCS broadband satellite networks

The next generation of broadband satellite networks is challenged to accommodate multimedia services while concurrently integrating with terrestrial IP networks. With IP applications dominating the Internet, carrying IP traffic over the satellite has been under intensive study. Originally developed to bring digital television home through satellites, the DVB-S and DVB-RCS standards empower interactive satellite communications with economical standardized satellite terminals. Furthermore, onboard switching technology is increasingly gaining attention, due to optimized bandwidth usage, fully meshed network topology through one satellite hop, and quality of service guarantee. This article investigates the onboard switching technologies in DVB-S/DVB-RCS broadband satellite networks. Aside from the network system infrastructure and switch hardware architecture, the QoS mechanisms supported by the switch onboard the satellite are discussed in depth.     

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     

VoIP Performance over HSPA with Different VoIP Clients

High Speed Packet Access (HSPA) Holma H, Toskala A (in HSDPA/HSUPA for UMTS, 2006) is expected to provide enough bandwidth for voice over IP (VoIP) service. In this article we assess the performance of VoIP over HSPA with different VoIP clients and voice codecs. The simulations results show that VoIP can have a good voice quality over HSPA if a proper VoIP client and codec is used. However it is possible that the delay can increase with early HSPA implementations (mobile, network).     

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.     

Quality of experience (QoE) driven adaptation scheme for?voice/video over IP

Network quality of service (NQoS) of IP networks is unpredictable and impacts the quality of networked multimedia services. Adaptive voice and video schemes are therefore vital for the provision of voice over IP (VoIP) services for optimised quality of experience (QoE). Traditional adaptation schemes based on NQoS do not take perceived quality into consideration even though the user is the best judge of quality. Additionally, uncertainties inherent in NQoS parameter measurements make the design of adaptation schemes difficult and their performance suboptimal. This paper presents a QoE-driven adaptation scheme for voice and video over IP to solve the optimisation problem to provide optimal QoE for networked voice and video applications. The adaptive VoIP architecture was implemented and tested both in NS2 and in an Open IMS Core network to allow extensive simulation and test-bed evaluation. Results show that the scheme was optimally responsive to available network bandwidth and congestion for both voice and video and optimised delivered QoE for different network conditions, and is friendly to TCP traffic.     

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.     

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.     

Voice communications over zigbee networks

This article provides an overview of ZigBee-enabled wireless networks and discusses the feasibility of supporting voice communications over ZigBee networks. We begin by providing an overview of the ZigBee technology followed by an evaluation of voice quality and performance over such an impoverished wireless channel. Two types of voice communications, namely full-duplex voice over IP (VoIP) and half-duplex push-to-talk (PTT) are considered. Voice quality of VoIP is measured using the R-factor [1] (a well known objective speech quality metric). The quality of PTT, however, is evaluated based on packet-loss rate, delay, and jitter. The simulation results demonstrate that a low-power, low-rate wireless sensor network can support a limited range of voice services.     

Rate adaptive speech coding for universal multimedia access

This article reviews state-of-the-art in transport adaptation techniques for mobile networks. It discusses the mechanisms for rate adaptation to combat quality degradations of speech caused by the radio links. It begins with a review of dynamic schemes for adaptation of speech encoders in cellular networks where we observe two distinct approaches to rate adaptation: network controlled and source controlled. The issues associated with adaptive voice over IP (VoIP) mechanisms are considered next. Here, the encoder detects some form of network congestion to judge how to behave itself for the good of the network. It is noted that this altruistic behavior will only benefit coordinated IP networks such as private intranets and its application to the public Internet is improbable.     

Improving quality of experience in wireless VoIP through novel call scheduling

In the last few years we have experienced a dramatic increase in the use of IP networks for voice applications (VoIP) over wireless networks due to increased bandwidth availability and enhanced device capabilities. Since demand often exceeds available capacity, Call Admission Control mechanisms are in place to prevent the uncontrolled usage of bandwidth. Through the use of an intermediary gateway, VoIP calls are in many cases terminated to a normal landline or cellphone; the capacity of such a gateway is also a finite resource since the number of users can vary significantly as many are mobile. In this article we propose an enhanced scheme that aims to manage access to the lines available so that they are used in a fair manner and utilized to the highest degree possible. This management is facilitated by enhancing a proxy implementation with a number of call scheduling policies. The ability to satisfy pending call requests as soon as lines become available, results in increased user service satisfaction. Moreover, it increases line utilization which is crucial from an economic viewpoint. The ultimate goal is to improve Quality of Experience which is deemed as highly important especially considering that wireless network users experience opportunistic and intermittent connectivity.     

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.     

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.     

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.     

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。     

Voice over Internet protocol (VoIP)

During the Internet stock bubble, articles in the trade press frequently said that, in the near future, telephone traffic would be just another application running over the Internet. Such statements gloss over many engineering details that preclude voice from being just another Internet application. This paper deals with the technical aspects of implementing voice over Internet protocol (VoIP), without speculating on the timetable for convergence. First, the paper discusses the factors involved in making a high-quality VoIP call and the engineering tradeoffs that must be made between delay and the efficient use of bandwidth. After a discussion of codec selection and the delay budget, there is a discussion of various techniques to achieve network quality of service. Since call setup is very important, the paper next gives an overview of several VoIP call signaling protocols, including H.323, SIP, MGCP, and Megaco/H.248. There is a section on telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls     

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.     

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。     

Toward an all-IP-based UMTS system architecture

Looking into the future, two main drivers for the mobile telecommunications market can be identified: third-generation mobile systems (e.g., UMTS) and the Internet (e.g., the introduction of IP technologies like voice/multimedia over IP in mobile networks). UMTS is seen as the enabler of wireless multimedia applications and portability of a personalized service set across network/terminal boundaries, as defined within the virtual home environment (VHE) system concept. In light of these evolutions, this article investigates the impact of the evolution toward an all-IP UMTS network architecture on the UMTS service architecture, which is based on the VHE concept. The article discusses two possible scenarios for supporting VoIP services in the UMTS service architecture and analyzes their applicability in an all-IP-based UMTS network. The first is based on the traditional centralized IN service architecture. The second proposes a new decentralized architecture based on direct control of VoIP call control equipment by open service architecture interfaces     

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.