QoS enabled voice support in the next generation Internet: issues,existing approaches and challenges

The Internet is under rapid growth and continuous evolution in order to accommodate an increasingly large number of applications with diverse service requirements. In particular, Internet telephony, or voice over IP is one of the most promising services currently being deployed. Besides the potentially significant cost reduction, Internet telephony can offer many new features and easier integration with widely adopted Web-based services. Despite these advantages, there still exist a number of barriers to the widespread deployment of Internet telephony. The most prominent one, however, is how to ensure the QoS needed for voice conversation. The purpose of this article is to survey the state-of-the-art technologies in enabling the QoS support for voice communications in the next-generation Internet. In this article, we first review the existing technologies in supporting voice over IP networks, including the basic mechanisms in the IETF Internet telephony architecture and ITU-T H.323-related Recommendations. We then discuss the IETF QoS framework, specifically the Intserv and Diffserv framework. Finally, we present two leading companies' (Cisco and Lucent) solutions to offering IP telephony services as examples to illustrate how real systems are implemented     

Real-time voice over packet-switched networks

We discuss the architecture and technical viability of transporting real-time voice over packet-switched networks such as the Internet. The value of integrating voice and data networks onto a common platform is well known. The telephony industry has proposed the ATM standard as a means of upgrading the Internet to provide both real-time and data services. In contrast, voice services may be added to traditional IP networks that were originally designed for data transmission alone. We consider the feasibility and expected quality of service of audio applications over IP networks such as the Internet. In particular, we examine possible architectures for voice over IP and discuss measured Internet delay and loss characteristics     

The IETF Internet telephony architecture and protocols

Internet telephony was first used as a simple way to provide point-to-point voice transport between two IP hosts. However, the growing interest in providing integrated voice, data, and video services has caused its scope to be extended. Internet telephony now encompasses a range of services, including not only traditional conferencing, call control, multimedia, and mobility services, but also new ones that integrate Web, e-mail, presence, and instant messaging applications with telephony. Internet telephony and traditional circuit-switched telephony will coexist for quite some time, requiring interworking between the two. In this article we present a suite of protocols, developed in the IETF, which provide a partial solution to this complex problem     

Voice and Internet Multimedia in UMTS Networks

Voice telephony is the predominant service on today's cellular mobile networks, in terms of number of customers, revenues and network usage. However, it is difficult to predict how long this will be the case given the rising demand for new Internet multimedia services. It is therefore essential that 3rd generation (3G) mobile networks support a voice telephony service, but also that these networks are also capable of providing Internet multimedia services using the same technology.This paper provides an overview of how voice telephony is provided in the initial phase of the universal mobile telecommunications system (UMTS). It then describes how this is expected to evolve in later phases — so that voice telephony becomes one of a large number of multimedia services provided from a common Internet protocol-based mobile network.     

Talk about a change [VoIP replacing PSTN]

The rapid rise of Internet telephony services means that existing telecoms networks could soon be redundant. This paper examines the technological and financial issues concerning voice over Internet protocol (VoIP) and the probability of it substantially replacing existing public switched telephone networks (PSTN) within the next few years.     

话音业务的多元化发展

随着电信网络和业务的发展,传统话音业务逐渐从电话网上分离,数据网上传话音进而提上日程。ＩＰ电话标准已然成熟,但相关产品仍缺乏兼容性,话音质量也不够理想;帧中继具有传送话音的巨大潜能,有较强的竞争优势;ＡＴＭ虽然已发展多年,但在传话音方面仍存在许多问题,不能完全尽如人意。作者对话音业务的多元化发展进行了分析和比较。     

Internet telephony in India

Internet telephony is a novel and cheaper method of communication and conducting business over the Internet. The paper presents an overview of Internet telephony, its methods, viz. PC-to-PC, PC-to-telephone, telephone-to-telephone and telephone-to-PC; benefits in cost advantage, simplification, consolidation, higher efficiency and reliability, etc., quality issues, protocols and drivers; challenges and regulatory framework; and status of Internet telephony in Asia Pacific region. Further, highlights its potentiality for India, implications of guidelines of Internet telephony, issues of concern, etc. Concludes that Internet telephony cannot make compromises in voice quality, reliability, scalability and manageability, and work seamlessly with telephone systems all over the world. Internet telephony will prove to be a boon for a price-sensitive market like India and rural telephony will receive an impetus. The Government of India may further deregulate the market and allow phone-to-phone telephony through the Internet and open long distance calling within the country for ISPs to realize “telecom for the common man” or “telecom for all” a reality.     

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.     

The generic architecture for data/internet telephony in Nigeria

Nigerians were yet to get value for their money in telephony services through NITEL and GSM operators in Nigeria. After several years or months of commencement of operations, the service is not only too expensive, limited but inefficient. People because of the inefficiency in the sectors, got to a level where they have to procure sets connected to MTN, Econet and NITEL among others. In spite of this, there is problem of connectivity and reliability. Tariffs for the service are rather too expensive for both local and international calls.

This paper proposes an innovative generic model for delivering voice and data traffics over IP to residential, small offices, medium and large enterprises in Nigeria at moderately reduced price, which cannot break one’s backbone while eliminating the current problems associated with NITEL telephony services. The paper also presents the benefits, the barriers of implementing Internet telephony as well as the Internet telephony implementation options. The cost for implementing the model is outlined in the paper.     

IP语音包的自适应编码和封装算法的研究

IP电话与传统电话相比语音质量较差，其中最主要的原因是因特网的带宽变化较大，导致丢包率较大。该文根据因特网带宽变化的特点提出了1种应用在IP电话网关中的语音自适应编码与封装策略，采用该策略的编码器能根据网络的带宽变化动态调节语音编码速率和语音包封装大小。据此，本文提出了4种算法：一种基于RTP协议语音包丢失率的计算算法、变速率编码算法，不同长度IP语音包的封装算法和根据丢包率来调整编码速率和封装的自适应算法。     

Traffic engineering standards in IP-networks using MPLS

The explosive growth of the Internet over the last few years has made the IP protocol suite the most predominant networking technology. Furthermore, the convergence of voice and data communications over a single network infrastructure is expected to happen over IP-based networks. Traditional IP-networks offer little predictability of service, which is often unacceptable for applications such as telephony, as well as for emerging and future real-time applications such as telemedicine. One of the primary goals of traffic engineering is to enable networks to offer predictable performance. Due to the need for better traffic control by network service providers, there has been considerable activity in the Internet Engineering Task Force to develop standards for traffic engineering in IP-based networks. This article discusses the direction taken by the IETF and some of the recent standardization efforts for traffic engineering using multiprotocol label switching (MPLS). Our primary focus is on the signaling protocols developed for this purpose     

The EMA system: a CTI based e-mail alerting service

The integration of Internet services and telephony services is a new area for the development of telecommunications services. One example is an e-mail alerting service that uses the telephony network for e-mail notification. The EMA system is a computer telephony integration (CTI) application that checks a user's mailbox on the mail server and informs him/her over the phone when new e-mail arrives, eliminating the need for permanent Internet connection. The EMA system has a Web-based interface, enabling the user to configure service parameters. The EMA system is developed as a distributed and concurrent application. It consists of seven modules: the console, Web interface, Web handler, controller, voice machine, database, and mail checker, using communication solutions based on component object model (COM) technology. This article describes the structure of the EMA system, its implementation, and advantages for users     

From phones to football: The changing strategic focus of BT

For over a decade BT has been investing significant sums to shift its focus away from the provision of voice telephony towards the Internet. This shift is epitomised by the widespread availability of broadband and the company’s purchase of the rights to broadcast English Premier League football games. This paper argues that its dominance in the voice telephony market funded its initial expansion into these new markets, and that broadband and sports content are mutually supportive lines of business. The paper also highlights the significant contribution that Openreach makes to the overall profitability of BT, and the challenges that exist as a result.     

Packet telephony support for the IEEE 802.11 wireless LAN

Packet telephony is one of the most promising applications in the Internet. In this paper, we propose a modified MAC protocol supporting voice traffic over the IEEE 802.11 WLAN. The proposed scheme adapts the power-saved mode of the IEEE 802.11 specifications in such a way that it approaches the TDM access mode carrying voice traffic, and is compatible with the IEEE 802.11 standard. Simulation results show that the proposed scheme does not degrade the performance of the IEEE 802.11 WLAN using the DCF and also provides good voice quality     

Voice over Internet protocol and human-assisted e-commerce

By fostering the finalization of open standards and the convergence of voice, video, and data, the Internet protocol provides an ideal driver for the definition of the infrastructure for new multimedia and advanced communications applications. Voice over IP represents not only the chance to achieve cost-effective real-time voice communication over IP-based networks, but also the opportunity to build an integrated and open communications service delivery infrastructure. Developments of Web-based information systems and IP telephony in order to enable future e-commerce applications are summarized     

Unified Mobility Manager: enabling efficient SIP/UMTS mobile network control

Internet telephony is viewed as an emerging technology not only for wireline networks, but also for third-generation wireless networks. Although IP end to end is considered the ultimate approach to future wireless voice services, there is still a long way to go before IP voice packets can be effectively transported over the air. Therefore, Internet telephony and today's circuit-switched wireless network will coexist for years to come, and it is essential to effectively perform interworking between these networks. This article proposes the Unified Mobility Manager (UMM) that achieves efficient interworking between traditional wireless networks and Internet telephony networks. The main characteristic of the UMM is that it combines UMTS HLR and SIP proxy functionality in one logical entity, which helps eliminate the performance degradation due to interworking between SIP and UMTS. This article identifies seven potential network architectures with and without the UMM and with varying degrees of IP penetration in the wireless core networks, and performs comparative analysis in terms of their call setup signaling latency. Our performance results show that for SIP originated calls, the architecture with the UMM can achieve better performance than existing UMTS networks without the UMM. Our results further show that when the backbone network is fully IP-enabled, dramatic performance gains can be accomplished with the UMM for PSTN originated calls as well as for SIP originated calls. The article also demonstrates that the UMM allows graceful migration from today's circuit-switched wireless networks to hybrid SIP/circuit-switched wireless networks, and toward the IMS architecture for all-IP UMTS networks in the future.     

Internet protocol support for telephony

In this paper, we consider the evolution of telephone networks from time-division multiplexing circuit switching to packet switching and, in particular, to packet switching-based on Internet Protocol (IP-supported telephony). We analyze IP-supported telephony design solutions by proposing a layered reference model in which each layer is associated to a subset of the functions that support telephony. We use the reference model to establish a terminology and a framework for the comparison of the design solutions. We group the design solutions in scenarios and compare them in terms of the reference model proposed. We then focus on IP telephony, in which IP is used in telephone company networks, and on Internet telephony, in which the Internet is used to support telephony. We show that they both can be seen as implementations of the same architecture, which consists of a set of components, associated to functions, and of the interactions among these components. We then consider the issue of voice-data integration and analyze the variety of design solutions that can be adopted to integrate voice and data.     

Telephony's next act

This paper discusses the impending merger of traditional telephone networks and the Internet via a new technology, called voice over Internet Protocol (VoIP). A key issue that must be addressed is how to merge the respective databases of the two communication networks. With a new type of data record, called ENUM (electronic number), the legacy information of both telephony and the Internet can be accommodated and allow a phone number to be converted to an Internet address. DNS capabilities, however, first need to be strengthened to ensure they can meet the demands of ENUM and other new network technologies.     

Internet telephony: services, technical challenges, and products

The rapid proliferation of the Internet has given rise to a strong interest in carrying telephony over the Internet. Because the Internet supports data communications, a range of other services can be bundled together with Internet telephony. The Internet, however, was designed for non-real-time data communications, and hence it poses several technical challenges that must be overcome before the Internet can be successfully used for carrying telephone services. This article discusses new services we can expect from Internet telephony, the technical challenges and solutions, and the emerging products that promise to support Internet telephony     

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