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     

IP电话的若干网络技术问题

IP电话近年来发展很快。文章讨论了IP电话的若干重要的网络技术问题,如:在接收端对话音分组进行还原时采取的策略;对不同的分组给予不同的服务以及IP电话的应用层协议等。     

MPLS在移动IP网络中的应用

移动通信网络和IP网络的融合推动了电信产业的发展。在一个统一的IP网络平台上传输话音、数据、视频、图像将是大势所趋。目前针对移动用户的服务质量(QoS)要求,也正面临着很大的挑战。在移动IP网络中,由于用户的频繁移动使得端到端的QoS保证和移动切换(handoff)管理更加复杂。多协议标签交换(MPLS)技术作为借鉴了面向连接网络的优越性和实现流量工程的重要技术,为网络提供了较好的QoS保证。文章介绍了MPLS技术是如何应用在移动IP网络中,解决了诸如认证、QoS、流量工程等问题。     

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.     

Evaluation of header compression schemes for IP-based wireless access systems

For mobile IP-based telephony (voice over IP) and IP-based real-time multimedia over cellular radio systems, an economically viable solution is needed. It is an absolute requirement that, for example, the 60-octet IPv6/UPD/RTP headers on IP telephony packets be reduced in size to conserve bandwidth in the radio spectrum. We evaluate the performance of two header compression schemes, based on RFCs 2508 and 3095, under the conditions of cellular radio access technology. The results presented in this article refer to voice and Web browsing traffic and are based on the implementation of compression/decompression algorithms for the aforementioned standards. We find that RFC 3095 performs significantly better if used for mobile communications.     

Vulnerability analysis and best practices for adopting IP telephony in critical infrastructure sectors

IP telephony has been rapidly introduced to replace the traditional circuit switched infrastructure for telephony services. This change has had an enormous impact on critical-infrastructure (CI) sectors, which are expected to become increasingly dependent on IP telephony services. Reliable and secure telephony service is a key concern confronting most organizations in the critical-infrastructure sector today. With the proliferation of voice over IP (VoIP) services in these organizations, it is important for them to understand the security vulnerabilities and come up with a set of best practices during the evolution of the IP telephony services. This article outlines the potential security issues faced by CI sectors as they transform their traditional phone systems into VoIP systems. Vulnerability analyses are conducted to understand the impact of VoIP security challenges in the new convergent network paradigm. The most common security measures are analyzed to identify their strengths and limitations in combating these new security challenges. A set of recommendations and best practices are offered to address the key issues of VoIP security as IP telephony is being introduced into critical infrastructure.     

局域网IP电话系统

局域网ＩＰ电话系统是在基于以太网技术的局域风上传输ＩＰ语音的系统。局域网ＩＰ电话系统不仅能传输话音,而且能提供多种智能服务,完全可以取代传统的ＰＢＸ系统。如果把ＩＰ电话在骨干网（如Ｉｎｔｅｒｎｅｔ）上的应用和局域网上的应用结合起来,将会使ＩＰ电话的应用更为完善,并能在骨干网和局域网上同时体现ＩＰ电话的优点。     

Internet telephony: going like crazy

Interet telephony is possibly the fastest-growing part of communications today. This article discusses what exactly it is, who needs it, and how it works. Internet telephony, or voice over Internet protocol (VoIP), is the provision of phone service over the Internet. But in sharp contrast with conventional telephony, it carries voice traffic as data packets over a packet-switched data network instead of as a synchronous stream of binary data over a circuit-switched, time-division multiplexed (TDM) voice network. There are some substantial benefits (as well as some sticky problems) to the scheme, which is why companies and individuals are finding it increasingly attractive     

Mechanisms and hierarchical topology for fast handover in wireless IP networks

We propose a mechanism to perform fast handover in IP-based wireless networks for real-time applications such as Internet telephony and videoconferencing. Our proposal is designed to reestablish the communication session traffic flow quickly and to minimize the service disruption delay that occurs during mobile IP handover. In this scheme, we propose two different mechanisms to handle micromobility and inter-subdomain mobility, respectively. Micromobility handover handles movements within the same subdomain. Inter-subdomain handover supports handovers between two adjacent subdomains. The reason for having several subdomains is to deploy the network over a wider area to keep the mobile user in the same network for as long as possible. The novelty of the scheme is to retransmit the buffered packets during micromobility handover and to use multicasting to reestablish traffic flow during inter-subdomain movement. The entire scheme is performed within a hierarchical topology based on next-generation IP networks. We analyze both micromobility and inter-subdomain mobility handovers, and display simulation results for both voice and video over IP for micromobility handover.     

A multiplexing scheme for H.323 voice-over-IP applications

Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed.     

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     

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 (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     

UMTS-to-IP QoS mapping for voice and video telephony services

Quality of service mapping between UMTS services and IP transport is crucial for maintaining a suitable end-to-end delay for emerging UMTS multimedia telephony. However, due to incompatibilities in QoS classifications within these two technologies, straightforward mapping is impossible and current proposals within the 3GPP could lead to unpredictable and undesirable behavior for certain services. In this article we focus on two very important UMTS services, voice and video telephony, and establish the QoS issues that exist for these services. We then propose a refined QoS mapping that differentiates between the transmission of voice and video-telephony and a weighted fair queuing scheduler to schedule the transmissions. Through a simulation study, we show the effect on the queuing delays of both traffic types when their WFQ weights vary and then derive an optimal weight that provides the best overall delays for multimedia telephony services.     

IP Telephony Solutions for the Customer Premises

Recent developments in IP telephony equipment have made it possible to provide a fully converged communications solution. This paper specifically concentrates on the customer-premises-based IP telephony solutions for the business market and in particular it looks at the next generation in IP-based telephones and private branch exchanges (PBXs). This also leads to the importance both of the network infrastructure, including local and wide-area networks, and of providing end-to-end quality of service to support voice as a real-time service. Suppliers are striving to develop the systems that will exploit the new converged world while at the same time deliver the reliability and functionality that customers demand. It is currently very much an early adopter market for these IP telephony solutions and experience drawn from delivering trials to external customers as well as the internal pilot work is used to provide a practical snapshots of today's solutions.     

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。     

Multiservice optical network: main concepts and first achievementsof the ROM program

With the tremendous introduction of internet protocol (IP) applications, the quality-of-service (QoS) becomes more and more an emergent issue. Concrete solutions can be adopted (IP/ATM/SONET/WDM) opening the way to new types of applications (interactive applications through the exploitation of voice and video) in a short-term approach. However, all the telecommunication community tries to provide new solutions offering capacity and flexibility in a simpler manner. In this paper, we present the concepts of a multiservice optical network studied in the framework of a French Research Program. The QoS could be offered through the combined exploitation of electronic memories in the edges and optical resources in the core of the optical network and through the coexistence of different types of connections. In particular, the traffic shaping in the edges is highlighted through simulation and demonstrates the real impact of this function to maintain the logical performance at its highest level. To propose concrete solutions for its implementation, two network scenarios are proposed. The first one, for the backbone, exhibits a novel optical packet switching architecture taking benefit of the massive presence of wavelengths to solve the contention. The second one, for the metro, shows a second optical packet switching architecture really adapted to the cost constraints (upgradability, compactness, granularity)     

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.     

NGN的IP承载方案

NGN是完全基于数据网络、特别是分组网络（IP网络）的IP承载网来实现语音业务及其他增值业务的开放平台.分析了NGN基于IP的承载方案：NGN对承载网的要求、NGN的承载方式、QoS保证及国内的著名应用介绍等.     

Analysis of a polling system for telephony traffic with application to wireless LANs

Recently, polling has been included as a resource sharing mechanism in the medium access control (MAC) protocol of several communication systems, such as the IEEE 802.11 wireless local area network, primarily to support real-time traffic. Furthermore, to allow these communication systems to support multimedia traffic, the polling scheme often coexists with other MAC schemes such as random access. Motivated by these systems, we develop a model for a polling system with vacations, where the vacations represent the time periods in which the resource sharing mechanism used is a non-polling mode. The real-time traffic served by the polling mode in our study is telephony. We use an on-off Markov modulated fluid (MMF) model to characterize telephony sources. Our analytical study and a counterpart validating simulation study show the following. Since voice codec rates are much smaller than link transmission rates, the queueing delay that arises from waiting for a poll dominates the total delay experienced by a voice packet. To keep delays low, the number of telephone calls that can be admitted must be chosen carefully according to delay tolerance, loss tolerance, codec rates, protocol overheads and the amount of bandwidth allocated to the polling mode. The effect of statistical multiplexing gain obtained by exploiting the on-off characteristics of telephony traffic is more noticeable when the impact of polling overhead is small.