VoIP mobility in IP/cellular network internetworking

This article explores VoIP mobility in the context of IP and cellular networks interworking. ITU-T Rec. H.323 gateways provide the interconnection between IP networks and switched circuit networks. They allow a call originating from an SCN phone to be transmitted over an IP network to an H.323 terminal, or bridged to another SCN phone. While H.323 provides interoperability with other SCN terminals, the major efforts have been focused on IP/wired SCN (PSTN, ISDN, etc.) interworking. In this article we discuss the challenges associated with the interworking between IP networks and cellular networks through H.323 gateways, and propose an innovative approach using the existing call transfer supplementary service to provide VoIP mobility in the H.323 IP telephony networks. The proposed approach uses existing components in the H.323 standard, thereby allowing VoIP mobility service in hybrid IP/cellular networks to be a value-added feature in the existing H.323-compliant Internet telephony systems     

VoIP信令协议SIP和H.323的比较

VoIP系统凭借其低廉的话费和较好的语音质量,已经成为重要的电信业务,并有取代传统长途业务的趋势.许多组织研究并制定了IP网络上呼叫的协议标准,但有两种IP电话信令和控制标准最具有影响力.一种是ITU推荐的H.323协议,另一种是IETF的SIP.这两种协议代表了解决同一问题的两种不同的方法:H.323是信令基于ISDN Q.931和早期推荐的H系列协议的传统的电路交换的方法,而SIP是一种支持基于HTTP的IP网络的超轻量协议标准.本文,我们主要针对SIP和H.323的体系结构,可靠性,复杂性,可扩展性,可伸缩性以及支持业务类型方面进行比较.     

Extending Global IP Connectivity for Ad Hoc Networks

Ad hoc networks have thus far been regarded as stand-alone networks without assumed connectivity to wired IP networks and the Internet. With wireless broadband communications and portable devices with appropriate CPU, memory and battery performance, ad hoc connectivity will become more feasible and demand for global connectivity through ad hoc networking is likely to rapidly grow. In this paper we propose an algorithm and describe a developed prototype for connectivity between an ad hoc network running the ad hoc on-demand distance-vector protocol and a wired IP network where mobile IP is used for mobility management. Implementation issues and performance metrics are also discussed.     

Application versus network layer multicasting in ad hoc networks: the ALMA routing protocol

Application layer multicasting has emerged as an appealing alternative to network layer multicasting in wireline networks. Here, we examine the suitability of application layer multicast in ad hoc networks. To this effect, we propose a flexible receiver-driven overlay multicast protocol which we call Application Layer Multicast Algorithm (ALMA). ALMA constructs an overlay multicast tree in a dynamic, decentralized and incremental way. First, ALMA is receiver-driven: the member nodes find their connections according to their needs. Second, it is flexible, and thus, it can satisfy the performance goals and the needs of a wide range of applications. Third, it is highly adaptive: it reconfigures the tree in response to mobility or congestion. In addition, our protocol has the advantages of an application layer protocol: (a) simplicity of deployment, (b) independence from lower layer protocols, and (c) capability of exploiting features such as reliability and security that may be provided by the lower layers. Through extensive simulations, we show that ALMA performs favorably against the currently best application layer and network layer protocols. In more detail, we find that ALMA performs significantly better than ODMRP, a network layer, for small group sizes. We conclude that the application layer approach and ALMA seem very promising for ad hoc multicasting.     

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     

IP multicast for mobile hosts

We present alternative designs for efficiently supporting multicast for mobile hosts on the Internet. Methods for separately supporting multicasting and mobility along with their possible interactions are briefly described, and then various solutions to the combined problem are explored. We examine three different multicast delivery mechanisms and compare them based on their efficiency and impact on host protocol software. We describe proposals for integrating multicasting and mobility in the Internet architecture. We first present IP extensions for host mobility and other extensions for multicasting. We then examine local multicasting mechanisms, focusing on a group membership protocol that is optimized for wireless point-to-point links. Next, we examine the problems of sending and receiving multicast datagrams in a wide area network. For multicast reception, we describe three alternative proposals and compare them by examining both their applicability and their performance, as well as possible tradeoffs among the two     

Network mobility protocol for vehicular ad hoc networks

The goal of the network mobility management is to effectively reduce the complexity of handoff procedure and keep mobile devices connecting to the Internet. When users are going to leave an old subnet and enter a new subnet, the handoff procedure is executed on the mobile device, and it may break off the real‐time service, such as VoIP or mobile TV, because of the mobility of mobile devices. Because a vehicle is moving so fast, it may cause the handoff and packet loss problems. Both of the problems will lower down the throughput of the network. To overcome these problems, we propose a novel network mobility protocol for vehicular ad hoc networks. In a highway, because every car is moving in a fixed direction at a high speed, a car adopting our protocol can acquire an IP address from the vehicular ad hoc network through the vehicle‐to‐vehicle communications. The vehicle can rely on the assistance of a front vehicle to execute the prehandoff procedure, or it may acquire a new IP address through multihop relays from the car on the lanes of the same or opposite direction and thus may reduce the handoff delay and maintain the connectivity to the Internet. Simulation results have shown that the proposed scheme is able to reduce both the handoff delay and packet loss rate. Copyright © 2013 John Wiley & Sons, Ltd.     

Secure interconnection protocol for integrated Internet and ad hoc networks

Integration of ad hoc networks with the Internet provides global Internet connectivity for ad hoc hosts through the coordination of mobile IP and ad hoc protocols. In a pure ad hoc network, it is difficult to establish trust relationship between two ad hoc hosts due to lack of infrastructure or centralized administration. In this paper, an infrastructure‐supported and distributed authentication protocol is proposed to enhance trust relationships amongst ad hoc hosts. In addition, an effective secure routing protocol (SRP) is discussed to protect the multi‐hop route for Internet and ad hoc communication. In the integrated ad hoc networks with Internet accessibility, the ad hoc routing security deployed with the help of infrastructure has a fundamental impact on ad hoc hosts in term of Internet access, integrity, and authentication. Both analysis and simulation results demonstrate the effectiveness of the proposed security protocol. Copyright © 2007 John Wiley & Sons, Ltd.     

H.323: the multimedia communications standard for local areanetworks

This article provides an overview of ITU-T Recommendation H.323, “Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service.” This recommendation applies to multimedia communications over packet-switched networks, such as Ethernet, which run TCP/IP, IXP/SPX, or other related protocols. In addition to the multimedia terminal, other H.323 components are defined which provide for conference admissions (gatekeeper), multipoint communications (multipoint controller, multipoint processor), and interoperability (gateway) with terminals on other types of networks. H.323 has application to a variety of network media, including local area networks, enterprise networks, metropolitan area networks, wide area networks, dial-up line connections to LANs, and the Internet. This provides the capability to have global multimedia communications from the desktop using existing network infrastructures     

Mobility management alternatives for migration to mobile Internet session-based services

Session-based Internet protocol (IP) applications, such as Internet telephony, are an important component of the emerging mobile Internet. The ubiquitous availability of these services is critical to the success of the mobile Internet. Because all-IP networks will be deployed in phases and current mobile telecommunication systems will be in operation for decades to come, the interworking and migration between current network services and all-IP services is a key problem. In this paper, we address seamless roaming for session initiation protocol-based services across current cellular telecommunication networks and emerging all-IP wireless networks, such as those using third-generation and WiFi networks. We present an abstract mobility model, and map this model to three basic approaches for supporting seamless mobility: a master-slave approach, a federated system, and a unified approach. We discuss the challenges and implementation of an instance of the unified mobility management approach, called the Unified Mobility Manager, and then compare the tradeoffs of the three systems using a comparative performance analysis. We conclude that unified mobility management is most efficient if a great deal of interworking is required, and as more users invoke IP-based services; the federated approach is efficient when a single network technology is dominant and data access is limited, but requires sharing of data across networks; the master-slave approach is the least efficient, but is easy to introduce if the number of network types is small.     

基于RTP协议的IP电话QoS监测及提高策略

介绍了多媒体通信的发展趋势和当前存在的问题,对基于RTP协议的网络电话中音频数据传输技术进行了研究,对影响实时传输质量QoS的典型因素进行了分析。在局域网的环境下进行了语音包分析实验,探讨了基于RTP协议的QoS动态监测方法,并提出可行的改进方案。     

移动IP电话技术

本文根据ITU-T的最新研究动态,介绍了IP电话移动性技术要求,以及基于H.323的移动IP网络结构和协议功能,并给出了漫游和切换信令过程。     

PACMAN: passive autoconfiguration for mobile ad hoc networks

Mobile ad hoc networks (MANETs) enable the communication between mobile nodes via multihop wireless routes without depending on a communication infrastructure. In contrast to infrastructure-based networks, MANET's support autonomous and spontaneous networking and, thus, should be capable of self-organization and -configuration. This paper presents passive autoconfiguration for mobile ad hoc network (PACMAN), a novel approach for the efficient distributed address autoconfiguration of mobile ad hoc networks. Special features of PACMAN are the support for frequent network partitioning and merging, and very low protocol overhead. This is accomplished by using cross-layer information derived from ongoing routing protocol traffic, e.g., address conflicts are detected in a passive manner based on anomalies in routing protocol traffic. Furthermore, PACMAN assigns Internet protocol (IP) addresses in a way that enables their compression, which can significantly reduce the routing protocol overhead. The performance of PACMAN is analyzed in detail based on various simulation results.     

Supporting multicasting in mobile ad‐hoc wireless networks: issues,challenges, and current protocols

The basic philosophy of personal communication services is to provide user‐to‐user, location independent communication services. The emerging group communication wireless applications, such as multipoint data dissemination and multiparty conferencing tools have made the design and development of efficient multicast techniques in mobile ad‐hoc networking environments a necessity and not just a desire. Multicast protocols in mobile ad‐hoc networks have been an area of active research for the past couple of years. This paper summarizes the activities and recent advances in this work‐in‐progress area by identifying the main issues and challenges that multicast protocols are facing in mobile ad‐hoc networking environments, and by surveying several existing multicasting protocols. This article presents a classification of the current multicast protocols, discusses the functionality of the individual existing protocols, and provides a qualitative comparison of their characteristics according to several distinct features and performance parameters. Furthermore, since many of the additional issues and constraints associated with the mobile ad‐hoc networks are due, to a large extent, to the attribute of user mobility, we also present an overview of research and development efforts in the area of group mobility modeling in mobile ad‐hoc networks. Copyright © 2001 John Wiley & Sons, Ltd.     

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     

漫谈VoIP技术发展趋势——H.325:第三代多媒体系统

目前在Internet或IP网络上应用的VoIP技术主要是基于H．323或者SIP开发的。随着技术和需求的发展，VoIP要求能够同时提供话音、数据和视频等多种业务，向下一代网络NGN演进。为了更好地满足NGN的需求，弥补现有系统的不足，ITU提出了下一代多媒体系统H．325协议的概念，它的重点在于实现控制单元和服务单元的分离，更好地支持多种媒体编码协议的互通，提高系统的QoS以及安全性。H．325有望成为下一代VoIP技术的支撑协议。     

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.     

Unified transport layer support for data striping and host mobility

Data striping across multiple network interfaces and its applications to mobile environments was recently investigated (Phatak and Goff, 2002). Therein, a network-layer Internet protocol (IP)-in-IP encapsulation mechanism was proposed to aggregate the bandwidth of multiple network paths by striping a single transport layer [transmission control protocol (TCP)] connection across them. The analysis and experimental results from that study demonstrated fundamental limitations to TCPs performance in such scenarios. In this paper, we therefore propose a method of overcoming these limitations by striping data at the transport layer. For a proof-of-concept demonstration we use a substantially enhanced version of the stream control transmission protocol (SCTP). Our analytical results and experimental data illustrate that there are significant advantages to using a transport protocol with native support for the simultaneous use of multiple network interfaces, as opposed to stretching TCP to a point where it is no longer effective. This work naturally leads to another fundamental issue of end-to-end support for host mobility at the transport layer. Our analysis and results demonstrate that transport layer support for multiple network (IP) interfaces, together with the capability to dynamically add or delete IP addresses can yield the following advantages: higher bandwidth, load balancing, and increased fairness, enhanced reliability, and end-to-end support for host mobility. This is independent of the underlying network-layer and, hence, is applicable to static/wired, as well as wireless/ad hoc environments. The proposed protocol offers a unified solution to both data striping across multiple networks interfaces, as well as end-to-end mobility support.     

Quality of service for packet telephony over mobile ad hoc networks

IP telephony over mobile ad hoc networks is a topic of emerging interest in the research arena as one of the paths toward the fixed-mobile convergence in telecommunications networks. To investigate the performance characteristics of this service, we propose a complete system architecture, which includes a MAC protocol, a routing protocol, and the treatment of voice packets. The telephone system is analyzed in the case of point-to-point calls inside the ad hoc network, and the end-to-end performance is assessed in terms of the percentage of blocked and dropped calls, packet loss and packet delay. The analysis takes into account network scalability by investigating how; the size of the multihop ad hoc network impacts the quality of service. Moreover, the synthetic mean opinion score of the telephone service is evaluated according to the ITU-T E-model.     

Wireless Ad Hoc Multicast Routing with Mobility Prediction

An ad hoc wireless network is an infrastructureless network composed of mobile hosts. The primary concerns in ad hoc networks are bandwidth limitations and unpredictable topology changes. Thus, efficient utilization of routing packets and immediate recovery of route breaks are critical in routing and multicasting protocols. A multicast scheme, On-Demand Multicast Routing Protocol (ODMRP), has been recently proposed for mobile ad hoc networks. ODMRP is a reactive (on-demand) protocol that delivers packets to destination(s) on a mesh topology using scoped flooding of data. We can apply a number of enhancements to improve the performance of ODMRP. In this paper, we propose a mobility prediction scheme to help select stable routes and to perform rerouting in anticipation of topology changes. We also introduce techniques to improve transmission reliability and eliminate route acquisition latency. The impact of our improvements is evaluated via simulation.