A Context-Aware Network Equipment for Dynamic Adaptation of Multimedia Services in Wireless Networks

B3G (Beyond 3rd Generation) networks is the next step in the wireless world. It will enable the delivery of services to end-users, on different underlying wireless access networks and customised to users’ context. Furthermore, seamless mobility between networks and switch of devices will be handled in a transparent way. In this paper, a Context-Aware Network Equipment (CANE) is introduced, that is able to dynamically adapt multimedia services according to the users’ context. The term context includes the users’ device, the users’ preferences, the network conditions as well as the service provider adaptation policies. As a proof of concept, an implementation of a CANE is given for a 802.11 network use-case. An evaluation of the prototype as well as simulations of the proposed solution, both for an audio streaming service and a video streaming service, are presented. The results show that adapting multimedia services in case of network overload can enable to maintain an acceptable quality of service delivered to end-users and even to allow more users to enjoy the services, which will not be possible without the CANE.     

SIP-based mobility management in next generation networks [accepted from open call]

The ITU-T definition of next generation networks includes the ability to make use of multiple broadband transport technologies and to support generalized mobility. Next generation networks must integrate several IP-based access technologies in a seamless way. In this article, we first describe the requirements of a mobility management scheme for multimedia real-time communication services; then, we report a survey of the mobility management schemes proposed in the recent literature to perform vertical handovers between heterogeneous networks. Based on this analysis, we propose an application-layer solution for mobility management that is based on the SIP protocol and satisfies the most important requirements for a proper implementation of vertical handovers. We also implemented our proposed solution, testing it in the field, and proving its overall feasibility and its interoperability with different terminals and SIP servers.     

A Robust Seamless Handover Scheme for the Support of Multimedia Services in Heterogeneous Emerging Wireless Networks

With the rapid development of wireless technologies and numerous types of mobile devices, the need to support seamless multimedia services in Mobile and Ubiquitous Computing (MUC) is growing. To support the seamless handover, several mobility protocols such as Mobile IPv6 (MIPv6) (Johnson et al., Mobility Support in IPv6, IETF, RFC 3775, 2004) and fast handover for the MIPv6 (FMIPv6) (Koodli et al. Past handovers for mobile IPv6 (FMIPv6), IETF, RFC 4068, 2005) were developed. However, MIPv6 depreciates the Quality-of-Service (QoS) especially in multimedia service applications because of the long handover latency and packet loss problem. To solve these problems in the MIPv6, FMIPv6 is proposed in the Internet Engineering Task Force (IETF). However, FMIPv6 is not robust for the multimedia services in heterogeneous emerging wireless networks when the MN may move to another visited network in contrast with its anticipation. In MUC, the possibility of service failure is more increased because mobile users can frequently change the access networks according to their mobility in heterogeneous wireless access networks such as 3Generation (3G), Wireless Fidelity (Wi-Fi), Worldwide Interoperability for Microwave Access (WiMax) and Bluetooth co-existed. In this paper, we propose a robust seamless handover scheme for the multimedia services in heterogeneous emerging wireless networks. The proposed scheme reduces the handover latency and handover initiation time when handover may fail through the management of tentative Care-of Addresses (CoAs) that does not require Duplicate Address Detection (DAD). Through performance evaluation, we show that our scheme provides more robust handover mechanism than other scheme such as FMIPv6 for the multimedia services in heterogeneous emerging wireless networks.     

SIP信令与媒体流穿透NAT的实现

从会话初始化协议(SIP)消息的特点出发.基于网络地址转换(NAT)和防火墙自身的角度考虑,提出了一种无需扩展SIP协议的应用层解决方案.该方案通过引入UDP对NAT的简单穿透(STUN)协议,获得IP地址和端口的映射绑定关系,修改SIP和会话描述协议(SDP)消息中的内容来保证通信连接,从而实现了对NAT的穿透.     

Converged multimedia services in emerging Web 2.0 session mobility scenarios

The increasing request for converged multimedia services have motivated relevant standardization efforts, such as the Session Initiation Protocol (SIP) to support session control, mobility, and interoperability in all-IP next generation wireless networks. Notwithstanding the central role of SIP in novel converged multimedia, the potential of SIP-based service composition for the development of new classes of Web 2.0 services able to interoperate with existing HTTP-based services is still widely unexplored. The paper proposes an original solution to improve online user experience by integrating a SIP stack into the Web browser, thus enabling the execution of novel SIP-based applications directly at the client endpoint. In particular, our browser extension coordinates with our novel SIP-based Converged Application Server to enable session mobility and prevent abuses of the services available in the client. Experimental results show that our SIP-based solution is feasible in most common Internet deployment scenarios and enables session mobility with limited management cost.     

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.     

SIP电话的NAT穿越技术研究

Ip网络中网络地址转换器(NAT)的存在给基于会话初始协议(SIP)的IP电话(SIP电话)的大规模应用造成了较大困难,为解决SIP电话的NAT穿越问题,分析了完全圆锥型、地址受限圆锥型、端口受限圆锥型、对称型等NAT的特点,比较了应用层网关、基于用户数据报协议(UDP)的NAT简单穿越、采用中继的NAT穿越、会话边界控制器等各种NAT穿越技术的优缺点.以一典型的SIP电话系统为例,针对其应用需求,提出了一种NAT穿越方案,并给出了通信流程.该NAT穿越方案特别适合于对网络结构无限制,网络设备不改造的应用环境.     

SIP在3G网络中的应用

第三代移动通信系统是能提供多种类型IP数据业务的通信系统.会话起始协议SIP是3G的IP多媒体子系统中提供多媒体业务的核心技术.文章首先介绍了SIP的基本工作原理,然后对3GPP UMTS R5定义的IMS进行了简要描述,最后详细阐述了SIP在IMS提供服务的过程及对漫游用户的处理.     

VoIP系统中NAT穿越技术的研究与实现

基于会话初始化协议（SIP）的VoIP系统在Internet上已经取得了广泛应用,但在目前的实际网络环境中,由于大量NAT设备的存在,使对等网络（P2P,Peer to Peer）之间的呼叫和数据通信难以实现。分析了四种NAT的类型特点,介绍了现有的NAT穿越方法,提出了一种基于STUN与TURN方式相结合的实现各种NAT穿越的VoIP系统设计方案。该方案对SIP信令采用可靠的TCP传输方式,对流媒体数据采取最大交付的UDP传输方式。经过校园网内部之间的网络环境测试,该方案达到了很好的接通率。     

SIP-based vertical handoff between WWANs and WLANs

Future-generation wireless networks have been envisioned as the integration of various wireless access networks, including both wireless wide area networks and wireless local area networks. In such a heterogeneous network environment, seamless mobility support is the basis of providing uninterrupted wireless services to mobile users roaming between various wireless access networks. Because of transparency to lower-layer characteristics, ease of deployment, and greater scalability, the application-layer-based session initiation protocol has been considered the right candidate for handling mobility in heterogeneous wireless networks. However, SIP entails application-layer transport and processing of messages, which may introduce considerable delay. As a case study of the performance of mobility management protocols in the heterogeneous wireless networks, we analyze the delay associated with vertical handoff using SIP in the WLAN-UMTS internetwork. Analytical results show that WLAN-to-UMTS handoff incurs unacceptable delay for supporting real-time multimedia services, and is mainly due to transmission of SIP signaling messages over erroneous and bandwidth-limited wireless links. On the other hand, UMTS-to-WLAN handoff experiences much less delay, mainly contributed by the processing delay of signaling messages at the WLAN gateways and servers. While the former case requires the deployment of soft handoff techniques to reduce the delay, faster servers and more efficient host configuration mechanisms can do the job in the latter case.     

Moving toward seamless mobility: state of the art and emerging aspects in standardization bodies

The challenge to provide seamless mobility in the near future emerges as a key topic in various standardization bodies. This includes first of all the support of seamless handover between homogeneous networks. Distinct technologies—such as IEEE 802.11WLANs (Wi-Fi) and IEEE 802.16 MANs WiMAX—have recently augmented such support to existing standards to enable seamless homogeneous handover. Cellular networks, in contrast, already included this inherently from the start. Currently considerable effort goes into coupling of different radio access technologies. Therefore, the second key topic in standardization is seamless heterogeneous handovers. IEEE, IETF, as well as 3GPP consider different approaches toward architectures and protocols enabling seamless mobility management. In this work, we discuss recent and on-going standardization activities within IEEE, IETF, and 3GPP toward seamless homogeneous as well as heterogeneous mobility support.     

A middleware architecture supporting seamless and secure multimedia services across an intertechnology radio access network - seamless content delivery in the future mobile internet

This article presents a middleware architecture to support multimedia services across intertechnology radio access networks in a secure and seamless manner. The proposed architecture uses the media-independent handover framework, where the handover decision function is based on triggering/collecting statistics from the physical, network, and application layers so that an ongoing multimedia session (video) can be transferred seamlessly and securely (using context transfer) across intertechnology radio access networks. Simulation results show that when a vertical handover is based on the proposed MIH framework (including the context transfer of AAA information), handover latency is reduced by 38 percent during WiFi to UMTS handover and 20 percent during UMTS to WiFi handover compared to a non-MIH-based handover scheme.     

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。     

Deployment of IP multimedia streaming services in third-generation mobile networks

In this article, an end-to-end quality of service framework for streaming services in 3G mobile networks is considered. Under this scenario, the interaction between UMTS and IETF's protocols and mechanisms for a streaming session is analyzed. By signaling flowcharts, it is shown that both groups of protocols and mechanisms can co-operate to provide seamless end-to-end real-time services. Specifically, the article proposes to make the IP multimedia subsystem aware of the real time streaming protocol, in order to extend its control from SIP to RTSP-based services, such as multimedia streaming services. Supported by this proposed framework, provisioning of audio streaming services over 3G mobile networks is also outlined.     

Handoff with DSP Support: Enabling Seamless Voice Communications across Heterogeneous Telephony Systems on Dual-Mode Mobile Devices

In this paper we investigate the problem of voice communications across heterogeneous telephony systems on dual-mode (WiFi and GSM) mobile devices. Since GSM is a circuit-switched telephony system, existing solutions that are based on packet-switched network protocols cannot be used. We show in this paper that an enabling technology for seamless voice communications across circuit-switched and packet-switched telephony systems is the support of digital signal processing (DSP) techniques during handoffs. To substantiate our argument, we start with a framework based on the Session Initiation Protocol (SIP) for vertical handoffs on dual-mode mobile devices. We then identify the key obstacle in achieving seamless handoffs across circuit-switched and packet-switched systems, and explain why DSP support is necessary in this context. We propose a solution that incorporates time alignment and time scaling algorithms during handoffs for supporting seamless voice communications across heterogeneous telephony systems. We conduct testbed experiments using a GSM-WiFi dual-mode notebook and evaluate the quality of speech when the call is migrated from WiFi to GSM networks. Evaluation results show that such a cross-disciplinary solution involving signal processing and networking can effectively support seamless voice communications across heterogeneous telephony systems.     

A novel scheme for mobility management in heterogeneous wireless networks

In this paper, we present a mobility management scheme for real-time multimedia sessions over heterogeneous wireless networks. Most approaches in the current literature use Mobile IP (MIP) or Session Initiation Protocol (SIP) to maintain real-time sessions during mobility. In this paper, we analyze MIP and SIP in terms of the mobility rate, packet loss and packet overheads in the user plane and propose a mechanism by which the network can choose the optimum protocol for mobility management. We perform the analysis for constant bit rate (CBR) as well as for variable bit rate (VBR) traffic. We show that for CBR traffic, the proposed mechanism leads to 12–35% improvement in the system capacity, while for VBR traffic, capacity improvements ranging from about 35–50% can be obtained. Our proposed approach and the analysis are applicable to handovers between different IP domains both in homogeneous as well as in heterogeneous wireless networks.     

Detection and resolution of feature interactions in IP multimedia subsystem

The IP Multimedia Subsystem (IMS) enables the delivery of multimedia services through different access networks. 3GPP adopted the Session Initiation Protocol (SIP) as the IMS signalling protocol. However, IMS interaction with these services faces the ‘feature interaction’ problem encountered in traditional telephony networks. This problem occurs as features invoked during a session behave correctly when processed separately from each other, but not when running together. Although much research work has explored managing feature interactions, applying these solutions in IMS remains challenging and further investigation is needed. This paper aims to analyze the feature interaction issue in IMS and to propose a new solution. We defined a SIP‐based algorithm and the associated mechanisms to enhance IMS service invocation. This algorithm is the core of the Service Broker, a new functional entity in charge of managing feature interaction. We validated our proposal through a performance evaluation and prototyped the Service Broker on an open source IMS platform. Copyright © 2008 John Wiley & Sons, Ltd.     

一种基于SIP网络的微移动方案

首先介绍会话发起协议(SIP)的发展状况,SIP是IETF制定的建立对等IP通信的协议,其功能是控制会话的建立、维持和终止。SIP本身只能传送会话控制信息,它必须与其他协议配合才能提供完整的通信控制功能。未来全IP综合网络环境下的数据业务将成为最主要业务,全IP网络面临服务质量(QoS)保证、移动性支持和其他综合性业务等关键问题。针对域内切换中的微移动性问题,重点分析一种基于SIP网络环境的层次性移动SIP(HMSIP)方案,最后论述该方案在为SIP网络提供QoS保证方面的可行性。     

MobileNAT: A New Technique for Mobility Across Heterogeneous Address Spaces

We propose a new network layer mobility architecture called Mobile NAT to efficiently support micro and macro-mobility in and across heterogeneous address spaces common in emerging public networks. The key ideas in this architecture are as follows: (1) Use of two IP addresses – an invariant virtual IP address for host identification at the application layer and an actual routable address at the network layer that changes due to mobility. Since physical address has routing significance only within a domain, it can be a private address and therefore, does not deplete the public IP address resource. (2) New DHCP enhancements to distribute the two addresses. (3) A new signaling element called Mobility Manager (MM) that uses Middlebox Communication (MIDCOM) framework to signal the changes in packet processing rules to the Network Address Translators (NATs) in the event of node mobility. Our proposal does not require any modifications to the access networks and can seamlessly co-exist with the existing Mobile IP mechanisms and therefore, can be used to provide seamless mobility across heterogeneous wireline and wireless networks. We report implementation details of a subset of our ideas in a testbed with Windows XP clients and Linux based NATs. Milind M. Buddhikot is a Member of Technical Staff in the Center for Networking Research at Lucent Bell Labs, Holmdel, NJ. Milind holds a Doctor of Science (D. Sc.) in computer science (July 1998) from Washington University in St. Louis, and a Master of Technology (M.Tech.) in communication engineering (December 1988) from the Indian Institute of Technology (I.I.T), Bombay. His current research interests are in the areas of systems and protocols for integrated public wireless networks, authentication and dynamic key exchange protocols, Voice-over-IP (VOIP) networks, and sensor and ad-hoc networks. He has authored over 26 research papers and 9 patent submissions in the areas of design of multimedia systems and protocols, layer-4 packet classification, MPLS path routing, authentication and dynamic key exchange, and 802.11/3G integration. Milind currently serves as the Associate Editor of the IEEE/ACM Transactions on Networking. He also served as a co-guest-editor of IEEE Network magazine’s March 2001 Special issue on Fast IP Packet forwarding and Classification for Next Generation Internet Services. He has served in the capacity of a tutorial chair for IEEE LCN 94, 95, as a publicity chair for NOSSDAV97, and as a program committee member for MMCN 2001, 2003, IEEE ICNP2002, 2003 and IEEE LCN 93-2000 conferences. Adiseshu Hari received a Doctorate in Computer Science from Washington University in St. Louis in 1999, and has since been at Bell Labs, Lucent Technologies, where he’s worked on projects related to network design and optimization. He’s currently looking at design issues in large scale SIP based VoIP networks. Kundan N. Singh received a B.E.(Hons) degree in Computer Science from Birla Institute of Technology and Science in India, M.S. in Computer Science from Columbia University, and is continuing his studies towards Ph.D. degree in the same field at Columbia University in New York City. As a research assistant in the Internet Real-Time Lab at Columbia University, he is doing research on Internet telephony, SIP-H.323 signaling gateway, unified messaging systems and multimedia conferencing. Scott Miller is the Director of the High Speed Mobile Data Research department at Bell Labs in Holmdel, New Jersey. He has B.S. and M.S. degrees in electrical engineering from Cooper Union in New York City. His current research involves the integration of 802.11 and 3G wireless data service and the related mobile networking issues concerning seamless mobility, authentication, security, roaming, and accounting. Prior to his work on 802.11/3G integration, Scott has led several systems research efforts in wireless applications, implementing novel systems for wireless messaging, speech-driven directory services, wireless instant messaging, carrier-based content billing, and multi-media content adaptation.This revised version was published online in August 2005 with a corrected cover date.     

An Empirical Analysis of Handoff Performance for SIP,Mobile IP,and SCTP Protocols

Over the last decade, we have witnessed a growing interest in the design and deployment of various network architectures and protocols aimed at supporting mobile users as they move across different types of networks. One of the goals of these emerging network solutions is to provide uninterrupted, seamless connectivity to mobile users giving them the ability to access information anywhere, anytime. Handoff management, an important component of mobility management, is crucial in enabling such seamless mobility across heterogeneous network infrastructures. In this work, we investigate the handoff performance of three of the most widely used mobility protocols namely, Mobile IP, Session Initiation Protocol (SIP), and Stream Control Transmission Protocol (SCTP). Our empirical handoff tests were executed on an actual heterogeneous network testbed consisting of wired, wireless local area, and cellular networks using performance metrics such as handoff delay and handoff signaling time. Our empirical results reveal that Mobile IP yields the highest handoff delay among the three mobility protocols. In addition, we also found that SIP and SCTP yield 33 and 55% lower handoff delays respectively compared to Mobile IP.