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.     

SIP and H.323 — Interworking VoIP Networks

H.323 is currently the industry's predominant standard for offering multimedia services over IP networks, although the session initiation protocol (SIP) continues to generate significant interest in the market-place for rendezvous-based real-time applications. With its greater maturity, H.323-based equipment continues to be deployed in operational networks. However, with the advent of SIP-based third generation mobile networks and the continued development of voice over cable TV networks based upon SIP, significant deployments of SIP-based networks are on the horizon. Interworking between SIP and H.323-based equipment is of significant importance to BT and its Joint Venture partners. This paper highlights some of the issues surrounding protocol interworking between the H.323 and SIP protocol sets.It can be shown that, while possible, interworking inevitably results in sub-optimal operation and a loss of features. Thus, as interworking is inevitably going to be required for a long time to come, and the result is less than perfect, the standards bodies should be encouraged to develop service-enabling extensions to their protocols that are compatible between both protocol suites wherever possible.     

基于SIP的VoIP系统设计与实现

近年来,SIP作为VoIP的相关协议得到了越来越广泛的应用和越来越多厂商的支持。文章提出了设计和实现基于SIP的VoIP系统的方案,该系统是一个softphone,支持音频和视频,使位于不同物理位置的网络用户可以“面对面”地进行交流。     

一种SIP的NAT穿越机制

针对SIP音视频通信应用的快速发展与IPv4地址资源不足的矛盾,提出了NAT穿越机制.现有的NAT穿透方案或者扩展性不好,或者未采用标准的协议栈,或者对移动性支持较差.文中提出了一种新的SIP协议穿越NAT机制(NATUT),采用使客户端主动修改SIP协议域值为NAT映射后的公网地址的方式,实现了SIP协议的通畅穿越NAT设备.测试结果表明,该机制能较好地解决了扩展性等方面的问题.     

基于SIP的MPEG-4视频电话中媒体传输模块的设计

详细阐述了基于SIP的MPEG4视频电话中媒体传输模块的设计和实现方案,论述了MPEG4网络传输关键技术和解决方案,并提出了一个基于RTP/SIP的反馈拥塞控制模型.该设计和解决方案目前已应用于基于嵌入式硬件的视频电话系统开发中,完全能够满足该系统对于多路音视频实时传输的通信需求.     

Efficient Architecture and Handoff Strategy used for VoIP Sessions in SIP Based Wireless Networks

In the near future, the Internet is likely to become an All-IP network that provides various multimedia services over wireless networks. Although the earliest VoIP applications did not consider the end-node mobility, researchers have attempted to support mobility in current VoIP protocols, such as Session Initial Protocol (SIP)-based mobility. The SIP-based mobility is considered because it can readily support mobility. However, calling disruptions may occur in traditional SIP mid-call terminal mobility because handoff procedure may be required, depending on the implementation and the real network deployment considerations. In any case, issues in the combined SIP/RSVP for guaranteeing QoS of VoIP service under mobile environment are also considered to be crucial. Therefore, this study describes the solutions by devising novel hierarchy network architecture. Also, the mechanisms including help with neighboring users in adjacent cells and the third party call control to overcome those issues are included. The simulation results indicate that the proposed technique is practical and better executive than conventional schemes.     

Deployment issues of CCS links on self-healing rings

Fiber-based self-healing rings (SHRs) are being deployed for trunking purposes for voice/data/video services because of their high reliability and survivability. To reduce deployment and maintenance costs of provisioning different facilities for these services and common channel signaling (CCS) networks, network planners have begun to deploy CCS network signaling links on SHRs. This paper examines some issues of deploying A/B/C/D link sets on SHRs     

Ontology mapping for the interoperability problem in network management

Interoperability between different network management domains, heterogeneous devices, and various management systems is one of the main requirements for managing complex enterprise services. While substantial advances have been made in low-level device and data interoperability using common data formats and specifications such as simple network management protocol's (SNMP's) SMI and TMF's SID, various interoperability issues including semantic interoperability offer interesting research challenges. While semantic interoperability is a difficult problem in its own right, the semantic web that incorporates intelligent agents necessitates an interoperability solution requiring agents to communicate unambiguously and reason intelligently to perform cooperative management tasks. Agents need a formal representation of knowledge; an ontology is capable of modeling the rich semantics of the managed environment (and especially, relationships between managed entities) so that agents can act on them. This paper presents an ontology-driven approach for solving the semantic interoperability problem in the management of enterprise services, illustrated here with a router configuration management application.     

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。     

移动互联网WebRTC及相关技术

针对Web实时通信（WebRTC）技术,提出一种WebRTC实时通信服务的改进设计方案.方案在客户端侧、服务端侧、客户端与服务端之间的通信等部分对WebRTC进行了改进.改进方案采用了HTML5 WebSocket技术、媒体协商与合成技术、网络地址转换（NAT）/防火墙穿越技术、基于会话发起协议（SIP）栈的信令交互技术、P2P通信安全技术,较好地解决了跨浏览器运行、对客户端侧处理能力的过分依赖和带宽消耗等问题,提高了系统的可扩展性.     

SIP终端语音聊天的设计与实现

SIP协议是多媒体通信网的控制协议，在分组交换网中，它提供基本的呼叫控制，负责建立、修改和终止多媒体（话音、数据、视频等）会话等应用。概述了SIP终端软件的设计思想，并详细介绍了SIP终端软件中语音聊天功能的关键技术，从宏观和微观2个角度阐述了语音聊天功能的实现过程，构造了系统测试环境。SIP终端软件实现了语音聊天功能，具有良好的性能。     

Experimenting with Feature Interaction Management in SIP Environment

This article reports architectural aspects of a solution for detecting and resolving feature interactions (FI) in SIP-based IP telephony architectures. The solution takes into account the special context of SIP that permits end user programmability, which means the possibility for end users to design their own tailored services and personalize them as much as they like. Programmability renders more frequent the so called multi-component FI situations, where the conflicting services reside in different network components. This type of FI is the more complicated one. The authors describe how the different components of the presented SIP architecture operate together in order to run services and avoid this type of interactions. A prototype of the solution has been developed.     

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.     

基于VoIP技术视频会议系统研究与设计

为了实现视频会议的IP网络与传统的PSTN网络互通,由于现有的SmartMeeting系统是一个纯粹的IP网络视频会议,这样就不能够利用传统的PSTN网络下的电话进行视频会议,故该系统采用增加VoIP网关进行对H．232协议和SIP协议做了相应的协议装换的分析,在现有SmartMeeting系统的基础上对IP网络与PSTN网络进行互通功能实现进行详细的研究与设计。系统是在VC＋＋6．0平台下开发,采用C／S模式实现的。     

Signaling for Multimedia Conferencing in Stand-Alone Mobile Ad Hoc Networks

Mobile ad hoc networks (MANETs) are infrastructure-less and can be set up anywhere, anytime. They can host a wide range of applications in rescue operations, military, private, and commercial settings. Multimedia conferencing is the basis of a wealth of "killer” applications that can be deployed in MANETs. Some examples are audio/video conferencing, multiplayer games, and online public debating. Signaling is the nerve center of multimedia conferences—it establishes, modifies, and tears down conferences. This paper focuses on signaling for multimedia conferences in MANETs. We review the state of the art and propose a novel architecture based on application-level clusters. Our validation employed SIP as the implementation technology and OPNET as our simulation tool. Our clusters are constructed dynamically and the nodes that act as cluster heads are elected based on their capabilities. The capabilities are published and discovered using a simple application-level protocol. The architectural principles and the clustering operations are discussed. Our SIP-based implementation is also presented along with the performance evaluation.     

SIP视频监控客户端在Linux平台的实现

设计与实现了以AVS算法为视频标准,在PC机上的Linux系统为平台的视频监控客户端.利用SIP协议作为控制协议,RTP协议作为传输协议,来接收网络上来自视频监控系统服务器端摄像头采集信号后进行编码工作的视频码流数据.然后将接收到的视频码流进行解码并且在屏幕上的多路画面显示.最后利用QT进行界面GUI设计.该客户端支持多平台移植,对视频监控嵌入式发展有积极的推动意义.     

Implementation of a broadband integrated services hybrid network

The issues involved in the design of a broadband integrated services digital network (B-ISDN), that utilizes optical fibers in the subscriber loop, are considered. A hybrid scheme that provides integrated access to ISDN and broadband services by delivering voice/data signals in a digital format while using an analog format for full-motion video services to subscribers is proposed. The discussion covers planning for a B-ISDN, the system concept, implementation, the loop fiber network, subscriber equipment, and cost comparison and analysis     

基于SIP的H.264视频电话终端设计

考虑到实现的难易程度、生产成本和升级更新等因素,介绍了一种基于SIP的H.264视频电话终端的设计和实现方法。主要讨论了采用高速数字处理芯片TMS320DM643完成H.264编解码和利用ARM9芯片S3C2410实现SIP协议。给出具体的软、硬件设计方案,对研制的样机测试表明该视频电话终端能进行语音和视频信息的实时通信,遵循开放协议,具有良好的兼容性。     

SIP的扩展在IMS QoS优化中的应用研究

基于SIP(会话初始协议)的IMS(IP多媒体子系统)的产生与发展,使得固网与移动网络的融合成为可能.为这种新型的网络提供多媒体业务的QoS(服务质量)保障是一个关键问题.根据SIP的简单、易扩展性,文章提出了一种利用SIP的扩展采改善IMS QoS的方案.     

Fast Handoff in SIP-Based Next Generation Mobile Networks

It is commonly held that next generation mobile systems will be developed on the Internet in combination with diverse access technologies, as the future network architecture will be the coming together of various overlapping wireless access networks. Integrating various wireless networks in future heterogeneous networking environments poses many difficulties, the most critical challenge of which is efficient support for seamless mobility. SIP is a promising nominee for managing mobility in heterogeneous networks as it provides mobility within the application layer and the characteristics of the lower layer protocols are invisible to it. However, the performance of SIP-based mobility management is downgraded, resulting from its adoption of TCP/UDP for signaling and its strict separation between the lower layers and the application layer of the protocol stack. In this paper, a SIP-based cross-layer design for fast handoffs is proposed to shorten the service interruption time when a mobile node crosses the overlapped area of a WLAN/3G cellular system. As will be shown by the simulation results, the SIP-based solution proposed in this paper effectively lessens the handoff delays caused by either the horizontal handoff or vertical handoff in future all-IP heterogeneous wireless networks.