一种可支持多种多媒体业务的基于IMS的IPTV架构

IP多媒体子系统用于实现端到端的IP多媒体通信,具有接入无关性、对各种接入技术广适性的特点,因此,下一代网络采用IP多媒体子系统作为核心架构.提出一个基于IP多媒体子系统的IP电视架构,在IP多媒体子系统核心模块的基础上,以一组互相独立的基于会话初始化协议的功能模块构成应用服务器,并设计了能够支持和提供不同IM S业务的客户端与内容提供商端,使该IPTV框架可以在下一代网络上支持基于IP多媒体子系统的包括IPTV业务在内的多种多媒体业务.     

A transport-level proxy for secure multimedia streams

To provide a secure traversal service, firewalls need more than static packet filtering and application-level proxies. SOCKS (Secure sOCKets) is an application-independent transport-level proxy that offers user-level authentification and data encryption. An extended SOCKS UDP (user datagram protocol) binding model with appropriate socket calls is proposed to provide complete support for UDP-based multimedia streaming applications     

穿越NAT/防火墙的SIP端到端网络安全机制

介绍了会话初始协议的一种扩展，实现了会话描述协议（SDP）和穿越NAT／防火墙的端到端网络安全机制。该解决方案基于安全多用途网际邮件扩充协议（S／MIME）和中间体通信（MIDCOM）协议实现。用户授权代理服务器代替自己加密会话描述信息，该代理选定接收方并为接收域中的SIP代理眼务器加密SDP。当每个终端用户能经由一条安全链接联系到它可信赖的SIP代理并授权该代理加密信号数据时，会话信息就得到了端到端的安全保护。     

Design and realization of ad-hoc VoIP with embedded p-SIP server

In this paper we design and implement the pseudo session initiation protocol (p-SIP) server embedded in each mobile node to provide the ad-hoc voice over Internet protocol (VoIP) services. The implemented p-SIP server, being compatible with common VoIP user agents, integrates the standard SIP protocol with SIP presence to handle SIP signaling and discovery mechanism in the ad-hoc VoIP networks. The ad-hoc VoIP signaling and voice traffic performances are analyzed using E-model R rating value for up to six hops in the implemented test-bed. We also conduct the interference experiments to imitate the practical ad-hoc VoIP environment. The analyzed results demonstrate the realization of ad-hoc VoIP services by using p-SIP server.     

Protocol considerations for a prefix-caching proxy for multimedia streams

The increasing popularity of multimedia streaming applications introduces new challenges in content distribution. Web-initiated multimedia streams typically experience high start-up delay, due to large protocol overheads and the poor delay, throughput, and loss properties of the Internet. Internet service providers can improve performance by caching the initial segment (the prefix) of popular streams at proxies near the requesting clients. The proxy can initiate transmission to the client while simultaneously requesting the remainder of the stream from the server. This paper analyzes the challenges of realizing a prefix-caching service in the context of the IETF's Real-Time Streaming Protocol (RTSP), a multimedia streaming protocol that derives from HTTP. We describe how to exploit existing RTSP features, such as the Range header, and how to avoid several round-trip delays by caching protocol information at the proxy. Based on our experiences, we propose extensions to RTSP that would ease the development of new multimedia proxy services. In addition, we discuss how caching the partial contents of multimedia streams introduces new challenges in cache coherency and feedback control. Then, we briefly present our preliminary implementation of prefix caching on a Linux-based PC, and describe how the proxy interoperates with the RealNetworks server and client.     

基于移动多媒体CTMM的研究

移动多媒体用户数量的增加使得网络瓶颈现象日益严重,以及多媒体源服务器和代理服务器的负载较大,这严重地影响了移动多媒体的QoS（Quality of Service）。为了提高移动多媒体网络传输带宽,减小多媒体源服务器和代理服务器的负载,提高多媒体请求数据的命中率,减小请求响应延迟时间,在传统的移动多媒体系统两层架构模型的基础上,本文提出基于簇的三层中间件架构移动多媒体传输模型CTMM（Cluster-basedThree-tier Middleware Model）。研究CTMM提出的三层架构,介绍第二层架构的核心部分local-mediator的模块设计及主要模块采用的关键技术。仿真实验说明该模型的有效性。     

基于P2P＋SIP的流媒体服务系统的设计

从网络多媒体系统的业务需求和功能模块分析出发,提出了在底层P2P网络上建立流媒体服务系统,并使用SIP技术进行呼叫、建立、断开等系统的功能。该体系结构保证了在P2P系统中的零配置、鲁棒性和适应性,另外还可以与现有的SIP设备连通。虽然增加了呼叫建立传输时间,但由于信息是由双方直接发送,无论是C/S结构还是P2P结构,都不经过SIP代理服务器,因此不影响信息延时。     

A study of performance and scalability metrics of a SIP proxy server – a practical approach

In recent years, Internet Protocol (IP) telephony has been a real alternative to the traditional Public Switched Telephone Networks (PSTN). IP telephony offers more flexibility in the implementation of new features and services. The Session Initiation Protocol (SIP) is becoming a popular signalling protocol for Voice over IP (VoIP) based applications. The SIP proxy server is a software application that provides call routing services by parsing and forwarding all the incoming SIP packets in an IP telephony network. The efficiency of this process can create large scale, highly reliable packet voice networks for service providers and enterprises. We established that the efficient design and implementation of the SIP proxy server architecture can enhance the performance characteristics of a SIP proxy server significantly. Since SIP proxy server performance can be characterised by its transaction states of each SIP session, we emulated the $M/M/1$ performance model of the SIP proxy server and studied some of the key performance benchmarks such as average response time to process the SIP calls, and mean number of SIP calls in the system. We showed its limitations, and then studied an alternative $M/M/c$ based SIP proxy server performance model with enhanced performance model and studied additional key performance characteristics such as server utilisation, queue size and memory utilisation. Provided the comparative results between the predicted results with the experimental results conducted in a lab environment.     

SIP和RTSP协议转换模块的设计与实现

随着流媒体技术与IP通信技术的快速发展,SIP协议与RTSP协议的互联互通问题日益凸显.论文从信令对应、地址映射、交互流程设计等几个方面对SIP与RTSP协议互通进行了论述,并在RTSP代理服务器的基础上,设计并完成了SIP与RTSP协议转换模块,实现了SIP信令与RTSP信令的转换功能.最后,论文以SIP Client 访问RTSP摄像头获取实时画面为例,验证了该方法的可行性与正确性,为基于SIP的多媒体调度系统与RTSP设备的集成问题提供了较好的解决方案.     

Design of a password-based authenticated key exchange protocol for SIP

The Session Initiation Protocol (SIP) is a signaling communications protocol, which has been chosen for controlling multimedia communication in 3G mobile networks. In recent years, password-based authenticated key exchange protocols are designed to provide strong authentication for SIP. In this paper, we address this problem in two-party setting where the user and server try to authenticate each other, and establish a session key using a shared password. We aim to propose a secure and anonymous authenticated key exchange protocol, which can achieve security and privacy goal without increasing computation and communication overhead. Through the analysis, we show that the proposed protocol is secure, and has computational and computational overheads comparable to related authentication protocols for SIP using elliptic curve cryptography. The proposed protocol is also provably secure in the random oracle model.     

SIP协议控制多媒体会话的研究与应用

SIP是NGN(Next Generation Network)呼叫控制中的一个重要协议，其应用研究正方兴未艾。该文基于NGN关键设备——媒体服务器的开发实践，对其中运用SIP控制媒体服务器提供多媒体业务进行了说明和讨论，特别以交互语音应答(Interactive Voice Response, IVR)为例，给出了一种用SIP及其扩展控制多媒体会话的具体实现方案。     

综合UDP打洞与Http代理的SIP穿越NAT方案

软交换是下一代网络中的核心技术,SIP协议作为下一代网络最重要的协议之一,已经被广泛应用于VoIP系统中。 SIP协议无法支持SIP信令和媒体流的NAT穿越,从而限制了其在广域网上的应用和发展。虽然目前解决NAT穿越的方案已经很多,但都存在着一定的局限性。文中详细解释了NAT对SIP通信的影响,介绍了UDP打洞技术的基本原理,介绍了UDP打洞技术穿越锥形NAT的流程,以及Http代理网关方式穿越各种NAT的流程。文中通过比较各种NAT穿越方案的优缺点,提出一种综合UDP打洞与Http代理网关的NAT穿越方案。经过论证与实验,证明了该方案的可行性。     

Integrating Internet telephony services

Cost savings and the ease of developing and adding new services have motivated great interest in Internet telephony, which integrates services provided by the Internet with the public switched telephone network (PSTN). Internet telephony relies on several protocols, including the real-time transport protocol (RTP) for multimedia data transport and the session initiation protocol (SIP) or H.323 for establishing and controlling sessions. SIP can integrate with other Internet services, such as email, the Web, voice mail, instant messaging, conference calling, and multimedia collaboration. We have implemented a SIP-based software suite called the Columbia Internet extensible multimedia architecture (Cinema), which we installed and integrated with the existing private branch exchange (PBX) infrastructure in the computer science department at Columbia University. The Cinema environment provides interoperability with the PSTN, programmable Internet telephony services, and IP-based voice mail. It also integrates Web access and e-mail for unified messaging and supports multiparty multimedia conferencing. The setup lets us extend our PBX capacity and will eventually let us replace it while keeping our existing phone numbers. It also provides an environment in which we can easily add new services and features, including interoperation with existing multimedia tools, e-mail access from standard. telephones, network appliance control, and instant messaging support     

VoIP系统SIP—QoS代理服务器设计方案

文中以解决VoIP系统的语音质量问题为目标,深入研究了基于SIP的VoIP系统QoS控制技术。参照IMS网络结构,考虑通信业务的QoS要求,研究以SIP为信令协议的VoIP系统如何进行呼叫控制、资源预留和策略决策,融人到SIP用户代理、SIP代理服务器,提出了一个具有QoS能力的SIP代理服务器的设计方案,增加了策略决策功能（PDF）等网络实体,支持QoS的能力得到增强。文中详细讨论了支持QoS的siP网络的增强能力,具有QoS能力的SIP代理服务器的功能结构,QoS功能模块,以及QoS资源授权和预留决策过程。     

A robust method for soft IP handover

Multimedia streaming over wireless networks - often called mobile multimedia streaming lets users access music, movie, and news services at any time, regardless of location. Given that multimedia streaming is a key goal of third-generation and future wireless networks, vendors will soon deploy streaming clients in advanced mobile terminals. Current mobile terminals, however, fail to adequately support mobile multimedia communication because wireless networks have high packet-loss rates. To eliminate packet loss during handover, we use a packet path diversity scheme and an end-to-end bicasting mechanism that enables soft IP handover. To offset wireless errors, we use a forward error correction (FEC) scheme and embed it in the bicasting mechanism. Our bicasting method encodes the data stream and then splits it, providing more effective diversity than general bicasting, which sends the same data down both paths.' To support our method, we propose the mobile multimedia streaming protocol (MMSP), a new transport-layer protocol that supports multihoming and bicasting in combination with FEC.     

A Cross-Layer Analysis of Session Setup Delay in IP Multimedia Subsystem (IMS) With EV-DO Wireless Transmission

This paper analyzes the session setup delay in the IP multimedia subsystem (IMS) with the CDMA2000 evolution data only rev. A (EV-DO rev. A) standard for wireless transmission. Session setup delay is particularly critical for interactive multimedia applications, such as gaming, push-to-X and voice over IP (VoIP), as it directly translates in user perception of service quality. Keeping signaling delay low, however, is a challenge in IMS due to the text-based nature of the session initiation protocol (SIP) for signaling, and, more significantly, due to the lossy and capacity constrained wireless links. To address this challenge, we analyze the session setup delay end-to-end, by taking into account key system properties across all layers, ranging from radio links to IMS signaling architecture. We present a model for cross-layer performance analysis and simulation, which includes the statistical properties of the EV-DO (rev. A) wireless channel, and also takes into consideration the properties of transport protocols (TCP, UDP) and SIP signaling (message size and compression). By means of analysis and simulations, we study the setup delay performance of a generic, multi-operator IMS communication scenario between two mobile users. We describe how session setup delay can be estimated and reduced in realistic IMS settings and we propose architecture alternatives to the basic IMS scenario. The results derived from this study show that the proposed methods can incrementally lead to a lower setup delay and less sensitivity to the radio transmission quality and frame error rate compared to the base IMS scenario     

SIP多媒体网络模型在Windows平台上的实现

SIP多媒体网络模型包括各种类型的SIP服务器及SIP终端。文章分析了SIP协议的特点和优势，描述了Windows平台SIP多媒体网络的构架，提出了多用途SIP服务器的实现以及SIP终端的实现过程。     

IMS Application Servers: Roles, Requirements, and Implementation Technologies

The IP multimedia subsystem (IMS) defines a generic architecture to support communication services over a Session Initiation Protocol (SIP) infrastructure. In the IMS architecture, application servers host and execute the IMS service logic. These servers can be SIP application servers, open services architecture (OSA) application servers, or a customized applications for mobile networks using enhanced logic (Camel) service environment. Some technologies used in telephony and voice-over-IP (VoIP) application servers are also applicable to IMS application servers, but such servers have some unique requirements that could limit the extent to which these technologies can meet them.     

UDP地址欺骗剖析及其对策

目前UDP协议在Internet上被广泛应用，但是UDP数据包易于伪造，其本身存在重大的安全隐患。对伪造数据报头中的源地址和端口的UDP地址欺骗的入侵行为进行了分析并提出了一种通过添加IP记录路由选项的方法来防止UDP地址欺骗，研究结果表明，这种方法可以一定程度上防止UDP地址欺骗的入侵行为。