SIP在3G通信网络中的应用研究

在高速移动的条件下方便快捷而又经济地通信,是市场的迫切需求.会话初始化协议(SIP)是能够在第3代移动通信系统(3G)中传输IP(Internet Protocol)多媒体业务的信令协议.它能够融合Internet和移动蜂窝系统.文中简要介绍和分析了SIP,并给出了一种基于该协议的在3G通信网络中的应用方案.     

SIP协议栈探讨与应用

SIP(会话初始化协议)是伴随着Internet的发展同时借鉴了Web业务成功经验的、由IETF制定的一套网络多媒体信令协议,主要用于创建、修改和终止多媒体呼叫与会话,是一个与HTTP和SMTP类似的、基于文本的协议,具有易读取、易扩展以及易于调试的特性。简单介绍了SIP协议的功能组件以及消息机制,提出了SIP协议栈实现的层次结构模型,并给出了SIP协议栈的结构以及软件流程。     

Enabling SIP-based sessions in ad hoc networks

The deployment of infrastructure-less ad hoc networks is suffering from the lack of applications in spite of active research over a decade. This problem can be solved to a certain extent by porting successful legacy Internet applications and protocols to the ad hoc network domain. Session Initiation Protocol (SIP) is designed to provide the signaling support for multimedia applications such as Internet telephony, Instant Messaging, Presence etc. SIP relies on the infrastructure of the Internet and an overlay of centralized SIP servers to enable the SIP endpoints discover each other and establish a session by exchanging SIP messages. However, such an infrastructure is unavailable in ad hoc networks. In this paper, we propose two approaches to solve this problem and enable SIP-based session setup in ad hoc networks (i) a loosely coupled approach, where the SIP endpoint discovery is decoupled from the routing procedure and (ii) a tightly coupled approach, which integrates the endpoint discovery with a fully distributed cluster based routing protocol that builds a virtual topology for efficient routing. Simulation experiments show that the tightly coupled approach performs better for (relatively) static multihop wireless networks than the loosely coupled approach in terms of the latency in SIP session setup. The loosely coupled approach, on the other hand, generally performs better in networks with random node mobility. The tightly coupled approach, however, has lower control overhead in both the cases. This work was partially done while the author was a graduate student in CReWMaN, University of Texas at Arlington. Dr. Nilanjan Banerjee is a Senior Research Engineer in the Networks Research group at Motorola India Research Labs. He is currently working on converged network systems. He received his Ph.D. and M.S. in computer science and engineering from University of Texas at Arlington. He received his B.E. degree in the same discipline from Jadavpur University, India. His research interests include telecom network architectures and protocols, identity management and network security, mobile and pervasive computing, measures for performance, modeling and simulation, and optimization in dynamic systems. Dr Arup Acharya is a Research Staff Member in the Internet Infrastructure and Computing Utilities group at IBM T.J. Watson Research Center and leads the Advanced Networking micropractice in On-Demand Innovation Services. His current work includes SIP-based services such as VoIP, Instant Messaging and Presence, and includes customer consulting engagements and providing subject matter expertise in corporate strategy teams. Presently, he is leading a IBM Research project on scalability and performance of SIP servers for large workloads. In addition, he also works on different topics in mobile/wireless networking such as mesh networks. He has published extensively in conferences/journals and has been awarded seven patents. Before joining IBM in 2000, he was with NEC C&C Research Laboratories, Princeton. He received a B.Tech degree in Computer Science from the Indian Institute of Technology, Kharagpur and a PhD in Computer Science from Rutgers University in 1995. Further information is available at Dr. Sajal K. Das is a Professor of Computer Science and Engineering and also the Founding Director of the Center for Research in Wireless Mobility and Networking (CReWMaN) at the University of Texas at Arlington (UTA). His current research interests include sensor networks, resource and mobility management in wireless networks, mobile and pervasive computing, wireless multimedia and QoS provisioning, wireless internet architectures and protocols, grid computing, applied graph theory and game theory. He has published over 400 research papers in these areas, holds four US patents in wireless internet and mobile networks. He received Best Paper Awards in IEEE PerCom’06, ACM MobiCom’99, ICOIN’02, ACM MSwiM’00 and ACM/IEEE PADS’97. He is also recipient of UTA’s Outstanding Faculty Research Award in Computer Science (2001 and 2003), College of Engineering Research Excellence Award (2003), the University Award for Distinguished record of Research (2005), and UTA Academy of Distinguished Scholars Award (2006). He serves as the Editor-in-Chief of Pervasive and Mobile Computing journal, and as Associate Editor of IEEE Transactions on Mobile Computing, ACM/Springer Wireless Networks, IEEE Transactions on Parallel and Distributed Systems. He has served as General or Program Chair and TPC member of numerous IEEE and ACM conferences. He is a member of IEEE TCCC and TCPP Executive Committees.     

Cryptanalysis of smart‐card‐based password authenticated key agreement protocol for session initiation protocol of Zhang et al.

As the core signaling protocol for multimedia services, such as voice over internet protocol, the session initiation protocol (SIP) is receiving much attention and its security is becoming increasingly important. It is critical to develop a roust user authentication protocol for SIP. The original authentication protocol is not strong enough to provide acceptable security level, and a number of authentication protocols have been proposed to strengthen the security. Recently, Zhang et al. proposed an efficient and flexible smart‐card‐based password authenticated key agreement protocol for SIP. They claimed that the protocol enjoys many unique properties and can withstand various attacks. However, we demonstrate that the scheme by Zhang et al. is insecure against the malicious insider impersonation attack. Specifically, a malicious user can impersonate other users registered with the same server. We also proposed an effective fix to remedy the flaw, which remedies the security flaw without sacrificing the efficiency. The lesson learned is that the authenticators must be closely coupled with the identity, and we should prevent the identity from being separated from the authenticators in the future design of two‐factor authentication protocols. Copyright © 2014 John Wiley & Sons, Ltd.     

飞信协议识别与多元通联关系提取方法

针对飞信协议尚未公开与复杂互联网环境带来的飞信各类应用相关协议识别困难以及单包通联关系缺失等问题,基于SIP协议的基本框架,从文本聊天、文件传输以及音/视频通信三方面解析了飞信常用业务的协议交互过程;提出了端口与正则表达式相结合的飞信协议识别方法和基于会话还原的飞信通联关系提取方法,能够从大量混杂的数据包中快速定位飞信业务报文,获得飞信多种通信行为的通联关系。实验结果证明了本文方法的有效性。     

IETF有关Internet上多媒体会议系统的相关协议

文章在简介IETF的基础上,分别介绍了有关Internet上多媒体会议系统的相关协议,包括SDP、SAP、SIP、RTSP、RTCP等协议的基本内容及其应用。     

SIP在移动Internet中的应用

移动Internet是目前学术界研究的热点问题之一,SIP是IETF提出的IP电话信令协议。结合移动Internet技术介绍了一些扩展SIP以提供移动性支持的机制,主要讨论注册、定位和切换等方面的问题。     

基于ZXSS10 SS1b软交换的SIP终端会话设计

软交换技术是现代交换技术中一个重要的分支,其开放的体系结构支持多种信令协议,而SIP协议是其中重要协议之一,主要用于实现实时会话通信。通过对软交换体系结构的介绍以及软交换中SIP协议的应用分析,利用中兴公司的ZXSS10 SS1b软交换控制设备实现了在实验室条件下的SIP终端会话设计,给出了具体的设计过程及实验结果,为高校通信与电子信息类专业建设现代通信实验室、开设相关实验提供了有效方案。     

SIP: Simple Internet Protocol

Several features of the Simple Internet Protocol (SIP) are described and compared to those offered by the Internet Protocol (IP). The changes required for other protocols in the TCP/IP suite to accommodate SIP are discussed, as are the mechanisms available to allow gradual transition of the Internet from IP to SIP. Future directions for SIP development, a report on current implementation status, and a summary of the specific improvements offered by SIP over IP are presented     

Analysis and Applications of Session Initiation Protocol (SIP) in Internet Telephony

1　Introduction　Internet telephony, also known as Voice over IP (VoIP)or IP telephony (IPtel), is the real time delivery of voice(and possibly other multimedia data types) between two ormore parties, across networks using the Internet protocols,and the exchange of information required to control this de livery. Internet telephony offers the opportunity to design aglobal multimedia communications system that may eventual ly replace the existing telephony infrastructure.　Internet Engine…     

G.723.1 Codec在OpenH323协议栈上的实现

OpenH323是一个开放源码的VoIP(Voice over IP)协议栈,支持H．323和SIP等多媒体通信协议,为多媒体应用提供了一个很好的开发平台。G．723．1是ITU-T建议在中低速率多媒体通信中使用的语音压缩算法,目前该算法已在IP电话系统中得到广泛应啊。基于OpenH323协议栈实现G．723．1Codec有着十分重要的应用价值。介绍在OpenH323的软件终端上实现G．723．1Codec的基本方法,并可推广到G．729等其它多种语音压缩算法。     

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.     

H.323和SIP在IP多媒体网络中互通的实现

随着IP电话和视频通信的发展,H.323和SIP作为IP多媒体通信领域中被广泛采纳的两种信令控制协议,受到业界的普遍重视。如何有效地实现这两种协议之间的互通,成为近年来国内外研究的热点。本文在简要分析H.323和SIP互通要求的基础上,提出了两者互通的实现方案,并对互通需要解决的关键问题进行了讨论。     

SIP及其在NGN业务集成中的应用

会话发起协议(SIP)是IP网络应用层的控制协议,用于建立、修改和终止两个或多个参与者之间的会话.SIP的设计方法和结构特点使得它成为下一代网络软交换体系的重要技术.3GPP也决定在基于IP的核心网络中采用SIP协议栈实现多媒体会话控制信令.文章介绍了SIP的基本概念并着重阐述和分析了SIP的本质特性及SIP的适用场景,在此基础上,分析了SIP在下一代网络多媒体业务集成中的核心地位和要解决的关键问题并提出了一个SIP和相关技术结合的开放业务集成框架.     

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.     

基于JAIN SIP的多媒体会议系统设计与实现

会话初始化(SIP)协议作为下一代网络中应用层的核心控制协议,正得到越来越广泛的关注.JAIN SIP是SUN公司用于实现SIP应用而提供的一套标准JAVA接口.本文基于JAIN SIP的体系结构和机制,设计并实现了SIP多媒体会议系统.该系统支持多种SIP终端接入,具有灵活的会议召开和加入方式.     

Policy-based distributed management architecture for large-scale enterprise conferencing service using SIP

Recent advances in broadband communication and computing technology have accelerated the proliferation of Internet protocol-based multimedia conferencing services in large-scale enterprises. Most of the research on session initial protocol (SIP)-based multimedia conferencing, however, has been limited in scalability due to the centralized management of conference control by a single server. In order to overcome this limitation, we have designed policy-based distributed management architecture for a large-scale enterprise conferencing service by extending the Internet Engineering Task Force's (IETF's) approach. The salient feature of the proposed management architecture is that in addition to the distribution of media processing, both participant membership control and authorization functions are dynamically distributed in accordance with the management policy in order to improve scalability. In order to implement these distributed management functions, we have extended both SIP and conference policy control protocols of the IETF. We also show the procedures for the distributed conference management using the extended SIP signaling methods. Finally, we have evaluated by simulation the performances of the proposed architecture in comparison with the centralized architecture of the IETF. The simulation results show that the proposed architecture greatly improves scalability.     

The Session Initiation Protocol: Internet-centric signaling

The Session Initiation Protocol (SIP) provides advanced signaling and control functionality for a wide variety of multimedia services. SIP can efficiently and scalably locate resources based on a location-independent name and then negotiate session characteristics. It can find use in applications ranging from Internet telephony and conferencing to instant messaging, event notification, and the control of networked devices. We summarize the main protocol features and describe a range of extensions currently being discussed within the Internet Engineering Task Force     

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.     

VoIP网络中H.323与SIP的比较与互通研究

在VoIP网络中,H.323协议在SIP协议出现之前就已经得到了广泛使用,因此,要实现H.323协议和SIP协议的互通是当前需要解决的一个重要问题。通过简要介绍这两种协议的体系结构,进一步分析互通过程中需要处理的主要问题,提出了实现H.323与SIP互通的网络结构模型,同时对互通所必须的信令网关进行了初步研究,从而解决了两种协议之间的地址转换与映射、消息转换与映射、媒体能力协商等。经实践证明,该互通方案是可行的。