共查询到19条相似文献,搜索用时 78 毫秒
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比较了当前流行使用穿透NAT的几种方案,分析了基于SIP协议的IP电话穿透NAT的问题所在。提出了一种新的穿透方式,即采用远程服务获取真实端口的策略,对当前终端、NAT设备不做任何修改,不增加新设备的前提下,达到穿透NAT。通过实际测试,验证了该方案的可行性。 相似文献
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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. 相似文献
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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. 相似文献
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网络地址转换(NAT)是会话发起协议(SIP)应用中一个巨大的障碍,如何解决SIP穿透NAT的问题已成为当前互联网领域研究的热点。文章剖析了NAT的工作原理,并针对SIP协议信令过程的特点,提出了采用ALG设备来解决NAT的穿透问题,具体设计了ALG的结构和相关实现算法,并给出了详细的实现方法。 相似文献
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NAT使得SIP端到端的通信变得非常困难,文章分析了几种SIP穿越NAT方法及其不足,提出了一种基于最短路径的NAT穿越方法SPNT(Shortest Path for NAT Traversal).其基本思想是根据SIP终端和代理服务器之间的信令交互,判断终端所在网络的NAT类型,代理服务器对不同的NAT类型采用不同的方式实现SIP信令穿越,而媒体流的穿越则通过终端进行媒体地址的连通性检测,使终端之间能够动态的建立最短的媒体数据连接.该方法在不改变现有NAT的情况下,在应用层实现了对所有NAT的有效穿越,避免了单独使用某一方法而带来的缺陷. 相似文献
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VoIP系统中NAT穿越技术的研究与实现 总被引:1,自引:1,他引:0
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。 相似文献
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Stefano Salsano Luca Veltri Andrea Polidoro Alessandro Ordine 《Wireless Personal Communications》2007,43(3):1019-1034
This paper describes a Session Initiation Protocol (SIP) based solution for mobility management that provides seamless mobile
multimedia services in a heterogeneous scenario where different radio access technologies are used (802.11/ WiFi, Bluetooth,
2.5G/3G networks). The solution relies on the so called “Session Border Controllers” which are now widely used in many commercial
SIP telephony solutions, mainly to deal with NAT traversal. Session Border Controller functionality has been extended to support
seamless mobility for multimedia applications. A prototype of the proposed solution focused on VoIP services has been implemented
in a test bed which is able to perform seamless handovers (and NAT traversal) using the 802.11, Bluetooth and 3G (UMTS) access
networks. Measurements results are reported which analyze the performance of the solution in a real world environment, using
commercial WiFi and 3G services. 相似文献
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基于ARM和DSP的VoIP网关的设计与实现 总被引:3,自引:3,他引:0
文章介绍了基于ARM和DSP的VoIP网关的软硬件设计.其中硬件主要由基于ARM内核的微处理器S3C44B0子系统和基于LSI403LP芯片的DSP子系统以及电话接口模块构成.软件设计主要负责控制SIP核心协议栈oSIP、RTP/RTCP核心协议栈ccRTP的正常运转,控制外围DSP芯片工作.测试证明,文章所设计的VoIP网关能够实现网关的主要功能,具有成本低廉、应用灵活、可扩展性好的特点. 相似文献
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Jung‐Shian Li Chuan‐Kai Kao Jau‐Jan Tzeng 《International Journal of Communication Systems》2011,24(7):837-851
A secure architecture is proposed in the home/office networking environment with a special appliance, called Building Security Gateway (BSG), for cheap SIP devices and unsupported soft user agents (UAs) performing security functions successfully, such as construction of secure sessions and encrypting/decrypting media. Therefore, we apply BSG procedure to VoIP for getting secure session. Furthermore, VoIP call monitoring and intercepting functions are also implemented in BSG. Secure session monitoring is supported in this architecture. A prototype BSG is built in a home/office network, and BSG‐based session mobility cases between UAs are examined and the protocol primitives for security and call intercepting are verified. Copyright © 2010 John Wiley & Sons, Ltd. 相似文献
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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. 相似文献
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SIP协议在VoIP终端的设计和实现 总被引:1,自引:0,他引:1
SIP是下一代网络中的重要协议,而基于SIP协议的VoIP业务已经对传统话音业务形成了重要的威胁,并成为各大运营商竞争的重点业务之一.概述了VoIP业务的发展现状,介绍了SIP协议的实体、消息机制以及它所提供的业务,通过对SIP协议的原理及工作流程的分析,论证了其在实现一个VoIP系统中的优势,在此基础上设计并实现了一个基于SIP协议的VoIP终端. 相似文献
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由于MANET自组织、无中心等特点,传统的VoIP体系结构不能直接应用在MANET之上,这就要求对传统VOIP系统进行一定的改造,使之能够适应MANET环境。本文提出将P2P的簇状组网结构引入SIP,从而可以在MANET上构建基于SIP的可灵活扩展的VOIP通信系统,分析了在该架构中的主要操作的典型工作机制,采用本方案的VOIP系统既可以适应MANET环境的需求,又可以与现有的SIP系统无缝通信。本文最后展望了未来的研究方向。 相似文献
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Voice over IP signaling: H.323 and beyond 总被引:4,自引:0,他引:4
《Communications Magazine, IEEE》2000,38(10):142-148
Signaling has been one of the key areas of Voice over IP (VoIP) technologies since inception. H.323 was the key protocol that allowed interoperability of VoIP products and moved the industry away from the initial proprietary solutions. Once the VoIP industry started maturing, some limitations of H.323 came to light. In this article we provide an overview of H.323, describe its capabilities, and discuss how its limitations are being addressed using the concept of gateway decomposition. We also discuss how H.323 can coexist with other protocols such as MGCP, H.248, and SIP which are attracting a lot of interest in the VoIP industry today 相似文献