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Scaling up VoIP: Transport Protocols and Controlling Unwanted Communication Requests.

机译:扩大VoIP:传输协议和控制不需要的通信请求。

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摘要

Millions of people worldwide use voice over IP (VoIP) services not only as cost-effective alternatives to long distance and international calls but also as unified communication tools, such as video conferencing. Owing to the low cost of new user accounts, each person can easily obtain multiple accounts for various purposes. Rich VoIP functions combined with the low cost of new accounts and connections attract many people, resulting in a dramatic increase in the number of active user accounts. Internet telephony service providers (ITSPs), therefore, need to deploy VoIP systems to accommodate this growing demand for VoIP user accounts. Attracted people also include bad actors who make calls that are unwanted to callees. Once ITSPs openly connect with each other, unwanted bulk calls will be at least as serious a problem as email spam. This dissertation studies how we can reduce load both on ITSPs and end users to ensure continuing the success of VoIP services.;From ITSPs' perspective, the scalability of VoIP servers is of importance and concern. Scalability depends on server implementation and the transport protocol for SIP, VoIP signaling. We conduct experiments to understand the impact of connection-oriented transport protocols, namely, TCP and SCTP, because of the additional costs of handling connections. Contradicting the negative perception of connection-oriented transport protocols, our experimental results demonstrate that the TCP implementation in Linux can maintain comparable capacity to UDP, which is a lightweight connection-less transport protocol. The use of SCTP, on the other hand, requires improving the Linux implementation since the not-well-tested implementation makes a server less scalable. We establish the maximum number of concurrent TCP or SCTP connections as baseline data and suggest better server configurations to minimize the negative impact of handling a large number of connections. Thus, our experimental analysis will also contribute to the design of other servers with a very large number of TCP or SCTP connections.;From the perspective of end users, controlling unwanted calls is vital to preserving the VoIP service utility and value. Prior work on preventing unwanted email or calls has mainly focused on detecting unwanted communication requests, leaving many messages or calls unlabeled since false positives during filtering are unacceptable. Unlike prior work, we explore approaches to identifying a "good" call based on signaling messages rather than content. This is because content-based filtering cannot prevent call spam from disturbing callees since a ringing tone interrupts them before content is sent.;Our first approach uses "cross-media relations." Calls are unlikely to be unwanted if two parties have been previously communicated with each other through other communication means. Specifically, we propose two mechanisms using cross-media relations. For the first mechanism, a potential caller offers her contact addresses which might be used in future calls to the callee. For the second mechanism, a callee provides a potential caller with weak secret for future use. When the caller makes a call, she conveys the information to be identified as someone the callee contacted before through other means. Our prototype illustrates how these mechanisms work in web-then-call and email-then-call scenarios. In addition, our user study of received email messages, calls, SMS messages demonstrates the potential effectiveness of this idea.;Another approach uses caller's attributes, such as organizational affiliation, in the case where two parties have had no prior contact. We introduce a lightweight mechanism for validating user attributes with privacy-awareness and moderate security. Unlike existing mechanisms of asserting user attributes, we design to allow the caller to claim her attributes to callees without needing to prove her identity or her public key. To strike the proper balance between the ease of service deployment and security, our proposed mechanism relies on transitive trust, through an attribute validation server, established over transport layer security. This mechanism uses an attribute reference ID, which limits the lifetime and restricts relying parties. Our prototype demonstrates the simplicity of our concept and the possibility of practical use.
机译:全球数以百万计的人使用IP语音(VoIP)服务不仅是长途电话和国际电话的经济有效替代方案,而且还用作视频会议等统一通信工具。由于新用户帐户的成本低廉,每个人都可以轻松获得用于各种目的的多个帐户。丰富的VoIP功能以及低成本的新帐户和连接吸引了许多人,从而导致活动用户帐户的数量急剧增加。因此,Internet电话服务提供商(ITSP)需要部署VoIP系统,以适应对VoIP用户帐户不断增长的需求。被吸引的人还包括坏演员,他们打来的电话对被呼叫者来说是不需要的。一旦ITSP彼此公开连接,不必要的批量呼叫至少会像垃圾邮件一样严重。本文研究了如何降低ITSP和最终用户的负担,以确保VoIP服务的持续成功。从ITSP的角度来看,VoIP服务器的可伸缩性至关重要。可伸缩性取决于服务器实施方式以及SIP,VoIP信令的传输协议。我们进行实验以了解面向连接的传输协议(即TCP和SCTP)的影响,因为处理连接会产生额外费用。与对面向连接的传输协议的否定看法相反,我们的实验结果表明,Linux中的TCP实现可以保持与UDP相当的容量,而UDP是一种轻量级的无连接传输协议。另一方面,SCTP的使用需要改进Linux的实现,因为未经良好测试的实现会使服务器的可伸缩性降低。我们将并发TCP或SCTP连接的最大数量作为基线数据,并建议采用更好的服务器配置,以最大程度地减少处理大量连接的负面影响。因此,我们的实验分析也将有助于其他具有大量TCP或SCTP连接的服务器的设计。从最终用户的角度来看,控制不必要的呼叫对于保留VoIP服务的实用性和价值至关重要。防止不想要的电子邮件或电话的现有工作主要集中在检测不想要的通信请求上,由于过滤期间的误报是不可接受的,因此使许多消息或电话没有标签。与先前的工作不同,我们探索基于信令消息而非内容来识别“良好”呼叫的方法。这是因为基于内容的筛选无法阻止垃圾邮件打扰被呼叫者,因为铃声会在内容发送之前将其打断。我们的第一种方法是使用“跨媒体关系”。如果事先已通过其他通信方式将两方彼此通信,则呼叫不太可能是不需要的。具体来说,我们提出了两种使用跨媒体关系的机制。对于第一种机制,潜在的呼叫者提供了她的联系地址,该地址可能在以后的被呼叫者中使用。对于第二种机制,被呼叫者为潜在的呼叫者提供了较弱的秘密,以供将来使用。当呼叫者拨打电话时,她通过其他方式传达了被识别为被呼叫者之前联系过的人的信息。我们的原型说明了这些机制如何在网络先呼和电子邮件先呼的情况下工作。此外,我们对接收到的电子邮件,电话和SMS消息的用户研究证明了该想法的潜在有效性。在另一方没有事先联系的情况下,另一种方法是使用呼叫者的属性(例如组织隶属关系)。我们引入了一种轻量级的机制,用于通过隐私意识和中等安全性来验证用户属性。与声明用户属性的现有机制不同,我们设计为允许呼叫者向被调用者声明其属性,而无需证明自己的身份或公共密钥。为了在服务部署的简便性和安全性之间取得适当的平衡,我们提出的机制依靠通过在传输层安全性之上建立的属性验证服务器的传递信任。此机制使用属性引用ID,该ID会限制生存期并限制依赖方。我们的原型展示了我们概念的简单性和实际使用的可能性。

著录项

  • 作者

    Ono, Kumiko.;

  • 作者单位

    Columbia University.;

  • 授予单位 Columbia University.;
  • 学科 Computer Science.
  • 学位 Ph.D.
  • 年度 2012
  • 页码 207 p.
  • 总页数 207
  • 原文格式 PDF
  • 正文语种 eng
  • 中图分类
  • 关键词

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