设计对SIP呼叫设置的攻击

A. M. Hagalisletto, Lars Strand
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引用次数: 5

摘要

互联网上运行的许多协议既没有形式化,也没有正式分析。在实际应用中使用的电信协议的文档数量是巨大的,而可用的分析方法和工具需要精确和明确的协议条款。在IP语音(VoIP)应用程序中使用的会话发起协议(SIP)的手动形式化是不可行的。因此,通过结合从IETF发布的规范文档中检索到的信息和真实SIP流量的跟踪,除了协议的实现之外,我们还制定了协议的正式规范。在我们的工作过程中,我们发现了几个弱点,包括SIP呼叫设置和协议的Asterisk实现。这些弱点可能被利用,并对VoIP呼叫的身份验证和不可否认性构成威胁。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
Designing attacks on SIP call set-up
Many protocols running over the internet are neither formalised, nor formally analysed. The amount of documentation for telecommunication protocols used in real-life applications is huge, while the available analysis methods and tools require precise and clear-cut protocol clauses. A manual formalisation of the Session Initiation Protocol (SIP) used in Voice over IP (VoIP) applications is not feasible. Therefore, by combining the information retrieved from the specification documents published by the IETF and traces of real-world SIP traffic, we craft a formal specification of the protocol in addition to an implementation of the protocol. In the course of our work we detected several weaknesses, both of SIP call set-up and in the Asterisk implementation of the protocol. These weaknesses could be exploited and pose as a threat for authentication and non-repudiation of VoIP calls.
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