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An efficient technique for detecting Skype flows in UDP media streams

机译:一种检测UDP媒体流中Skype流的有效技术

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As the use and popularity of VoIP applications grows, more and more Internet traffic are being generated by them. Many VoIP applications uses RTP to carry media traffic. Notable examples includes Gtalk, Google+ Hangouts, Asterisk based VoIP and Apple's FaceTime. On the other hand, Skype uses a proprietary protocol based on P2P architecture. It uses encryption for end to end communications and adopts obfuscation and anti reverse engineering techniques to prevent reverse engineering of the Skype protocol. This makes the detection of Skype flows a challenging task. Although Skype encrypts all communications, still a portion of Skype payload header known as Start of Message (SoM) is left unencrypted. In this paper, we develop an efficient technique for detection of Skype flows in UDP media streams. Our detection techniques relies on heuristics based on the information contained in Skype SoM and RTP headers.
机译:随着VoIP应用程序的使用和普及,它们正在产生越来越多的Internet流量。许多VoIP应用程序使用RTP承载媒体流量。著名的例子包括Gtalk,Google +环聊,基于Asterisk的VoIP和Apple的FaceTime。另一方面,Skype使用基于P2P架构的专有协议。它使用加密进行端到端通信,并采用混淆和反逆向工程技术来防止对Skype协议进行逆向工程。这使得检测Skype流量成为一项艰巨的任务。尽管Skype会加密所有通信,但仍保留未加密的一部分Skype有效负载头(称为消息开始(SoM))。在本文中,我们开发了一种用于检测UDP媒体流中Skype流的有效技术。我们的检测技术依赖于基于Skype SoM和RTP标头中包含的信息的启发式方法。

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