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A mechanism to recover voice from logged VoIP sessions based on RTP

机译:基于RTP的记录VoIP会话中恢复语音的机制

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The convergence of voice and data on Internet Protocol (IP) networks has led to tremendous growth of Voice over IP (VoIP). While SIP/ H.323 are dominant signaling protocols, RTP is widely used for media transfer. Certain situations may require logging of RTP sessions along with the signaling information for call and voice analysis. Recovering voice from RTP packets requires explicit handling of RTP stack and decoding. In this paper, we propose a scheme that uses softphones with appropriate codec support for voice recovery from logged RTP packets. It is done using inherent provisions of RTP, specifically SSRC-multiplexing. The scheme also suggests the possibility of an active stream injection attack in RTP based VoIP.
机译:Internet协议(IP)网络上的语音和数据的融合导致IP语音(VoIP)的巨大增长。虽然SIP / H.323是主导信令协议,但RTP广泛用于媒体传输。某些情况可能需要记录RTP会话以及呼叫和语音分析的信令信息。从RTP数据包恢复语音需要明确处理RTP堆栈和解码。在本文中,我们提出了一种使用SOFTPHONS的方案,该方案具有适当的编解码器对来自Logged RTP数据包的语音恢复。使用RTP的固有规定,特别是SSRC多路复用。该方案还表明基于RTP的VoIP中的主动流注入攻击的可能性。

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