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ZRTP (composed of Z and Real-time Transport Protocol) is a cryptographic key-agreement protocol to negotiate the keys for encryption between two end points in a Voice over Internet Protocol (VoIP) phone telephony call based on the Real-time Transport Protocol. It uses Diffie–Hellman key exchange and the Secure Real-time Transport Protocol (SRTP) for encryption. ZRTP was developed by Phil Zimmermann, with help from Bryce Wilcox-O'Hearn, Colin Plumb, Jon Callas and Alan Johnston and was submitted to the Internet Engineering Task Force (IETF) by Zimmermann, Callas and Johnston on March 5, 2006 and published on April 11, 2011 as RFC 6189. ==Overview== ZRTP ("Z" is a reference to its inventor, Zimmermann; "RTP" stands for Real-time Transport Protocol)〔(''Alan B. Johnston's Blog: ZRTP Published Today as RFC 6189'' ). Retrieved 2013-01-13〕 is described in the Internet Draft as a ''"key agreement protocol which performs Diffie–Hellman key exchange during call setup in-band in the Real-time Transport Protocol (RTP) media stream which has been established using some other signaling protocol such as Session Initiation Protocol (SIP). This generates a shared secret which is then used to generate keys and salt for a Secure RTP (SRTP) session."'' One of ZRTP's features is that it does not rely on SIP signaling for the key management, or on any servers at all. It supports opportunistic encryption by auto-sensing if the other VoIP client supports ZRTP. This protocol does not require prior shared secrets or rely on a Public key infrastructure (PKI) or on certification authorities, in fact ephemeral Diffie–Hellman keys are generated on each session establishment: this allows the complexity of creating and maintaining a trusted third-party to be bypassed. These keys contribute to the generation of the session secret, from which the session key and parameters for SRTP sessions are derived, along with previously shared secrets (if any): this gives protection against man-in-the-middle (MiTM) attacks, so long as the attacker was not present in the first session between the two endpoints. ZRTP can be used with any signaling protocol, including SIP, H.323, Jingle, and distributed hash table systems. ZRTP is independent of the signaling layer, because all its key negotiations occur via the RTP media stream. ZRTP/S, a ZRTP protocol extension, can run on any kind of legacy telephony networks including GSM, UMTS, ISDN, PSTN, SATCOM, UHF/VHF radio, because it is a narrow-band bitstream-oriented protocol and performs all key negotiations inside the bitstream between two endpoints. Alan Johnston named the protocol ZRTP because in its earliest Internet drafts〔 it was based on adding header extensions to RTP packets, which made ZRTP a variant of RTP. In later drafts the packet format changed to make it syntactically distinguishable from RTP. In view of that change, ZRTP is now a pseudo-acronym. 抄文引用元・出典: フリー百科事典『 ウィキペディア（Wikipedia）』 ■ウィキペディアで「ZRTP」の詳細全文を読む スポンサード リンク
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