What is ZRTP protocol and why it is used?

What is ZRTP protocol and why it is used?

ZRTP, short from Zimmermann Real-time Transport Protocol, is a cryptographic key-agreement protocol meant to negotiate the keys for encryption between two end points in Voice-over-Internet-Protocol (VoIP) telephony. In other words, ZRTP provides end-to-end encryption for VoIP calls.

What does ZRTP stand for?

Real-time Transport Protocol
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 IP (VoIP) phone telephony call based on the Real-time Transport Protocol.

What is ZRTP protocol?

Introduction ZRTP is a key agreement protocol that performs a Diffie-Hellman key exchange during call setup in the media path and is transported over the same port as the Real-time Transport Protocol (RTP) [RFC3550] media stream which has been established using a signaling protocol such as Session Initiation Protocol ( …

What is Srtp in networking?

The Secure Real-time Transport Protocol (SRTP) is a profile for Real-time Transport Protocol (RTP) intended to provide encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast applications.

What encryption does signal use?

Signal messages are encrypted with the Signal Protocol (formerly known as the TextSecure Protocol). The protocol combines the Double Ratchet Algorithm, prekeys, and an Extended Triple Diffie–Hellman (X3DH) handshake. It uses Curve25519, AES-256, and HMAC-SHA256 as primitives.

What are the two popular methods of exchange voice media key?

There are two types of keys used by SRTP one is the session key and other is the master key.

Is SRTP end to end encryption?

SRTP also allows developers to disable encryption with the NULL cipher. The NULL cipher does not perform any encryption, and instead operates as an identity function. It copies the input stream directly to the output stream without any changes.

Is SRTP secure?

RTP is suitable for video-streaming application, telephony over IP like Skype and conference technologies. The secure version of RTP, SRTP, is used by WebRTC, and uses encryption and authentication to minimize the risk of denial-of-service attacks and security breaches.

Does WebRTC use TLS?

WebRTC uses TLS sessions or QUIC for its signaling transport – both are encrypted in nature. All other avenues for non-encrypted signaling don’t really exist in WebRTC.

Is DTLS end-to-end?

It is possible to use end-to-end encryption for the signaling messages which then establish a binding between an identity and the fingerprint. This even applies if your traffic is routed through a TURN server, which by design does not know the encryption keys negotiated via DTLS. It is encrypted to the peer.

How is ZRTP used in voice over Internet Protocol?

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…

Which is cryptographic key exchange protocol does ZRTP use?

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 IP (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.

What’s the difference between SRTP, TLS and ZRTP?

TLS SSL and the newer version TLS are cryptographic protocols that provide security on the Internet. TLS with SIP is used to encrypt sip signaling whereas SRTP (Secure Real-time Transport Protocol) / ZRTP (Z and Real-time Transport Protocol) is used to encrypt media streams.

Who are the authors of the ZRTP project?

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.