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The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS).
Skype for Business Server is real-time communications server software that provides the infrastructure for enterprise instant messaging, presence, VoIP, ad hoc and structured conferences and PSTN connectivity through a third-party gateway or SIP trunk. These features are available within an organization, between organizations and with external users on the public internet or standard phones.
A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).
A secure telephone is a telephone that provides voice security in the form of end-to-end encryption for the telephone call, and in some cases also the mutual authentication of the call parties, protecting them against a man-in-the-middle attack. Concerns about massive growth of telephone tapping incidents led to growing demand for secure telephones.
PGPfone was a secure voice telephony system developed by Philip Zimmermann in 1995. The PGPfone protocol had little in common with Zimmermann's popular PGP email encryption package, except for the use of the name. It used ephemeral Diffie-Hellman protocol to establish a session key, which was then used to encrypt the stream of voice packets. The two parties compared a short authentication string to detect a Man-in-the-middle attack, which is the most common method of wiretapping secure phones of this type. PGPfone could be used point-to-point over the public switched telephone network, or over the Internet as an early Voice over IP system.
Gizmo5 was a voice over IP communications network and a proprietary freeware soft phone for that network. On November 12, 2009, Google announced that it had acquired Gizmo5. On March 4, 2011, Google announced that the service would be discontinued as of April 3, 2011.
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. It was developed by a small team of Internet Protocol and cryptographic experts from Cisco and Ericsson. It was first published by the IETF in March 2004 as RFC 3711.
Opportunistic encryption (OE) refers to any system that, when connecting to another system, attempts to encrypt communications channels, otherwise falling back to unencrypted communications. This method requires no pre-arrangement between the two systems.
ZRTP 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. 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.
This is a comparison of voice over IP (VoIP) software used to conduct telephone-like voice conversations across Internet Protocol (IP) based networks. For residential markets, voice over IP phone service is often cheaper than traditional public switched telephone network (PSTN) service and can remove geographic restrictions to telephone numbers, e.g., have a PSTN phone number in a New York area code ring in Tokyo.
A VoIP VPN combines voice over IP and virtual private network technologies to offer a method for delivering secure voice. Because VoIP transmits digitized voice as a stream of data, the VoIP VPN solution accomplishes voice encryption quite simply, applying standard data-encryption mechanisms inherently available in the collection of protocols used to implement a VPN.
Twinkle is a free and open-source application for voice communications over Voice over IP (VoIP) protocol.
Jitsi is a collection of free and open-source multiplatform voice (VoIP), video conferencing and instant messaging applications for the Web platform, Windows, Linux, macOS, iOS and Android. The Jitsi project began with the Jitsi Desktop. With the growth of WebRTC, the project team focus shifted to the Jitsi Videobridge for allowing web-based multi-party video calling. Later the team added Jitsi Meet, a full video conferencing application that includes web, Android, and iOS clients. Jitsi also operates meet.jit.si, a version of Jitsi Meet hosted by Jitsi for free community use. Other projects include: Jigasi, lib-jitsi-meet, Jidesha, and Jitsi.
Linphone is a free voice over IP softphone, SIP client and service. It may be used for audio and video direct calls and calls through any VoIP softswitch or IP-PBX. Linphone also provides the possibility to exchange instant messages. It has a simple multilanguage interface based on Qt for GUI and can also be run as a console-mode application on Linux.
Blink is a Session Initiation Protocol (SIP) client distributed under the Blink license.
CSipSimple is a Voice over Internet Protocol (VoIP) application for Google Android operating system using the Session Initiation Protocol (SIP). It is open source and free software released under the GPL-3.0-or-later license.
Acrobits is a privately owned software development company creating VoIP Clients for mobile platforms, based in Prague, Czech Republic.
References
- ↑ "The Zfone Project - FAQ".
- ↑ "Zfone, Getting Started With" . Retrieved June 27, 2011.
- ↑ "Zfone Project, Server Problems" . Retrieved June 27, 2011.
- ↑ RFC 6189.
