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Federated VoIP is a form of packetized voice telephony that uses voice over IP between autonomous domains in the public Internet without the deployment of central virtual exchange points or switching centers for traffic routing. Federated VoIP uses decentralized addressing systems, such as ENUM, for location and identity information of participants and implements secure, trusted communications (TLS) for identify verification.
The traditional telephone networks are based on the principle of point-to-point communications, for example, an office building sends and receives phone calls over an ISDN line to the telephone exchange. Early Voice over IP deployments have copied this model, sending phone calls over SIP Trunking (a virtual equivalent of ISDN) to an Internet telephony service provider (ITSP) (a virtual telephone exchange). Such paradigms have striven to emulate many of the aspects of traditional telephony, such as the telephone number.
Although the use of virtual exchanges (soft-switches) and ITSPs has reduced the cost of telephony, they have retained many constraints of traditional telephone systems. The fundamental principle of federated voice over IP is that it removes this concept of a virtual exchange, just as email has eliminated the concept of a centralized system of post offices.
Although there is no formal specification for federated VoIP as a whole, the Session Initiation Protocol (SIP) and Extensible Messaging and Presence Protocol (XMPP/Jabber) both specify similar and compatible ways of implementing federated VoIP. In particular:
Implementation of federated VoIP involves a number of initiatives:
To achieve maximum success, most deployments involve both SIP and Jabber, to ensure connectivity to any other party that supports only one or the other. This is not so difficult in practice, as a network that uses SIP internally can operate a Jabber gateway, and a network that uses Jabber internally can operate a SIP gateway. In either case, the SIP and Jabber addresses are in the form of email addresses and are almost always identical, so there is no extra effort required for the user of the service.
Various open-source VoIP server products provide detailed instructions on how to implement federated VoIP. [1] [2]
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).
Telephony is the field of technology involving the development, application, and deployment of telecommunication services for the purpose of electronic transmission of voice, fax, or data, between distant parties. The history of telephony is intimately linked to the invention and development of the telephone.
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for voice calls, the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.
Extensible Messaging and Presence Protocol is an open communication protocol designed for instant messaging (IM), presence information, and contact list maintenance. Based on XML, it enables the near-real-time exchange of structured data between two or more network entities. Designed to be extensible, the protocol offers a multitude of applications beyond traditional IM in the broader realm of message-oriented middleware, including signalling for VoIP, video, file transfer, gaming and other uses.
Telephone number mapping is a system of unifying the international telephone number system of the public switched telephone network with the Internet addressing and identification name spaces. Internationally, telephone numbers are systematically organized by the E.164 standard, while the Internet uses the Domain Name System (DNS) for linking domain names to IP addresses and other resource information. Telephone number mapping systems provide facilities to determine applicable Internet communications servers responsible for servicing a given telephone number using DNS queries.
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints, such as customary telephone sets, destinations on the public switched telephone network (PSTN), and devices or services on voice over Internet Protocol (VoIP) networks. Its name comes from the asterisk (*) symbol for a signal used in dual-tone multi-frequency (DTMF) dialing.
A business telephone system is a multiline telephone system typically used in business environments, encompassing systems ranging in technology from the key telephone system (KTS) to the private branch exchange (PBX).
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is a standardised architectural framework for delivering IP multimedia services. Historically, mobile phones have provided voice call services over a circuit-switched-style network, rather than strictly over an IP packet-switched network. Alternative methods of delivering voice (VoIP) or other multimedia services have become available on smartphones, but they have not become standardized across the industry. IMS is an architectural framework that provides such standardization.
An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet.
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).
SipXecs is a free software enterprise communications 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 Open Settlement Protocol (OSP) is a client/server protocol used by Internet service providers to exchange authorization, accounting, and usage information to support IP telephony. Open Settlement Protocol is implemented in voice telephony gateways such as softswitches, H.323 multimedia conferencing gateways, and Session Initiation Protocol (SIP) proxies.
The SIP URI scheme is a Uniform Resource Identifier (URI) scheme for the Session Initiation Protocol (SIP) multimedia communications protocol. A SIP address is a URI that addresses a specific telephone extension on a voice over IP system. Such a number could be a private branch exchange or an E.164 telephone number dialled through a specific gateway. The scheme was defined in RFC 3261.
H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.
Snom Technology GmbH is a German company which manufactures Voice over Internet Protocol (VoIP) telephones, based on the IETF standard Session Initiation Protocol (SIP). Snom's products are targeted at the small- to medium-sized business sector, home offices, Internet service providers, carriers, and original equipment manufacturers. The company, founded in 1996 and headquartered in Berlin, is a wholly owned subsidiary of VTech Holdings Limited, since 2016.
A softphone is a software program for making telephone calls over the Internet using a general purpose computer rather than dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other computer and allows the user to place and receive calls without requiring an actual telephone set. Often, a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a handset, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC or with a USB phone.
SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. Most unified communications applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard.
VaxTele SIP Server SDK is a complete development toolkit, which allows software vendors and Internet telephony service providers (ITSP) to develop SIP Server and (SIP) Session Initiation Protocol based VoIP systems for Microsoft Windows to install computer to computer voice chat, chat rooms, IVR systems, call center services, calling card services, dial/receive computer to PSTN and mobile phone calling services.