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).
In computer networking, peering is a voluntary interconnection of administratively separate Internet networks for the purpose of exchanging traffic between the "down-stream" users of each network. Peering is settlement-free, also known as "bill-and-keep" or "sender keeps all", meaning that neither party pays the other in association with the exchange of traffic; instead, each derives and retains revenue from its own customers.
Voice over Internet Protocol (VoIP), also known as IP telephony, refers to a set of technologies used for voice communication sessions over Internet Protocol (IP) networks, such as the Internet. VoIP enables voice calls to be transmitted as data packets, facilitating various methods of voice communication, including traditional applications like Skype, Microsoft Teams, Google Voice, and VoIP phones. Regular telephones can also be used for VoIP by connecting them to the Internet via analog telephone adapters (ATAs), which convert traditional telephone signals into digital data packets that can be transmitted over IP networks.
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.
A business telephone system is a telephone system typically used in business environments, encompassing the range of 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. Various voice over IP technologies are available on smartphones; IMS provides a standard protocol across vendors.
The next-generation network (NGN) is a body of key architectural changes in telecommunication core and access networks. The general idea behind the NGN is that one network transports all information and services by encapsulating these into IP packets, similar to those used on the Internet. NGNs are commonly built around the Internet Protocol, and therefore the term all IP is also sometimes used to describe the transformation of formerly telephone-centric networks toward NGN.
An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet.
Direct inward dialing (DID), also called direct dial-in (DDI) in Europe and Oceania, is a telecommunication service offered by telephone companies to subscribers who operate private branch exchange (PBX) systems. The feature provides service for multiple telephone numbers over one or more analog or digital physical circuits to the PBX, and transmits the dialed telephone number to the PBX so that a PBX extension is directly accessible for an outside caller, possibly by-passing an auto-attendant.
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).
An Integrated Access Device is a customer premises device that provides access to wide area networks and the Internet. Specifically, it aggregates multiple channels of information including voice and data across a single shared access link to a carrier or service provider PoP. The access link may be a T1 line, a DSL connection, a cable (CATV) network, a broadband wireless link, or a metro-Ethernet connection.
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.
Microsoft Response Point was an advanced software-based telephone system developed by Microsoft. Response Point, a PBX system targeting small businesses with less than 50 employees, was launched in March 2007, with systems available on the market in the fourth quarter of that year. Response Point is VoIP-based, and uses SIP as its signaling and call setup protocol. Response Point supports voicemail and multi-party calling in addition to two party VoIP calls. Response Point features innovative voice recognition technology to manage calls and voice mail. Voicemail messages can, optionally, be sent to e-mail where they can be retrieved and archived. Response Point voice dialing can work with the Response Point phone directory which is currently limited to 1100 contacts per user. Contacts may be imported from the Windows Address Book or Microsoft Outlook. Response Point automatically detects gateways and phones connected to the network.
An IP PBX is a system that connects telephone extensions to the public switched telephone network (PSTN) and provides internal communication for a business. An IP PBX is a PBX system with IP connectivity and may provide additional audio, video, or instant messaging communication utilizing the TCP/IP protocol stack.
IP exchange or (IPX) is a telecommunications interconnection model for the exchange of IP based traffic between customers of separate mobile and fixed operators as well as other types of service provider, via IP based Network-to-Network Interface. IPX is developed by the GSM Association.
Broadvox is a VoIP service provider for business telecommunications. It offers voice and data network solutions for telecommunications, cable, and wireless carriers, as well as ISPs, ITSPs, Over-the-top (OTT) service providers, MVNOs, and various other business partners. Broadvox operates its own nationwide competitive local exchange carrier (CLEC) facilities.
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.
The Internet Telephony Service Providers Association (ITSPA) is a British body representing providers of Internet Telephony services (VoIP).
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.
