Voice peering, also called VoIP peering, refers to the forwarding of calls from one ITSP to another ITSP directly using VoIP technology. The call is not forwarded over the PSTN and this leads to costs savings (no call charges) and better quality because there is no transcoding between the VoIP cloud and the PSTN, and then back from the PSTN to the next VoIP cloud. VoIP peering may occur on Layer 2 basis, i.e. a private network is provided, and carriers connected with it manage peering between one another, or on a layer 5 basis, i.e. peering occurs on open networks, with routing and signaling managed by a central provider.
Transcoding is the direct digital-to-digital conversion of one encoding to another, such as for movie data files, audio files, or character encoding. This is usually done in cases where a target device does not support the format or has limited storage capacity that mandates a reduced file size, or to convert incompatible or obsolete data to a better-supported or modern format.
Voice peering can occur on a bilateral or multilateral basis. Bilateral peering does not scale when many service providers seek to interconnect and peer with one another. Standards on Multilateral, layer 5 peering are under development by the IETF working group on VoIP Peering, SPEERMINT.
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.
Distributed Universal Number Discovery (DUNDi) is a VoIP routing protocol that provides directory services for Asterisk systems. With DUNDi peered nodes share dialplan information with each other. The protocol does not actually carry any calls, but rather provides addressing information.
The iNum Initiative is a project started in November 2008 by Voxbone to create a global dial code for IP communications. Voxbone is a Belgian company specializing in wholesale telephone numbers for VoIP applications. The International Telecommunication Union allocated a portion of the non-geographic "country code" +883 to this initiative. Specifically, iNums are 15-digit telephone numbers in the +883 5100 area code.
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Related Research Articles
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks 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 users of each network. The pure definition of 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 called IP telephony, is a methodology 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 public Internet, rather than via the public switched telephone network (PSTN).
A softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, across a telecommunication network or the public Internet, entirely by means of software running on a general-purpose computer system. Most landline calls are routed by purpose-built electronic hardware; however, soft switches using general purpose servers and VoIP technology are becoming more popular.
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is an 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 to provide such standardization.
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 a private branch exchange (PBX) system. 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, instead of 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.
VoIP User was a community driven and financed SIP based VoIP network. The projects aim is to introduce people to the concept of VoIP by allowing members to experiment with SIP and IAX2 devices.
Origination in VOIP telephony refers to calls that originate in the PSTN public switched telephone network and are carried to their destination over the Internet.
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.
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 enables the end point’s PBX to send and receive calls via Internet. As SIP is applied for the signalling protocol for multiple real-time application, SIP trunk is able to control voice, video and messaging applications.
It is also 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.
