Call origination, also known as voice origination, refers to the collecting of the calls initiated by a calling party on a telephone exchange of the PSTN, and handing off the calls to a VoIP endpoint or to another exchange or telephone company for completion to a called party.
The calling party is a person who initiates a telephone call. The person who, or device that, receives a telephone call is the called party.
A telephone exchange is a telecommunications system used in the public switched telephone network or in large enterprises. An exchange consists of electronic components and in older systems also human operators that interconnect (switch) telephone subscriber lines or virtual circuits of digital systems to establish telephone calls between subscribers.
A telephone company, also known as a telco, telephone service provider, or telecommunications operator, is a kind of communications service provider (CSP) that provides telecommunications services such as telephony and data communications access. Many telephone companies were at one time government agencies or privately owned but state-regulated monopolies. The government agencies are often referred to, primarily in Europe, as PTTs.
In the VoIP world, the opposite of call origination is call termination, where a call initiated as a VoIP call is terminated to the PSTN.
The term is often used in referring to a VoIP trunking service that provides the ability for calls to originate on the PSTN and be delivered to a customer's VoIP endpoint. It is often referred to as "DID" (for Direct Inward Dialing) which more properly refers to a service entirely within the PSTN where individual PSTN numbers can terminate on a customer's PBX.
In telecommunications, trunking is a method for a system to provide network access to many clients by sharing a set of lines or frequencies instead of providing them individually. This is analogous to the structure of a tree with one trunk and many branches. Examples of this include telephone systems and the two-way radios commonly used by police agencies. Trunking, in the form of link aggregation and VLAN tagging, has been applied in computer networking as well.
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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).
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 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).
Enhanced 911, E-911 or E911 is a system used in North America to automatically provide the caller's location to 911 dispatchers. 911 is the universal emergency telephone number in the region. In the European Union, a similar system exists known as E112 and known as eCall when called by a vehicle.
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 telephone call is a connection over a telephone network between the called party and the calling party.
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.
An analog telephone adapter (ATA) is a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephony network.
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).
In telecommunications business, a white route is a route where both source and destination are legal termination.
This is opposed to a black route, which is a route that is illegal in both ends. Also common in telecom is the term grey route, which defines a route that is legal for one country or the party on one end, but illegal on the alternative end.
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.
IPKall was a public switched telephone network to voice over IP call forwarding service. Users were able to register with the service to obtain a phone number chosen from several Washington State area codes and have all calls to that number forwarded to their Session Initiation Protocol or Inter-Asterisk eXchange uniform resource identifier, including an Asterisk server.
Wideband audio, also known as wideband voice or HD voice, is high definition voice quality for telephony audio, contrasted with standard digital telephony "toll quality". It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 80 Hz to 14 kHz but traditional, voiceband or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio relaxes the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz or even up to 22 kHz. In addition, some wideband codecs may use a higher audio bit depth of 16-bits to encode samples, also resulting in much better voice quality.
The Media Gateway Control Protocol (MGCP) is a signaling and call control communications protocol used in voice over IP (VoIP) telecommunication systems. It implements the media gateway control protocol architecture for controlling media gateways on Internet Protocol (IP) networks connected to the public switched telephone network (PSTN). The protocol is a successor to the Simple Gateway Control Protocol (SGCP), which was developed by Bellcore and Cisco, and the Internet Protocol Device Control (IPDC).
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.
The media gateway control protocol architecture is a methodology of providing telecommunication services using decomposed multimedia gateways for transmitting telephone calls between an Internet Protocol network and traditional analog facilities of the public switched telephone network (PSTN). The architecture was originally defined in RFC 2805 and has been used in several prominent voice over IP (VoIP) protocol implementations, such as the Media Gateway Control Protocol (MGCP) and Megaco (H.248), both successors to the obsolete Simple Gateway Control Protocol (SGCP).
A VoIP gateway is a gateway device that uses Internet Protocols to transmit and receive voice communications (VoIP). The general term is ambiguous and can mean many different things. There are many such devices. They are quickly becoming the most common type of voice phone service in many areas.
