Omega Telecom

VoIP – Voice over Internet Protocol - 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 (voice, fax, SMS, voice-messaging) over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS).

SIP – Session Initiation Protocol - is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications.[1] 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).

The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).[2] A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message.

SIP works in conjunction with several other protocols that specify and carry the session media. Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages. SIP is designed to be independent of the underlying transport layer protocol and can be used with the User Datagram Protocol (UDP), the Transmission Control Protocol (TCP), and the Stream Control Transmission Protocol (SCTP). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS). For the transmission of media streams (voice, video) the SDP payload carried in SIP messages typically employs the Real-time Transport Protocol (RTP) or the Secure Real-time Transport Protocol (SRTP).

PSTN – Public Switched Telephone Network - provides infrastructure and services for public telecommunication. The PSTN is the aggregate of the world's circuit-switched telephone networks that are operated by national, regional, or local telephony operators. These consist of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all interconnected by switching centers which allow most telephones to communicate with each other.

Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital in its core network and includes mobile[1] and other networks, as well as fixed telephones.

The technical operation of the PSTN adheres to the standards created by the ITU-T. These standards allow different networks in different countries to interconnect seamlessly. The E.163 and E.164 standards provide a single global address space for telephone numbers. The combination of the interconnected networks and the single numbering plan allow telephones around the world to dial each other.

ISDN – Integrated Services Digital Network - is a set of communication standards for simultaneous digital transmission of voice, video, data, and other network services over the digitalised circuits of the public switched telephone network. Work on the standard began in 1980 at Bell Labs and was formally standardized in 1988 in the CCITT "Red Book". By the time the standard was released, newer networking system with much greater speeds were available, and ISDN saw relatively little uptake in the wider market. One estimate suggests ISDN use peaked at a worldwide total of 25 million subscribers at a time when 1.3 billion analog lines were in use.[3] ISDN has largely been replaced with digital subscriber line (DSL) systems of much higher performance.

Prior to ISDN, the telephone system consisted of digital links like T1/E1 on the long-distance lines between telephone company offices and analog signals on copper telephone wires to the customers, the "last mile". At the time, the network was viewed as a way to transport voice, with some special services available for data using additional equipment like modems or by providing a T1 on the customer's location. What became ISDN started as an effort to digitize the last mile, originally under the name "Public Switched Digital Capacity" (PSDC). This would allow call routing to be completed in an all-digital system, while also offering a separate data line. The Basic Rate Interface, or BRI, is the standard last-mile connection in the ISDN system, offering two 64 kbit/s "bearer" lines and a single 16 kbit/s "delta" channel for commands and data.

Although ISDN found a number of niche roles and some wider uptake in specific locales, the system was largely ignored and garnered the industry nickname "innovation subscribers didn't need."[4] It found a use for a time for small-office digital connection, using the voice lines for data at 64 kbit/s, sometimes "bonded" to 128 kbit/s, but the introduction of 56 kbit/s modems undercut its value in many roles. It also found use in videoconference systems, where the direct end-to-end connection was desirable. The H.320 standard was designed around its 64 kbit/s data rate. The underlying ISDN concepts found wider use as a replacement for the T1/E1 lines it was originally intended to extend, roughly doubling the performance of those lines.

PBX – Private Branch eXchange - is a telephone system within an enterprise that switches calls between users on local lines, while enabling all users to share a certain number of external phone lines. In contrast to a public switched telephone network (PSTN), the main purpose of a PBX is to save the cost of requiring a line for each user to the telephone company's central office.

Used as a business telephone system or private telephone network, a PBX is owned and operated by the enterprise rather than the telephone company -- the telephone company of which may be considered a supplier or service provider, however. Private branch exchanges used analog technology originally. Today, PBXs use digital technology -- digital signals are converted to analog for outside calls on the local loop using plain old telephone service (POTS). Nonetheless, PBXs can include network switching systems that accommodate analog phones into the enterprise's digital PBX system.

IVR – In telephony, Interactive Voice Response - is a technology that allows humans to interact with a computer-operated phone system through the use of voice and DTMF tones input via a keypad. In telecommunications, IVR allows customers to interact with a company's host system via a telephone keypad or by speech recognition, after which services can be inquired about through the IVR dialogue. IVR systems can respond with pre-recorded or dynamically generated audio to further direct users on how to proceed. IVR systems deployed in the network are sized to handle large call volumes and also used for outbound calling as IVR systems are more intelligent than many predictive dialer systems.

IVR systems can be used for mobile purchases, banking payments, services, retail orders, utilities, travel information and weather conditions. A common misconception refers to an automated attendant as an IVR. The terms are distinct and mean different things to traditional telecommunications professionals—the purpose of an IVR is to take input, process it, and return a result, whereas that of an automated attendant is to route calls. The term voice response unit (VRU) is sometimes used as well.

DID – Direct Inward Dialing - 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.[1][2] 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.