Internet Protocol Device Control (IPDC) is a 1998 specification of a communications protocol for voice over Internet Protocol (VoIP) telephony, developed by Level 3 Communications. [1]
Level 3 Communications was an American multinational telecommunications and Internet service provider company headquartered in Broomfield, Colorado. It ultimately became a part of CenturyLink, where Jeff Storey was installed as Chief Operating Officer becoming CEO of CenturyLink one year later in a prearranged succession plan.
IPDC divides the operation of telephony gateways between intelligent call routers in an Internet Protocol (IP) network and simple media gateways at the edge of the IP network and the public switched telephone network (PSTN).
The Internet Protocol (IP) is the principal communications protocol in the Internet protocol suite for relaying datagrams across network boundaries. Its routing function enables internetworking, and essentially establishes the Internet.
A media gateway is a translation device or service that converts media streams between disparate telecommunications technologies such as POTS, SS7, Next Generation Networks or private branch exchange (PBX) systems. Media gateways enable multimedia communications across packet networks using transport protocols such as Asynchronous Transfer Mode (ATM) and Internet Protocol (IP).
The public switched telephone network (PSTN) is the aggregate of the world's circuit-switched telephone networks that are operated by national, regional, or local telephony operators, providing infrastructure and services for public telecommunication. The PSTN consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all interconnected by switching centers, thus allowing 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 and other networks, as well as fixed telephones.
Internet Protocol Device Control was fused with the Simple Gateway Control Protocol (SGCP), a project independently in progress at Bellcore, to form the Media Gateway Control Protocol (MGCP). This group of protocols employs the media gateway control protocol architecture that is also the foundation of MEGACO/H.248, a similar protocol which became a standards-track protocol at the Internet Engineering Task Force (IETF).
The Simple Gateway Control Protocol (SGCP) is a communications protocol used within a voice over Internet Protocol (VoIP) system. It has been superseded by the Media Gateway Control Protocol (MGCP), another implementation of the media gateway control protocol architecture.
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).
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).
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).
The RTP audio/video profile (RTP/AVP) is a profile for Real-time Transport Protocol (RTP) that specifies the technical parameters of audio and video streams. RTP specifies a general-purpose data format, but doesn't specify how encoded data should utilize the features of RTP. An RTP profile specifies these details. The RTP audio/video profile specifies a mapping of specific audio and video codecs and their sampling rates to RTP payload types and clock rates, and how to encode each data format as an RTP data payload, as well as specifying how to describe these mappings using Session Description Protocol (SDP).
Simple Network Management Protocol (SNMP) is an Internet Standard protocol for collecting and organizing information about managed devices on IP networks and for modifying that information to change device behavior. Devices that typically support SNMP include cable modems, routers, switches, servers, workstations, printers, and more.
Network address translation (NAT) is a method of remapping one IP address space into another by modifying network address information in the IP header of packets while they are in transit across a traffic routing device. The technique was originally used as a shortcut to avoid the need to readdress every host when a network was moved. It has become a popular and essential tool in conserving global address space in the face of IPv4 address exhaustion. One Internet-routable IP address of a NAT gateway can be used for an entire private network.
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).
The Skinny Client Control Protocol (SCCP) is a proprietary network terminal control protocol originally developed by Selsius Systems, which was acquired by Cisco Systems in 1998.
Universal Plug and Play (UPnP) is a set of networking protocols that permits networked devices, such as personal computers, printers, Internet gateways, Wi-Fi access points and mobile devices to seamlessly discover each other's presence on the network and establish functional network services for data sharing, communications, and entertainment. UPnP is intended primarily for residential networks without enterprise-class devices.
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.
IBM Sametime is a client–server application and middleware platform that provides real-time, unified communications and collaboration for enterprises. Those capabilities include presence information, enterprise instant messaging, web conferencing, community collaboration, and telephony capabilities and integration. It is sold by the Lotus Software division of IBM.
The Gateway Control Protocol is an implementation of the media gateway control protocol architecture for providing telecommunication services across a converged internetwork consisting of the traditional public switched telephone network (PSTN) and modern packet networks, such as the Internet. H.248 is the designation of the recommendations developed by the ITU Telecommunication Standardization Sector (ITU-T) and Megaco is a contraction of media gateway control protocol used by the earliest specifications by the Internet Engineering Task Force (IETF). The standard published in March 2013 by ITU-T is entitled H.248.1: Gateway control protocol: Version 3.
An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet.
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).
UDP hole punching is a commonly used technique employed in network address translation (NAT) applications for maintaining User Datagram Protocol (UDP) packet streams that traverse the NAT. NAT traversal techniques are typically required for client-to-client networking applications on the Internet involving hosts connected in private networks, especially in peer-to-peer, Direct Client-to-Client (DCC) and Voice over Internet Protocol (VoIP) deployments.
Network address translator traversal is a computer networking technique of establishing and maintaining Internet protocol connections across gateways that implement network address translation (NAT).
Internet Gateway Device (IGD) Standardized Device Control Protocol is a protocol for mapping ports in network address translation (NAT) setups, supported by a certain number of NAT-enabled routers. It is a common communications protocol for automatically configuring port forwarding, and is part of an ISO/IEC Standard rather than an Internet Engineering Task Force standard.
A media server refers either to a dedicated computer appliance or to a specialized application software, ranging from an enterprise class machine providing video on demand, to, more commonly, a small personal computer or NAS for the home, dedicated for storing various digital media. This can also mean that these servers are specialized for media for streaming
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.