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The Use of H.323 in VoIP

The Use of H.323 in VoIP

H.323 was created in November 1996 by the International Telecommunication Union (ITU) in order to enable multimedia conferencing, specifically videoconferencing, through the use of a local area network (LAN). However, it soon became the standard way to transmit voice in Voice over Internet Protocol (VoIP) and wide area networks (WANs). The ITU Telecommunication Standardization Sector's (ITU-T) protocol set marked H.32x specializes in using 3G mobile networks, ISDN, SS7 or PSTN to send communications.
By utilizing a calling model similar to the model used by ISDN, the H.323 protocol implements IP telephony into PBX system networks already in existence, which also includes IP-based PBX. The IP-based PBX is a control element or gatekeeper that provides service to videophones or telephones, including basic service packages and additional features such as hold, pick-up, park, and call transfer. A gatekeeper is one feature in the H.323 system which provides functions such as address resolution, user authentication and admission control. The gatekeeper communicates with the multipoint control unit (MCU), gateway and terminals. These components of the H.323 network work together for a better end products, mainly a higher quality sound. Two terminals are needed for the network to succeed, and a gatekeeper oversees the process to ensure address resolution.

The H.323 network is used in two primary ways. The first is through VoIP, of which it has become a standard, in instances where remote locations are connected through wireless and wired technology over Internet telephony without requiring a VoIP service provider. Videoconferencing, both domestically and internationally, is the second way H.323 is used successfully because it can be used by anyone with a high-speed Internet connection. Users can opt for a desktop videoconferencing system, which is a type of add-on compatible with most PCs, or users can prefer a dedicated videoconferencing system that features all of the necessary components in one device.

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MGCP Follows Master-Slave Protocol Model
A Media Gateway Control Protocol (MGCP) is a Voice over Internet Protocol system used to handle the exchange of information and the management of a multimedia conference session. The exchange of information, known as signaling, is responsible for connecting, controlling and terminating sessions. Therefore, an MGCP protocol is used to set up, maintain and end calls between multiple points.
RTP Delivers Multimedia Data In Real Time
  A Real-time Transport Protocol (RTP) is a protocol standard defined by the Internet Engineering Task Force (IETF) that outlines a management system for programs using real-time transmission of data through mutlicast or unicast network services in multimedia situations. Originally designed by the IETF for videoconferencing for multiple participants, the protocol is largely used in Voice over Internet Protocol applications. Despite its name, the protocol cannot guarantee real-time delivery of data, however, the RTP does compensate for jitters and can detect when data arrives out of sequence, both issues being popular in VoIP communication. In IP telephony, RTP works with a signaling protocol, such as SIP or H.323, in order to set up connections in a network.
SDP Protocols Are Used To Transmit Media Session Information
In 1998, the Internet Engineering Task Force (IETF) published the specification for a Session Description Protocol or SDP as a format that describes parameters for streaming media. The original IETF Proposed Standard was updated in 2006 as RFC 4566. Although the SDP was created as a feature of Session Announcement Protocol (SAP), it can be used with Real-time Transport Protocol, Real-time Streaming Protocol (RTSP), and Session Initiation Protocol, as well as a standalone protocol. Parameter negotiation, session announcement and session invitation are included in the descriptive sessions from the SDP protocol. Rather than transmit data like other types of protocols, an SDP negotiates between media type endpoints, format and properties involved. A session begins when a connection is established, and the session is terminated only after every endpoint is no longer participating.
VoIP Favors IAX Due To Its Flexibility With Codecs, Firewalls
Inter-Asterisk eXchange (IAX) is a communication protocol used to initiate user sessions, similar to an SIP, in order to transmit and control streaming media, especially in Voice over Internet Protocol calls. IAX2 (the newest version) is a preferred method because of its flexibility, as it is compatible with a variety of codecs, and jitters and lag are minimal due to trunking and multiplexing, which means the amount of bandwidth and latency are also kept at minimal levels.
VoIP Favors SIP
Session Initiation Protocol (SIP) is a text-based standard put in place by the Internet Engineering Task Force (IETF) and is the primary protocol for Voice over Internet Protocol services. SIP is similar to a MGCP in that it's a signaling protocol that can create, maintain and terminate sessions in IP-based networks. These sessions include multimedia conferencing and two-way phone calls. SIP protocol can run on User Datagram Protocol (UDP), Stream Control Transmission Protocol (SCTP) and Transmission Control Protocol (TCP) because it was created to function independently from the underlying Transport Layer as an Application Layer.

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