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TechnologyVoice over Internet Protocol (VoIP) is a technology that allows voice signals to be transmitted as data packets over any IP-based network such as the Internet rather than the public switched telephone network (PSTN). Traditional telephone networks employ circuit-switched technology for the voice transmission, and thus requires a dedicated voice channel for each conversation. VoIP network, on the other hand, employs the packet-switched technologies for the voice transmission. Analog voice signal is converted to digital format and then compressed/packetized into IP packets for transmission over the network. No dedicated channel is required for each conversation. The H.323 standard is one of the most widely adopted standard for the transmission of real-time audio, video, and data communications over packet-based networks. It specifies the components, protocols, and procedures providing multimedia communication over packet-based networks such as the Internet. H.323 can be applied in a variety of mechanisms - audio only (IP telephony); audio and video (videotelephony); audio and data; and audio, video and data. H.323 can also be applied to multipoint-multimedia communications. The gatekeeper is the brain of an H.323 multimedia network. Although not absolutely required in an H.323 network, it performs essential control, administrative and managerial functions required to maintain the integrity of networks in an enterprise or carrier environment. A gatekeeper, such as Linksoft's SoLink Control Server, serves to manage all terminals, gateways, and MCUs within an H.323 zone. A zone includes at least one terminal and may include gateways or MCUs. A zone has one and only one active gatekeeper at a time. A zone may be independent of network topology and may be comprised of multiple network segments that are connected using routers or other devices. A gateway connects two dissimilar networks. An H.323-based VoIP gateway provides connectivity between an H.323 network and a non–H.323 network. For example, a gateway can connect and provide communication between an H.323 terminal and the public switched telephone network (PSTN). This connectivity of dissimilar networks is achieved by translating protocols for call setup and release, converting media formats between different networks, and transferring information between the networks connected by the gateway. A gateway is not required, however, for communication between two terminals on an H.323 network. Some sample VoIP gateways are Antek, Quintum, and Extreme. Used for real-time bidirectional multimedia communications, an H.323 terminal can either be a PC (e.g. softphone) or a stand-alone device running an H.323 stack and multimedia applications (e.g. IP phone). An H.323 terminal should support audio communications and can optionally support video or data communications. Because the basic service provided by an H.323 terminal is audio communications, an H.323 terminal plays a key role in VoIP services. A Multipoint Control Unit (MCU) enables three or more terminals and Gateways to participate in a multipoint conference. It may also connect two terminals in a point-to-point conference which may later develop into a multipoint conference. The MCU consists of two parts: a mandatory Multipoint Controller and optional Multipoint Processors. In the simplest case, an MCU may consist only of an MC with no MPs. An MCU may also be brought into a conference by the Gatekeeper without being explicitly called by one of the endpoints. Similar to H.323, the Session Initiation Protocol (SIP) is another widely adopted standard for the transmission of real-time audio, video, and data communications over packet-based networks. It is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists, and IP Centrex services. SIP supports a dynamic registration mechanism. SIP end-points send REGISTER requests to a Registration server to register their physical location with the SIP service of record. The REGISTER requests have an expiry and the end-points refresh their registration by sending REGISTER requests periodically, as determined by the end node configuration. Note that an end-point must be registered in order to receive incoming call from the server. The REGISTER requests tell the proxy/registrar where to route the calls for the given user name. Calls to unregistered destinations result in a SIP error, usually 480 Temporarily Unavailable, which will usually produce a fast-busy tone for the calling party (depending on their phone/software). SIP proxies are elements that route SIP requests to user agent servers and SIP responses to user agent clients. A request may traverse several proxies on its way to a UAS. Each will make routing decisions, modifying the request before forwarding it to the next element. Responses will route through the same set of proxies traversed by the request in the reverse order. A private branch exchange (PBX) is a proprietary telephone system that resides within an enterprise network and connects internal telephone extensions to each other and the public switch telephone network (PSTN). An IP PBX, on the other hand, delivers PBX-like services, but over an IP-based netweork rather than circuit switched networks. IP-PBX, such as SoLink IP-PBX Communication System, allows the convergence of voice and data network, thus provides flexibility as an enterprise grows, and can also reduce long-term operation and maintenance costs. |
