|VoIP phone system|
|Sunday, 12 November 2006|
Internet Protocol (IP) telephony or voice over IP (VoIP) generally relates to transmission of voice calls or voice information over packet-switched data networks that use IP as a datagram protocol. Voice over Internet Protocol (VoIP) is a technology that makes a phone call, but instead of using a circuit based system such as the telephone network, utilizes packetized data transmission techniques implemented in the Internet. The Internet Protocol (IP) is a protocol that defines the addressing of packets of information that can be transmitted over the Internet. Internet Protocol is a part of the TCP/IP family of protocols and tracks the Internet address of nodes, routes outgoing messages, and recognizes incoming messages. Such a data network may be the Internet or a corporate intranet, or any TCP/IP network. The Internet Protocol (IP) is one of the most popular packet communication and networking protocols being used today. It finds use both on the Internet and in wide area networks (WANs) and local area networks (LANs), such as asynchronous transfer mode (ATM) and Ether networks. Generally, the Internet Protocol establishes the nature and length of the packets and provides addressing information used by the various switches and routers to direct each individual packet to its intended destination. VoIP can be defined as the transport of telephony calls over an IP network. In contrast to traditional telephony, where an end-to-end circuit is set up between two endpoints, IP telephony uses the internet protocol to transmit voice packets over an IP network. Unlike the circuit-switched PSTN, packet-switched IP networks provide virtual circuit connections between users. Bandwidth is shared for improved utilization of network capacity, leading to lower costs for network users. A VoIP application segments the voice signals traffic into frames and stores them in voice packets. Voice information to be transmitted across an IP network is converted into digital data and broken up into multiple, discrete packets. The voice packets are transported via the network using any conventional multimedia (i.e., voice, video, fax, and data) protocol. Individual packets may travel over different network paths to reach the final destination where the packets are reassembled in the proper sequence to reconstruct the original voice information. A primary difference between circuit-switching and packet-switching is that, instead of allocating circuits for data streams, packet-switching breaks a data stream (such as a voice data stream, computer data stream, etc.) into individually-routable chunks of data (packets). Completely VoIP calls can be set up between two packet-networked computers or devices, as long as the calling endpoint has a way to determine the other's IP address.
Packet switched networks and related devices are becoming very efficient for voice communications. Packet-switched IP networks provide shared, virtual circuit connections between users. Voice information to be transmitted across an IP network is converted into digital data and broken up into multiple, discrete packets. Individual packets may travel over different network paths to reach the final destination where the packets are reassembled in the proper sequence to reconstruct the original voice information. VoIP applications can be installed on personal computers), other devices connected to the Internet, or on translation servers such as Internet-to-Telephone gateways. Voice data is transmitted through a network as packets. When a voice call is originated using the traditional telephone technology (circuit switching), the voice data is converted to such packets by a network device called a gateway. Such gateways are considered to be endpoints of the network in question. In a Voice over Internet Protocol call, an originating voice gateway quantizes an input audio stream into packets that are placed onto a packet network and routed to a destination voice gateway. The destination voice gateway decodes the packets back into a continuous digital audio stream that resembles the input audio stream. Streams of packets typically enter the network from packet switching edge devices or gateways which serve as portals to the interconnected web of routers comprising the network. A codec uses a compression/decompression algorithm on the quantized digital audio stream to reduce the communication bandwidth required for transmitting the audio packets over the network. In some VoIP phone systems, a gateway takes the voice communication from a traditional PBX or telephone switch, compresses it, packetizes it, converts it to IP format and routes it across the network to the destination. Another gateway at the destination receives the IP packet and reverses the operation to convert it back into the format needed for transmission to the receiving phone system. Gateways are key components in every IP telephony infrastructure. The IP network and the PSTN are connected through a gateway. The gateway converts a PCM (pulse code modulation) voice data applied from the PSTN into a packet to transfer it to the IP network. Conversely, when the packet is received from the IP network, the corresponding packet is converted into PCM voice data to be transferred to the PSTN. Gateways offer the advantage of IP telephony by bridging between the traditional telephone network and the Internet. For real-time communications through the Internet, gateways provide audio conversion and perform call setup and termination. The gateways may be implemented as special hardware boxes or devices or as software modules running on network servers.
VoIP phone systems have recently attracted wide interest because they offer significant advantages over conventional circuit-switched telephone communications. VoIP systems can handle more calls, do not need a separate switched circuit for each call, do not require a specified amount of bandwidth per call, and do not require a large number of geographically distributed central call switching offices, as with the public switched telephone network. The primary benefit in using VoIP over the public switched telephone network (PSTN) is the substantial savings that can be realized in local and particularly long distance telephone bills. Through deployment of IP telephony international and long-distance telephone service providers can offer Internet phone calls to customers with software for multimedia personal computer. The benefits of IP telephony are currently limited by the marginal support for Quality of Service (QoS), inadequate traffic management and the lack of security. For a voice call using VoIP to have acceptable quality, a number of Quality of Service ("QoS") characteristics have to be met, such as minimum throughput, maximum delay, maximum delay variation, and packet loss. The Quality of Service (QoS) of VoIP calls can degrade due to congestion on the packet network or failure of network processing nodes in the packet network. A disadvantage of VoIP networks is the variability of the quality of the signal received at the destination as determined by changing network conditions. The received signal quality depends on a large number of variable network factors such as packet loss, packet latency, queuing delay, and bandwidth availability. These network factors will vary depending on the volume of network traffic and the location of the destination. In packet switched IP networks, large volumes of data being transmitted are first divided into packets of a fixed or more often a variable length. The assembly of these packets entails the creation of a header having at least a packet sequence number, a source address, a destination address and the packet size, contained therein. The individual packets containing the header and data are then transmitted, usually to a gateway server, and then to routers in the case of the Internet. The routers take the data and then transmit it to routers located closer to the ultimate destination, taking into consideration traffic conditions, until the final destination is reached. The number of packets assembled and transmitted is directly dependent on the volume of data being transmitted. The route each packet takes to the destination may vary from packet to packet. The number of routers a particular packet must pass through may vary based on the route taken and traffic conditions.