VoIP (voice over Internet Protocol or IP) is a technology that enables voice communications over the Internet through the compression of voice into data packets that can be efficiently transmitted over data networks and then converted back into voice at the other end. Data networks, such as the Internet or local area networks (LANs), have always utilized packet-switched technology to transmit information between two communicating terminals (for example, a PC downloading a page from a web server, or one computer sending an e-mail message to another computer). The most common protocol used for communicating on these packet-switched networks is Internet protocol, or IP.
VoIP allows for the transmission of voice along with other data over these same packet-switched networks and provides an alternative to traditional telephone networks, which use a fixed electrical path to carry voice signals through a series of switches to a destination.
Session Initiation Protocol (SIP) is a call control protocol that establishes and terminates "media sessions." Examples of media sessions include phone calls, e-mail, streamed video, and instant messaging. SIP standardizes these communications by adding a protocol that is independent of the underlying network and infrastructure. It is not dependent on ports, pipes, hardware or software, but allows all of these components to talk with each other and media to be controlled independently from the network.
SIP is the packet voice protocol of choice by major IP voice initiatives such as the G3 wireless consortium and Microsoft. A highly flexible protocol, SIP specifies the basic and supplementary services to create, modify, and delete these multimedia sessions or calls. By utilizing SIP, we enable customers to use applications regardless of the client's network or access type.
SIP's foundation in HTTP allows it to integrate with Web, e-mail, and other data applications using a URL format, providing a highly flexible protocol. SIP's ease of use and flexibility pave the way for new services and capabilities in the future.
IP Trunking allows businesses with IP PBXs to extend the benefits of Voice over IP (VoIP) convergence from their LAN to the WAN and the Public Switched Telephone Network (PSTN).
It eliminates the need for expensive TDM gateways and trunks, helps to increase network efficiencies and cost-savings by converging voice and data traffic on your WAN, and retains all the features and services of your existing TDM connections.
In addition to these benefits, IP Trunking is specifically designed to allow you to connect multiple locations over the same VoIP trunk while maintaining local telephone numbers, calling plans and 911 services.
An IP (Internet Protocol) PBX (Private branch exchange) is a business telephone system designed to deliver voice over a data network and interoperate with the normal Public Switched Telephone Network (PSTN).
