VoIP communications systems can provide a wonderful array of features and functions within an organization or community. However, much of the world is still talking to each other over land-line PSTN systems. From the earliest days of internet phone calls it was obvious the VoIP and PSTN systems must inter-operate.
When a VoIP telephone user makes a call to someone the call initiates a series of processes which include identifying the intended party, establishing the connection procedure, digitizing the analogue voice signal, encoding the resulting data into a stream of packets which are then shipped out across the internet to the other party. In rough order of their application some of the protocols used to accomplish this are SIP, G7.11, UDP and RTP. At the other end of the call the process is reversed, applying the same protocols to deliver a reassembled stream of data packets to a digital-to-analogue converter. If the receiving party is also using a VoIP handset, the connection is established and the conversation continues until either party hangs up.
If the called (receiving) party is on a land-line provided by one of the international PSTNs, another step is required. The VoIP call must be converted into something the PSTN system recognizes before it will be forwarded to the intended party. An analogue telephone adapter (ATA) provides the conversion between IP packets and analogue voice signals and manages the call parameters. A PSTN gateway provides ATA functions and usually offers additional management services such as QoS, address translation or billing information.
“SIP/VoIP to PSTN termination” describes a phone call originated on an internet phone, using Session Initiation Protocol, which connects (terminates) at a land-line phone. The call could be from a single VoIP handset connected to the internet which uses a PSTN gateway service to convert the signals and carry the call to the land-line phone number. Or the configuration can be a LAN, with any number of VoIP phones, which employs a VoIP to PSTN service provider when any of the LAN phones calls a party on a wired network (PSTN). These providers have high-capacity connections in one or more of the national PSTN systems, such as T1 or E1 cables.
Such services organize their equipment to maximize the quality and capacity of the connections between internet and traditional phone systems. One method is to establish multiple entrance points into the PSTN. Having geographically dispersed entrance points (points of presence or POP) allows the service provider to route calls through the closest PSTN entrance or address congestion by using more than a single POP.
The integration of VoIP and PSTN services can permit an organization to “tune” their communication system to match more of their communication needs.