VoIP Gateway from Patton
Voice-over-Internet-Protocol is a packet-based technology for transmitting digitized voice over an Ethernet/IP network. Because many businesses deliver the packetized voice traffic across the Internet to lower the phone bill, VoIP is sometimes called Internet Telephony. VoIP technology, or packet-based voice, underlies such recent developments as unified communication (UC). Related terms include network transformation-referring to carriers that migrate to All-IP network service delivery-as well as real-time communication, since voice transmission is especially sensitive to network issues as delay and jitter (variance in delay).
A VoIP gateway-a.k.a. analog telephone adapter (ATA)-is a networking device that converts a traditional (legacy) phone signal (analog or digital) into a (digitized) packet-based, Internet Protocol (IP) communication stream. The gateway serves as the conversion point between Time Division Multiplexed (TDM) telephone network and an IP-based network such as the Internet or private corporate LAN or WAN.
Traditional business-class telephony services include include PRI, E1 and T1 trunks, plain old telephone service (POTS) interfaces like FXS and FXO (learn more about FXS/FXO technology) as well as BRI and PRI interfaces for integrated services digital network (ISDN). (Glossary of telecom terms.) The gateway converts voice received from the PBX into IP packets suitable for transmission over a data network. Similarly, the gateway re-formats the stream of incoming IP data packets for use by the PBX system.
VoIP Gateways can support numerous use cases (application scenarios) to enable IP telephony implementations. A media gateway may provide several types of bi-directional conversion, including SIP-to-TDM and TDM-to-SIP. Some high-end Voip Gateways may even provide such session border controller (SBC) functions as transcoding, security firewall, failover, and others.
Typical VoIP Gateway Applications
VoIP media gateways can be used to address many different challenges when implementing an IP-based voice system. Three of the most common use cases are summarized below.
PSTN (T1/E1/PRI) to All-IP Voice System
The primary driver for TDM-to-SIP deployment is known as “toll bypass” which reduces a company’s operating costs by avoiding the expensive tolls levied by a public switched telephone network (PSTN). Instead of a Telco trunk, the VoIP gateway connects a legacy business phone system to a much lower-cost SIP trunk, delivered by an Internet telephony service provider (ITSP).
VoIP Provider to Legacy PBX
ITSP or cloud/hosted/virtualized SIP trunking service to Analog POTS or Digital ISDN PBX.
A VoIP gateway can connect a state-of-the-art All-IP communications system (on-premise IP PBX or cloud/hosted/virtualized IP PBX) to a legacy PTSN trunk. You can upgrade a legacy business phone system using an IP media gateway so it can connect to an ITSP over a SIP trunk. SIP trunks are much less costly than expensive toll-based legacy T1 lines.
Migration from Legacy to IP PBX
Going the other way round, in a SIP-to-TDM deployment, the gateway connects a modern SIP communications system with a T1/E1/PRI service provided by a legacy carrier. Keep the existing business phone system up-and-running while transitioning to a new IP based phone system.