VoIP Gateway

Overview

A VoIP gateway bridges the old telephone world and the IP network. On one side it speaks to analog phones, fax machines, and PSTN trunk lines, or to a digital E1/T1 span; on the other it carries those calls as RTP media streams set up by SIP. Inside, a host processor runs the call-control stack while a dedicated voice DSP does the per-channel signal work: encoding and decoding audio, cancelling echo, and detecting dial digits. The result lets an organization keep its existing handsets and trunks while connecting them to a modern IP PBX or carrier.

How it works

Analog and digital lines land on the Telephony Interface Module. Each FXS port drives a SLIC that supplies battery and ring voltage to an attached phone and senses when it goes off-hook, while FXO Port interfaces present a phone-like termination to a PSTN trunk. Every line couples through a Line Transformer for isolation and surge tolerance. A digital span instead enters the E1 Port and is decoded by the E1/T1 Framer, which recovers the individual voice timeslots.

Audio from these interfaces flows to the Mainboard, where the Voice DSP compresses each channel with the selected codec, removes echo, and packetizes the result. The Compute SoC Module runs the SIP/H.323 stack that negotiates calls and maps PSTN endpoints to IP destinations, with the Microcontroller handling supervision and the line-state housekeeping.

Packetized calls leave through the Ethernet Module, where magnetics-isolated ports carry the RTP and SIP traffic onto the LAN. The Front Panel Indicators shows per-channel status, and administrators configure dial plans and trunks through the Management Port, with the whole board powered from the internal Power Supply.