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1. Introduction
Product Overview
The stand-alone VoIP Gateway carries both voice and facsimile over the IP network. It supports
SIP industry standard call control protocol to be compatible with free registration services or VoIP
service providers' systems. It works in two different modes: UA (User Agent) or Server. As a
standard user agent, it is compatible to all well-known Soft Switches and SIP proxy servers.
While running the optional server software, the gateway can be configured to establish a private
VoIP network over the Internet without a 3
The gateway can be seamlessly integrated to existing network by connecting to a phone set,
PBX, key telephone system, fax machine or PSTN line. With only a broadband connection such
as ADSL bridge/router, Cable Modem or leased line router, it allows you to gain access to voice
and fax services over the IP in order to reduce the cost of international and long distance calls.
In addition, the in-built 4 ports Ethernet switch supports comprehensive Internet gateway
functions to accommodate other PCs or IP devices to share the same broadband stream. QoS
function allows voice and data traffic to flow through where voice traffic is transmitted in the
highest priority. With TOS bit enabled, it guarantees voice packets to have first priority to pass
through a TOS enabled router.
With the support of DDNS, it makes the gateway reachable by its domain name where the ISP
dynamically assigns the IP address. It helps users to host a web site or mail server in a PPPoE
or DHCP network. By enabling the CDR function & setting up a simple server, administrators are
allowed to log in and view all call records such as call duration, time and date of calls and
latency.
The gateway can be assigned with a fixed IP address or by DHCP, PPPoE. It adopts the G.711,
G.726, G.729A or G.723.1 voice compression format to save the network bandwidth while
providing real-time and toll quality voice. In addition, in the event that the power supply fails or
Internet connection is lost, the gateway can automatically divert the FXS end to the PSTN
network on the FXO port so users can still use the conventional PSTN line to make calls. This
feature is especially useful while dialing emergency calls (i.e. 911)
rd
party SIP Proxy Server.
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