Airlive VOIP-400R SERIES DATASHEET

VoIP-400R series

VOIP Gateway Router

he AirLive VoIP-400R series are designed for
the best versatility in a IP sharing environment.
The Gateways come equipped with a built-in 4-

T

port NAT router that ensure the quality of Voice
transmission while allowing your entire network to
share one internet connection. It comes in varies 4­port FXO/FXS configuration and allow up to 4
simultaneous voice/fax communications.

High Quality Internet Telephony

With AirLive eLive series of VoIP gateways, the dream of making unlimited long distance phone call and fax
using conventional phone is realized. Once setup, the VoIP gateway will automatically take call request
from conventional telephone handset or through company's PBX network, then make connections through
Internet with the remote gateway to make traditional phone calls. All this process is transparent to users.
They will enjoy the freedom to communicate freely without worrying about the telephone charge.

New Dimension of Communication freedom

The latest Voice over IP technology completely open a new dimension of freedom in real time
communication. For corporate environment, VoIP allows branch offices to keep in touch as effectively as in
the local office. Using the gateway technology, company can even provide remote support to oversea
customers without incurring extra cost. For family or friends living far away from each other, VoIP make the
world much closer. Best of all, all these technology does not require you to operate the phone call any
differently. All your need is the VoIP gateways and your ordinary telephone set.

Quality and Standard Compliance

Equipped with high performance RISC and DSP processors running on real-time OS. The VoIP-400R
ensure high quality voice transmission. Furthermore, its support of H.323 protocol ensure compatibility with
most VoIP equipments including Cisco's ATA186 and IP phones. QoS Support for VoIP packets maintain the
quality of transmission for VoIP packets.

Simplified Configuration

The AirLive VoIP gateway can be configured through web browser, Telnet, or console port. Each model
comes with detailed instruction manual and set up example to help users start their VoIP operation quickly.
Having the broadband built-in also means quantity of service and dynamic IP issues are solved
automatically. It is an all-in-one solution for the VoIP and Broadband sharing environment.

Configurations

· VoIP 422R: Metal housing, 2FXO + 2FXS, 4 LAN ports (H.323 only)

· VoIP 404R: Metal housing, 4FXO, 4 LAN ports (H.323 only)

· VoIP 440R: Metal housing, 4FXS, 4 LAN ports (H.323 only)

FXO: The FXO ports connect to the PSTN line (analogue telephone line from the telephone operator) or
company's PBX line. It allows the VoIP device to communicate with the ordinary telephone system. Having
the PXO port is necessary if you want your device to serve as a gateway for remote caller to reach your local
telephone line.

FXS: The FXS ports connect to your ordinary telephone or FAX. Through your telephone or FAX, your can
dial out through the gateway to other VoIP gateway or IP Phone.

WAN port: VoIP gateway has a WAN port that connects to the Internet directly or to a broadband router.
User also setup the web or telnet configuration through the WAN port.

Call Forward/Transfer: This function require FXO port. It allow remote PSTN users to dial into the FXO
port and use the gateway's dial-plan to make VoIP calls.

*The FXO or FXS port works with analogue telephone standard only.
If you have ISDN phone line or digital PBX, separate digital to analogue converter is required.

Specification

Key Feature

· Effective QoS scheme allowing

simultaneous VoIP and Data
communication

· 4 channel of real-time voice and fax

simultaneously

· Built-in TFTP and Flash memory for

firmware upgrade via network

· Call forward and Call Transfer

· Support multiple VoIP Gateway on a LAN

port

· Dynamic IP/ PPPoE client support

· NAT Routing Function

· Support dynamic bandwidth management

and QoS

Real-time FAX Relay:

· T.3 8

Speech

· Compression algorithms: ITU G. 711,

G.723.1, and G.729A/B.

· Hybrid echo cancellation G.168 (16 ms)

Auto switch between Fax and voice

· DTMF tone detection/regeneration

· Channel: four channels per module

· Comfort Noise Generation (CNG)

· User programmable Call Progress

detection/generation

· Voice Activity Detection (VAD)

· User programmable Gain Control