Planet Technology VIP-161SW User Manual

Wireless analog telephone adapter, Vip-161sw, Key feature data sheet

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Wireless Analog Telephone Adapter

Key Feature

Data Sheet

1

C

ombining the cutting edge of Internet telephony and ATA manufacturing

experience, PLANET now introduces the latest member of PLANET Wireless

ATA family: the VIP-161SW.

To bring the most satisfaction to customers, the VIP-161SW not only

provides the high quality of voice communications and wired Internet

sharing capabilities but also offers Access Point (AP) function for daily

wireless communication. With advanced router and VoIP DSP processor

technology, the VIP-161SW is able to make calls via SIP proxy voice

communications plus the IP sharing and the QoS mechanism.

The VIP-161SW is the ideal choice for Voice over IP communication and

integrates Internet sharing for the daily tasks. To give most flexibility to

users, the Wireless ATA provides direct analog interface for fax machine

and analog telephones. Users can not only make the daily VoIP

communication but also enjoy the convenience brought by FoIP

communications.

With the VIP-161SW, home users and companies are able to save the cost

of installation and extend their previous investments in telephones,

conferences and speakerphones. The VIP-161SW equipped with two

telephony interfaces, so users may register to different SIP proxy servers

and establish up to 2 concurrent VoIP calls for more flexibility in the voice

communications. The VIP-161SW can be the bridge between traditional

analog telephones and IP network with an extremely affordable

investment.

The VIP-161SW includes two Ethernet interface for Internet (PPPoE, DHCP

or Fixed IP) or office LAN connection. The dual Ethernet design brings the

greatest convenience when deploying VoIP network. With a built-in IEEE

802.11b/g wireless AP/CPE, the Wi-Fi ATA offers wireless connectivity via

54Mbps data transmissions.

VIP-161SW

Product Features

• IEEE 802.11b/g compliant

• Multi-mode: AP, AP-Client Mode

• Smart QoS mechanism to ensure the voice quality

• Auto-config feature for flexible, ease-of use system integration

• NAT Router, Static Routing, Virtual Server, DMZ

• Smart QoS mechanism to ensure the voice quality

• IP ToS (IP Precedence) / DiffServ

VoIP Features

• SIP 2.0 (RFC3261) compliant

• Up to 2 concurrent VoIP calls

• Voice codec support: G.711, G.729 AB, G.723, G.276

• T.38 FAX transmission over IP network (G.711 Fax pass-through)

• In-band and out-of-band DTMF Relay (RFC 2833)

• Three-way conference calls

• Call Waiting / Forward / Transfer / Hold / Resume / Screen

• Caller ID Detection/Generation: DTMF, Bellcore, ETSI, NTT

• Voice processing: VAD, CNG, Dynamic Jitter Buffer, G.168~2000

echo cancellation

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