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WIP330
The Linksys
WIP330 features peer-to-peer dialing, speed dial, 3-way conferencing,
call waiting, call transfer, and call forwarding, mute hold and
selectable ringtones. The large, full color high resolution LCD
display features an intuitive user interfaceenabling users to easily
and quickly configure te handset when traveling within range of
other Wireless-G networks so you can make VoIP calls wherever you
go.
You can also
surf the Internet with the built-in web browser and can even receive
live video from anywhere in the world and view it right on your
WIP330 Wireless-G IP Phone when you access any web camera, like
the WVC54GC Wireless-G Internet Video camera, also from Linksys.
Linksys WIP330
Features
- Supports
SIP V2.0 Standards
- Compliance
with IEEE 802.11b/g Wireless Standards
- Powered by
Linux
- High resolution
color LCD screen
- Supports
QoS (Quality of Service) to ensure best quality voice
- Enhanced
power saving design for extended standby and talk time
- 50 hours
standby time, 3 hours talk time on average
- 3-Way conferencing,
Call Hold and Resume, and CallerID
- Fast Hotspot
Authenticaton
- Supports
auto-provisioning using HTTP or HTTPS for configuration and upgrades
Comes with
- Handset
- USB Power
Adapter
- Li-Ion battery
- Quick Install
Guide
- User Guide
on CD-Rom
Specifications:
- Model:
WIP330-NA
- Standards:
802.11b, 802.11g
- Channels:
11 Channels (US, Canada), 13 Channels (Europe)
- Access
Control: CSMA/CA with ACK
- Band:
2.412~2.484 GHz
- Transmit
Power: 14 dB for 802.11b/g @ Normal Temp Range
- Radio
Range: Outdoor up to 300m via Embedded Antenna
- External
Interface: One mini-USB socket (for charging only), one stereo
earphone jack
- Display:
QVGA TIF 2.2 inch LCD (340x320 Pixels) with 65K colors
- Memory:
32MB Flash, 64MB SDRAM
- WIFI Account
Support: Multiple Access Point Registration Support
- Network
Protocols: TCP/UDP/IP, IPV4, DNS, SDP, ARP, ICMP, DHCP Client,
Static IP
- Security:
WEP (64/128), WPA-PSK Encryption
- QoS: ToS
- Voice
Protocols: SIP V2 Session Initiation Protocol (RFC3261)m SDP
(RFC2327), SIP Session Keep Alive
- Voice
Codecs: G.711 (A-Law an U-Law), G.729A
- DTMF Transmission:
In-band, Out-band (RFC2833)
- Voice
Quality
- G.168
echo cancellaton
- Jitter
Buffer Control - (default 180ms, max 900ms)
- Comfortable
Noise Generation
- Packet
Loss Concealment
- Speaker
and Microphone Volume Control
- VAD (Voice
Activity Detection)
- Telephony
and Supplementary Features
- 3-Way
Conferencing
- Peer-to-Peer
Dialing
- Call
Hold and resume
- Caller
ID Presentation
- Caller
ID Presentation Restriction
- Dial
by Phone Number
- Call
Forward
- DTMF
Tone Detection
- Consultation
Hold and Transfer
- Call
Waiting and Retreive
- Mute
- Speed
Dial
- Last
Number Redial
- Volume
Control
- Ringtones:
True Tones
- Phone
Book (250 records)
- Call
History (20 records)
- Language
(English / Spanish)
- Vibrator
(Silent Mode)
- Password
Security
- 10 Profiles
- Date
& Time (NTP Time Synchronization)
- Internet
Web Browser
- Management
Features
- AES or
SSL encryption
- Firmware
Upgrades using HTTP or HTTPS
- Configuration
change using HTTP or HTTPS
- Embedded
web configuration interface (with password protection)
- Power-Up
Diagnostics)
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