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PLANET continues to bring innovation to the Voice over IP communications market with cutting edge products and Internet telephony manufacturing experience. For a cost-effective and high performance VoIP communications today, PLANET introduces the desktop SIP IP Phone, VIP-251T, to fulfill the VoIP deployment needs from ITSP, enterprises to home.
Key Features
- SIP 2.0 (RFC3261) compliant
- G.711A, G.711U and G.729 voice codec
- Supports STUN, Outbound Proxy
- Supports up to 2 Proxy server registrations
- 3-Way Conference / Caller ID
- Built-in phone book
- Call Hold / Waiting / Transfer / Forward
- VAD / CNG / AEC / PLC / AJB /AGC
- Incoming / Outgoing / Miss Call record
- QoS / NAT / DHCP Server
- Auto-Provisioning / TR-069
- Web / Keypad management
The VIP-251T features high-quality speakerphone technology and provide various voice services including speaker on / off, call hold and call transfer with user-friendly speed dial button design. it also has additional features such as built-in DHCP clients, password-protected machine management, LCD menu display, speed-dial keys, last number redial, incoming message indicator and user-intuitive web administration system.
The VIP-251T brings the benefits of VoIP technologies to your daily life. It is the SIP IP phone featuring self-contained, service-integrated, intelligent phone functions, and powerful voice processing. The VIP-251T can effortlessly deliver toll voice quality equivalent to the regular SIP protocol connections by utilizing cutting-edge Quality of Service, echo cancellation, comfort noise generation (CNG) and voice compensation technology. Meanwhile, the dual Ethernet interfaces on the IP Phone allow users to install it in an existing network without interfering with desktop PC network connections. The VIP-251T is an ideal solution for office / home use as well as for Internet Telephony Service Provider (ITSP).
Hardware
- WAN 1 x 10/100Mbps RJ-45 port
- LAN 1 x 10/100Mbps RJ-45 port
- LCD display 4 x 16 characters
- Speaker Full duplex hands free speaker phone
Protocols and Standard
- Standard SIP 2.0 (RFC3261), MD5 for SIP authentication (RFC2069/ RFC 2617), SIP outbound proxy, SIP NAT Traversal Support STUN (RFC3489)
- Protocols SIP v1 (RFC2543), v2 (RFC3261), TCP/IP, UDP/RTP/RTCP, HTTP, ICMP, ARP, RARP, DNS, DHCP, SNTP, PPPoE
- Voice codec G.711: 64k bit/s (PCM)
- G.729: 8k bit/s
- Voice Standard Voice activity detection (VAD)
- Comfort noise generation (CNG)
- Acoustic echo canceller (AEC)
- G.165: Line echo canceller (LEC)
- Jitter Buffer
- Supplementary services Caller ID
- 3-Way conference
- Immediate (unconditional) call forwarding
- Busy call forwarding
- No answer calls forwarding
- Call Hold / Waiting / Transfer
- Call history Record incoming call
- Outgoing call
- Missed (not accepted) call history
Network and Configuration
- Access Mode Static IP, PPPoE, DHCP
- Management Web, LCD menu keypad, Auto-Provision
- Dimension (W x D x H) 220 x 187 x 97 mm
- Operating Environment 0~50 Degree C, 0~90% humidity
- Power Requirement 5V DC, 1A
- EMC/EMI CE, FCC Class B
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