Cost-effective, High-definition VoIP Phone
PLANET VIP-1000PT and VIP-1000T are high-definition but cost-effective IP Phones where the earlier model comes with the PoE technology and the latter is without PoE. Whatever, both models, through IP PBX, feature VoIP and traditional telephone communications, and converged data and voice networks which can be built from one location to another without considering distance, thus making communications convenient over a long distance.
 
In addition, the VIP-1000PT has a 1-line business IP feature. VoIP communications can be extended when using PPTP VPN or L2TP VPN. The VIP-1000PT also allows call to be transferred to anyone at any location within the voice system, which enables the enterprise to communicate more effectively and is helpful to streamline business processes.

Standard Compliance
Compliant with the Session Initiation Protocol 2.0 (RFC 3261), the VIP-1000PT is able to function with other PLANET and any third-party VoIP products.

Enhanced, Full-featured Business IP Phones
The VIP-1000PT is business IP phone that address the communication needs of the enterprises. They provide 1 voice line and 10/100Mbps Ethernet network. Furthermore, the VIP-1000PT delivers 20 multi-functional keys with speed dial and shortcut key. The VIP-1000PT supports all kinds of SIP-based phone features including call waiting, auto answer, music on hold, caller ID and call waiting ID, 3-way conferencing, call hold, call forwarding, black list, hotline, DTMF relay, in-band, out-of-band (RFC 2833) and SIP info method, among others. Besides office use, the VIP-1000PT is also the ideal solution for VoIP service offered by Internet Telephony Service Provider (ITSP).

Secure, High-Quality VoIP Communication
The VIP-1000PT supports SIP v2 for easy integration with general voice over IP system. It can also effortlessly deliver secured toll voice quality by utilizing cutting-edge 802.1p QoS (Quality of Service) and IP TOS technology. It also supports HD (High Definition) voice as G.722 to provide clear communications.

Enterprise IP Telephony Deployment of VIP-1000 Series
The VIP-1000 Series is much easier to install and configure than the traditional phone system. Its low cost and high-definition voice quality give you value for money. Based on standard SIP 2.0, it is compatible with all the standard SIP-based servers.
The VIP-1000 Series (The VIP-1000PT PoE model or the VIP-1000T non-PoE model) can be set up in any place to conveniently communicate with friends or business associates via IP PBX.

Highlights

• Supports SIP 2.0 (RFC3261)
• IEEE 802.3af Power over Ethernet compliant
• Supports HD voice (G.722)
• Voice Activity Detection
•  Auto Provisioning: TFTP, HTTP and HTTPS
• IP conflict detection

Advantageous Applications

• SIP supports SIP domain, DNS name of server, peer to peer/IP call
• In-band, out-of-band, SIP info, RFC 2833 DTMF relay
• Adaptive jitter buffer management
• Echo cancellation
• Full duplex hands-free speaker phone
• Hands-free headset ringing choice
• Voice codec setting for SIP line
• Customized ring tone

SIP Applications

• Call forward and transfer (blind/attended)
• Call holding and waiting
• 3-way conferencing
• Paging and intercom
• Call park, call pickup and join call
• Call history, and blacklist (Each supports 100 records)
• Supports phonebook with 500 records
• Supports shortcut keys and speed dial
• Supports CSV phonebook and browser

Call Control Features

• DTMF Relay: In-band, out-of-band (RFC2833) and SIP info
• Call log: redial list, answered calls and missed calls
• White list and limit call
• Do not disturb (DND)
• Caller ID, CLIR (rejects an anonymous call) and CLIP (make a call with anonymous)
• Dial without registration

Network Features

• PPPoE and DHCP client on WAN
• 802.1P and Q VLAN
• VPN (L2TP, PPTP)
• Main DNS and secondary DNS server
• DNS relay and SNTP client
• QoS with Layer 2 and Layer 3 (SIP/RTP/Data)

Maintenance and Management

• Integrated web server provides web-based administration and configuration
• Automated provisioning and upgrade via HTTPS, HTTP, TFTP
• User Authentication for configuration pages
• Local and remote Syslog (RFC 3164)
• SNTP time synchronization and TR-069

Hardware

• Lines (Direct Numbers)  1-line business-class IP phone
• Feature Keys  12 dialing buttons (0~9, *, #)
  4 x fixed function buttons
  20 multi-functional key
• Physical Interfaces  One 10/100BASE-TX RJ45 Ethernet port (IEEE 802.3)
  Handset: RJ9 connector
  Built-in speakerphone and microphone

Protocols and Standard

• Data Networking  MAC address (IEEE 802.3)
  IPv4 (RFC 791)
  Address Resolution Protocol (ARP)
  DNS: A record (RFC 1706), SRV record (RFC 2782)
  Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
  Internet Control Message Protocol (ICMP) (RFC 792)
  TCP (RFC 793)
  User Datagram Protocol UDP (RFC 768)
  Real-time Protocol RTP (RFC 1889, 1890)
  Real-time Control Protocol (RTCP) (RFC 1889)
  Differentiated Services (DiffServ) (RFC 2475)
  Type of Service (ToS) (RFC 791, 1349)
  VLAN tagging 802.1p/Q: Layer 2 Quality of Service (QoS)
  Simple Network Time Protocol (SNTP) (RFC 2030)
  Backward compatible with RFC 2543
  Session Timer (RFC 4028)
  SDP (RFC 2327)
  NAPTR for SIP URI Lookup (RFC 2915)
• Voice Gateway  SIP version 2 (RFC 3261, 3262, 3263, 3264)
  SIP support in NAT networks [including STUN (RFC 3489)]
  Message Waiting Indicator (RFC 3842)
  Voice algorithms:
  - G.711 (A-law and µ-law)
  - G.729A/AB with PAMS above 4.0
  - G.722
  - G.723
  Dual-tone multi-frequency (DTMF), in-band and out-of-band (RFC 2833) (SIP info)
  Voice activity detection (VAD)
  Adaptive jitter buffer management
  Comfort noise generation
  Echo cancellation
• Provisioning, Administration, and Maintenance:  Integrated web server provides web-based administration and configuration
  Automated provisioning and upgrade via HTTPS, HTTP, TFTP
  User authentication for configuration pages
  Local and remote Syslog (RFC 3164)
  SNTP time synchronization
  Capture wireshark trace via web
  Multi-user level
  SNMP v2
  TR069
  

Features

• Telephony Features  One voice line
  Call Waiting
  Auto Answer
  Music on Hold
  Caller ID
  3-way call conferencing
  Call Hold and Call Forwarding
  Call Transfer: blind transfer and attended transfer
  Call Log: redial list, answered calls and missed calls
  Volume Adjustment: handset, speaker and ringer
  Volume Gain: handset input and speakerphone input
  Delayed Hotline
  Redial, Speed Dial
  Pick Up, Call Park, Dial Plan
  Black List
  Do not disturb (DND)
  Full-duplex speakerphone
  Customized Ring Tone
  Call History (100 records )
  - Most Recently Missed Calls
  - Most Recently Received Calls
  - Most Recently Dialed Numbers
  Phone book ( 500 records)
  Blacklist (100 records )

Environment

• Power Requirements  5V DC, 1A
  IEEE 802.3af PoE class 3
  Max. 2W
• Operating Temperature  0 ~ 50 degrees C
• Operating Humidity  10 ~ 90% (non-condensing)
• Weight  488g
• Dimensions (W x D x H)  185 x 146 x 67 mm
• Emission  CE, FCC
• Connectors  One 10/100Mbps Ethernet, RJ45
  RJ9 handset connector
  DC power jack
  DND switch