- Overview
- Application
- Key Features
- Specification
Intuitive Design Brings Quality Communication
PLANET VIP-1120PT is a two-line SIP desktop phone with color display that brings lifelike richness and voice quality to phone calls. The VIP-1120PT is a reliable communication device as it features ease of use, versatility, high-quality design and superb audio performance. Besides, its full duplex speakerphone system with HD voice can definitely make communication between two or more business parties crystal clear with no interference noise in the background as it supports the G.722 wideband codec. | | | | | | | | | | | | | | IP Phone with Color Display
As the VIP-1120PT is an IP phone, it allows you to make calls from any location easily as long as it is online. Its 2.4-inch TFT color display with a resolution of 320 x 240 pixels offers a clear depiction of caller’s information. It complies with IEEE 802.3af PoE interface for flexible deployment and supports superior audio quality delivered by the advanced speaker and microphone system, and the digital signal processor (DSP).
| | | | | | | Compliant with SIP 2.0
SIP phones continue to gain popularity among businesses as the preferred protocol for enhancing communication across IP networks. The VIP-1120PT supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VIP-1120PT is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better voice over IP services.
| | | | | | | | | | | | | | Affordable for All Businesses
The VIP-1120PT is definitely affordable for all business establishments who want flexible deployment options and expansion. It can effortlessly deliver secured toll voice quality by utilizing cutting-edge 802.1p QoS (Quality of Service) and IP ToS technology.
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Enhanced, Full-Featured SIP IP Phone
The VIP-1120PT is optimized for executive use for administrative assistants and those working with bandwidth-intensive application on collocated PCs. Four programmable extension keys could be configured as IP PBX features like BLF, MWI, DND, Call Forward, Call Park and many others.
| | | | | | | | | | | | | | Enterprise IP Telephony Deployment of VIP-1120PT
Help your teams move faster with exceptional audio quality and built-in flexibility. The VIP-1120PT 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.
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Highlights- 2.4-inch color LCD monitor
- 2-line key with color LED
- 2 SIP identities
- Wideband G.722 HD audio
- Power over Ethernet (PoE) compliant
- QoS, TR069, auto-provisioning and multi-language
Advantageous Applications- Supports SIP 2.0 (RFC 3261)
- Inband, SIP info, RFC 2833 DTMF relay
- Soft keys and function keys programmable
- Echo cancellation: Supports G.168, and 96ms max. filter length in hands-free mode
- Supports voice gain setting, voice activity detection (VAD) and comfort noise generation (CNG)
- Full duplex hands-free speakerphone
- Hands-free headset ringing choice
- Voice codec setting for each SIP line
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
- Redial and click to dial
- Secondary dialing automatically
- Incoming calls, outgoing calls and missing calls (Each supports 300 records)
- SMS and speed dial
- Phonebook up to 500 records
Call Control Features- Flexible dial map, hotline and empty calling no. for rejected service
- Black list for rejected authenticated calls
- White list and call limit
- Do not disturb (DND)
- Caller ID display
- Dial without registration
Network Features- PPPoE/DHCP client
- 802.1 VLAN (voice VLAN/data VLAN)
- VPN (L2TP) and openVPN
- Quality of Service
Maintenance and Management- Integrated web server provides web-based administration and configuration
- Telephone keypad configuration via display menu/navigation
- Automated provisioning and upgrade via HTTPS, HTTP, TFTP
- User authentication for configuration pages
- SNTP time synchronization
- TR069
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| Hardware |
| Lines (Direct Numbers) |
2 SIP Lines |
| Physical Interfaces |
LCD x 1: 2.4-inch (320 x 240) color LCD
Keypad: 32 keys, including
- 4 soft-keys
- 12 standard phone digit keys
- 5 navigation keys
- 2-line key with tri-color LED
- 5 function keys
- 3 volume control keys -- Up, Down, Mute (microphone)
- 1 hands-free feature
HD hands-free speaker (0 ~ 7KHz) x 1
HD hands-free microphone (0 ~ 7KHz) x 1
HD handset (RJ9) x 1
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| Protocols and Standard |
| Protocols |
SIP2.0 over UDP/TCP/TLS
RTP/RTCP/SRTP
STUN
DHCP
PPPoE
802.1x
L2TP (basic unencryption)
OpenVPN
SNTP
FTP/TFTP
HTTP/HTTPS
TR069
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| Request for Comments (RFCs) |
354, 1321, 1350, 1769, 1889, 1890, 2131, 2132, 2616, 2617, 2661, 2833, 2976, 3261, 3262, 3263, 3264, 3265, 3268, 3311, 3489, 3711, 4346, 4566, 5630, 5865 |
| Networking |
| Networking |
Physical: 10/100Mbps Ethernet, dual bridged port for PC bypass
IP Configuration: Static, DHCP, PPPoE
Network Access Control: 802.1x
VPN: L2TP (basic unencryption), OpenVPN
VLAN
QoS
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| Deployment & Maintenance |
Auto-provisioning via FTP/TFTP/HTTP/HTTPS/DHCP/OPT66/SIP PNP/TR069
Web management portal
Web-based packet dump
Configuration backup/restore
Auto/Manual online software upgrade
Syslog
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| Features |
| Call Features |
Call out, answer, reject
Mute/unmute (microphone)
Call hold, resume
Call waiting
Intercom
Caller ID display
Call forwarding (always/busy/no answer)
Call transfer (attended/unattended)
Call parking, pick-up (depending on server)
Redial
Do-not-disturb (per line/per phone)
Auto-answering (per line)
Voice message (on server)
Local 3-way conferencing
Hot line
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| Phone Features |
Phone accessibility control by user PIN
Intelligent phonebook (up to 500 entries in total)
Remote phonebook (XML/LDAP)
Intelligent contact number matching/filtering
Call log (300 entries in total, in/out/missed)
Voice message waiting indication (VMWI)
Network time synchronization
Multi-language support in screen and web UI: English, Chinese (Traditional/Simplified), Japanese, Russian, Italian, Turkish, German, Dutch, Spanish, Hebrew, Polish, French, etc.
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| Audio Features |
HD voice microphone/speaker (handset/hands-free, 0~7KHz frequency response)
Wideband ADC/DAC 16KHz sampling
Narrowband codec: G.711a/u, G.723.1, G.726-32K, G.729AB
Wideband codec: G.722
Full-duplex acoustic echo canceller (AEC) – 96ms tail-length in hands-free mode
Voice activity detection (VAD), comfort noise generation (CNG), background noise estimation (BNE), noise reduction (NR)
Packet loss concealment (PLC)
Dynamic adaptive jitter buffer up to 300ms
DTMF: In-band, out-of-band – DTMF-relay (RFC 2833), SIP info
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| Environment |
| Power Requirements |
IEEE 802.3af Power over Ethernet
5V 600mA
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| Operating Temperature |
0 ~ 40 degrees C |
| Operating Humidity |
10 ~ 65% (non-condensing) |
| Weight |
TBD |
| Dimensions (W x D x H) |
TBD |
| Emission |
CE, FCC, RoHS |
| Connectors |
RJ9 phone jacket x 2: Handset x 1, headphone x 1
RJ45 Ethernet jacket x 2: Network x 1 (802.3af PoE Class 1 enabled), PC x 1 (Bridged Network)
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