Planet VIP-1140PT

4-SIP IP Phone with PoE

 

Intuitive, Aesthetic Design and Quality Communication

PLANET’s VIP-1140PT is a 4-line SIP VoIP phone set featuring color display, user-friendly, aesthetic user interface design and HD voices.

It is low profile when sitting on desktop but can perfectly provide great assistance to efficiently get communications done with ease of use, quality voice and the capability of up to 6-way conferencing.

For the support of wideband audio, in addition to the basic G.722 codec, the phone set also supports IETF’s Opus codec which has very low latency and significantly reduces the possibility of voice distortion or discontinuity.

Multi-language VoIP Telephony with Wide-Viewing-Angle Color Display 

The VIP-1140PT allows you to make digital phone calls utilizing existing broadband networks in homes and offices without installing new analog connections (e.g. copper wires). Its 2.4-inch wide-viewing-angle color display with a resolution of 320 x 240 pixels offers a clear depiction of caller’s information as well as supports up to 19 languages. Compliance with IEEE 802.3af PoE standards makes deployment convenient and flexible. In addition, the advanced speaker/microphone and the dedicated Digital Signal Processor assures superior audio quality. 

Compliant with SIP 2.0

SIP phones continue to gain popularity among businesses as the preferred protocol for enhancing communication experience across IP networks. The VIP-1140PT supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VIP-1140PT is able to broadly interoperate with equipment supplied by VoIP infrastructure providers, thus enabling them to offer their customers better voice over IP services.

 

Affordable for All Businesses

The VIP-1140PT 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.

High-quality G.722 HD and Opus Audio Codes 

The VIP-1140PT delivers with Harman Kardon speaker, wideband G.722 HD and Opus audio codec whose both hardware and software HD functions are the next generation of voice quality for telephony audio, making the quality of voice better than that (toll quality) of the standard digital telephony and come close to that of a room conversation. HD voice is transmitted in the audio frequency range of 50Hz to 7KHz or higher over telephone lines, resulting in higher quality voice and clearer communication. The VIP-1140PT keeps bringing the most premium sound for users.

 

Specifications :

Hardware
Lines (Direct Numbers)	4 SIP Lines
Physical Interfaces	LCD x 1: 2.4-inch (320 x 240) WVA color LCD Keypad: 36 keys, including 3 line keys with tri-color LED 4 soft-keys 5 navigation keys 8 function keys 12 standard phone digit keys 3 volume control keys — Up, Down, Mute (microphone) 1 hands-free key HD hands-free speaker (0 ~ 7KHz) x 1 HD hands-free microphone (0 ~ 7KHz) x 1 HD handset (RJ9) x 1
Connectors	RJ9 phone jacket x 2: Handset x 1 Headphone x 1 RJ45 10/100BASE-TX Ethernet jacket x 2: Network x 1 (802.3af PoE Class 1 enabled) PC x 1 (Bridged Network)
Power Requirements	IEEE 802.3af/at Power over Ethernet 5V 600mA
Weight	665g
Dimensions (W x D x H)	210 x 168 x 60 mm
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
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
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
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 6-way conferencing Hot line
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), Russian, Italian, German, French, Hebrew, Spanish, Catalan, Euskara, Galego, Turkish, Slovenian, Czech, Dutch, Korean, Ukrainian, Portuguese
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.729A/B. iLBC Wideband codec: Opus, 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
Environment
Operating Temperature	0 ~ 45 degrees C
Operating Humidity	10 ~ 65% (non-condensing)
Emission	CE, FCC, RoHS