Polycom IP 6000 PoE

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Polycom IP 6000 PoE

SKU: pol-2200-15600-001

Availability: In stock

The Polycom® SoundPoint® IP 6000 PoE is a business conference phone. The SoundPoint IP 6000 has industry leading sound quality to ensure your conference calls are crisp and clear. This Phone does not include a Power Supply, please add it as an accessory if you require it.

DESCRIPTION

Manufacturer: Polycom


The SoundStation IP 6000 is an advanced IP conference phone that delivers superior performance for small to midsize conference rooms. With advanced features, broad SIP interoperability and remarkable voice quality, the SoundStation IP 6000 offers a price/performance breakthrough for SIP-enabled IP environments. The SoundStation IP 6000 features Polycom® HD Voice™ technology, boosting productivity and reducing listener fatigue by turning ordinary conference calls into crystal-clear interactive conversations. It delivers high-fidelity audio from 220 Hz to 14 kHz, capturing both the deeper lows and higher frequencies of the human voice for conference calls that sound as natural as being there.



Polycom SoundStation IP 6000 Core Features:


  • Polycom HD Voice
  • Polycom’s patented Acoustic Clarity Technology
  • 12-foot microphone pickup
  • Industry-leading SIP software
  • Robust interoperability
  • High-resolution display
TECHNICAL INFO
SPECIFICATIONS
Display
  • IEEE 802.3af Power over Ethernet (built in)
  • Optional external universal AC power supply: 100-240V, 0.4A, 48V/19W
Feature Keys
  • Size (pixels): 248 x 68 (W x H)
  • White LED backlight with custom intensity control
Keypad
  • Standard 12-key keypad
  • Context-dependent soft keys: 3
  • On-hook/Off-hook, redial, mute, volume up/down
Audio Features
  • Loudspeaker
  • Frequency: 220-14,000 Hz
  • Volume: Adjustable to 86 dB at 1/2 meter peak volume
  • Individual volume settings with visual feedback for each audio path
  • Voice activity detection
  • Comfort noise fill
  • DTMF tone generation / DTMF event RTP payload
  • Low-delay audio packet transmission
  • Adaptive jitter buffers
  • Packet loss concealment
  • Acoustic echo cancellation
  • Background noise suppression
  • Supported Codecs
  • G.711 (A-law and Mu-law)
  • G.729a (Annex B)
  • G.722, G.722.1
  • G.722.1C
  • Siren 14
Call Handling Features
  • Shared call / bridged line appearance
  • Busy Lamp Field (BLF)
  • Distinctive incoming call treatment/call waiting
  • Call timer
  • Call transfer, hold, divert (forward), pickup
  • Called, calling, connected party information
  • Local three-way conferencing
  • One-touch speed dial, redial
  • Call waiting
  • Remote missed call notification
  • Automatic off-hook call placement
  • Do not disturb function
Network & Provisioning
  • Ethernet 10/100 Base-T
  • 2.5mm connection port
  • EX mic ports: Two RJ-9 ports
  • IP Address Configuration: DHCP and Static IP
  • Time synchronization with SNTP server
  • FTP / TFTP / HTTP / HTTPS server-based central provisioning for mass deployments. Provisioning server redundancy supported.
  • Web portal for individual unit configuration
  • QoS Support – IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP
  • Network Address Translation (NAT) support – static
  • RTCP support (RFC 1889)
  • Event logging
  • Local digit map
  • Hardware diagnostics
  • Status and statistics
  • User selectable ringtones
  • Convenient volume adjustment keys
  • Field upgradeable
Security
  • Transport Layer Security (TLS)
  • Encrypted configuration files
  • Digest authentication
  • Password login
  • Support for URL syntax with password for boot server
  • HTTPS secure provisioning
  • Support for signed software executables