Polycom SoundStation IP 6000 SIP-based IP Conference Phone

Description

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Polycom SoundStation IP 6000 SIP-Based IP Conference Phone with Power Supply
Condition: Brand New in Original Packaging
SKU: 220015660001
Features
Built-in IEEE 802.3af Power over Ethernet
Polycom HD Voice technology
High-fidelity calls at up to 14 kHz
Patented Polycom clarity technology
Picks up voices up to 12 feet (3.5m)
Microphone designed for small to midsize conference rooms
Advanced call handling, security, and provisioning features
High-resolution LED backlight display
Multi-language support
Supports optional expansion microphones
Product Description
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.
Polycom HD Voice Technology
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.
Delivers Advanced Audio Performance
For all conference calls, the SoundStation IP 6000 delivers advanced audio performance that far exceeds previous generations of conference phones. From full-duplex technology that eliminates distracting drop-outs to the latest echo cancellation advancements, only Polycom can deliver a conference phone experience with no compromises. Plus, Automatic Gain Control intelligently adjusts the microphone sensitivity based on where participants are seated in the conference room, making the conversations clearer for all participants. It also features technology that resists interference from mobile phones and other wireless devices, delivering clear communications without distractions.
Robust Standards-Based SIP Interoperability
The SoundStation IP 6000 leverages Polycom’s strong history in both conference phone and VoIP technology to deliver the most robust standards-based SIP interoperability in the industry. It shares the same SIP phone software base with Polycom’s award-winning SoundPoint IP products—the most comprehensive, reliable and feature-rich SIP products in the industry, with proven interoperability with a broad array of IP PBX and hosted platforms.
Integrated PoE and AC Power Support
Robust provisioning, management and security features make Polycom’s family of IP conference phones the only choice for meeting rooms in SIP-based environments. Integrated Power over Ethernet (PoE) simplifies setup, with an AC power kit included for non-PoE environments. Plus, the SoundStation IP 6000 includes a high-resolution backlit display for vital call information and multi-language support.
What's in the Box
Polycom SoundStation IP 6000 SIP-Based IP Conference Phone with Power Supply
Telephone Console
25-Feet Ethernet Cable
Universal Power Supply
7-Feet Region-Specific Power Cord
Power Insertion Cable
Quick Start Guide
Quick User Guide
Specifications
Polycom SoundStation IP 6000 SIP-Based IP Conference Phone with Power Supply
Power
IEEE 802.3af Power over Ethernet (built in)
External universal AC power supply (included): 100-240V, 0.4A, 48V/19W
Display
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
Other Features
Local feature-rich GUI
Time and date display
User-configurable contact directory and call history (missed, placed, and received)
Customizable call progress tones
Wave file support for call progress tones
Unicode UTF-8 character support
Multilingual user interface encompassing: Simplified Chinese, Danish, Dutch, English (Canada / US / UK), French, German, Italian, Japanese, Korean, Norwegian, Polish, Portuguese, Russian, Slovenian, Spanish, Swedish
Network and 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 ringer tones
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
Safety
CE Mark
EN60950-1
IEC60950-1
UL60950-1
CAN/CSA C22.2 No.60950-1-03
AS/NZS60950-1
RoHS Compliant
EMC
FCC Part 15 (CFR 47) Class B
ICES-003 Class B
EN55022 Class B
CISPR22 Class B
AS/NZS CISPR22 Class B
VCCI Class B
EN22024
Telecom
AS/ACIF S004
Telepermit
KCC
GOST-R
TRA
Protocol Support
IETF SIP (RFC 3261 and companion RFCs
Environmental Conditions
Operating temperature: 32 - 104 degree F (0 - 40 degree C)
Relative humidity: 20%-85% (noncondensing)
Storage temperature: -22 - 131 degree F (-30 - 55 degree C)
Shipping
Payment
Returns
Feedback
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