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Polycom SoundStation IP6000 SIP Expandable Conference Phone - PoE (2200-15600-001) Unused

SKU: 2200-15600-001-UNU

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MSRP: $659.00

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2200-15600-001


Polycom SoundStation IP6000 SIP Expandable Conference Phone - PoE


Unused


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Condition: Unused

Polycom 6000 Conference Phone
(Polycom Soundstation IP 6000)
Speak up to 20 feet away from the phone
Protocols: SIP
Manufacturer Part Number: 2200-15600-001
802.3af PoE Compatible

Does Not Include Power Supply
If you will be using 802.3af Power over Ethernet (PoE) to power the conference phone a power supply may not be necessary.


Polycom 6000 Product Description

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.

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.

The SoundStation IP 6000 leverages Polycoms 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 Polycoms 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.

Robust provisioning, management and security features make Polycoms 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 available for non-PoE environments. Plus, the SoundStation IP 6000 includes a high-resolution backlit display for vital call information and multi-language support.

Polycom IP 6000 Key Features & Benefits

Polycom HD Voice unparalleled clarity to make your conference calls more efficient and productive

Polycoms patented Acoustic Clarity Technology delivering the best conference phone experience with no compromises

12-foot microphone pickup combined with Automatic Gain Control for performance far beyond older SoundStation IP conference phones. Add up to two optional expansion microphones for even greater coverage.

Industry-leading SIP software leveraging the most advanced SIP endpoint software in the industry, with advanced call handing, security, and provisioning features

Robust interoperability compatible with a broad array of SIP call platforms to maximize voice quality and feature availability while simplifying management and administration

High-resolution display enables robust call information and multi-language support

Polycom IP 6000 Technical Specifications

Power
IEEE 802.3af Power over Ethernet (built in)
Optional external universal AC power supply: 100-240V, 0.4A, 48V/19W (sold separately)

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
  o Frequency: 220-14,000 Hz
  o Volume: Adjustable to 85 dB at 1/2 meter peak volume
Full-duplex: Type 1 compliant with IEEE 1329 full duplex standards
Individual volume settins 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 Chinese, Danish, Dutch, English (Canada / US / UK), French, German, Italian, Japanese, Korean, Norwegian, Portuguese, Russian, 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. rovisioning 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
UL1950
CE Mark
CSA C22.2, No 60950
EN60950
IEC60950
AS/NZSS3260

EMC
FCC (47 CFR Part 15) Class B
ICES-003 Class
EN55022 Class B
CISPR22 Class B
AS/NZS 3548 Class B
VCCI Class B
EN61000-3-2; EN61000-3-3
EN55024
ROHS compliant

Protocol Support: IETF SIP (RFC 3261 and companion RFCs)

IEEE 802.3af Power over Ethernet version ships with:
Telephone Console
25 foot Ethernet cable
Quick Start Guide
Quick User Guide

AC Power version ships with:
Telephone Console
25 foot Ethernet cable
Universal Power Supply
7 foot region-specific power cord
Power Insertion Cable
Quick Start Guide
Quick User Guide

Environmental Conditions
Operating temperature: 32 - 104 degrees F (0 - 40 degrees C)
Relative humidity: 20%-85% (noncondensing)
Storage temperature: -22 - 131 degrees F (-30 - 55 degrees C)

Warranty: 1 year

Country of Origin: China

Phone Dimensions: 14.5 x 12.25 x 2.5 in (36.8 x 31.1 x 6.4 cm) (L x W x H)
Phone Console Weight: 1.75 lb (0.8 kg)

Box Dimensions: 13.0 x 15.5 x 6.0 in (33 x 39.5 x 15 cm) (L x W x H)
Box Weight: 5.1 lb (2.32 kg)

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