DATA SHEET
Polycom® SoundStation® IP 6000
SIP-based IP conference phone
Next-generation IP conference phone designed for small to
mid sized rooms
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 Sound Station 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.
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 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.
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 available
for non-PoE environments. Plus, the SoundStation IP 6000 includes a high-resolution
backlit display for vital call information and multi-language support.
Benefits
• Polycom HD Voice –
unparalleled clarity to make your
conference calls more efficient
and productive
• Polycom’s patented Acoustic
Clarity Technology –
Deliver 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
DATA SHEET Polycom SoundStation IP 6000 Conference Phone Specifications
Product Specifications
Power
• IEEE 802.3af Power over
Ethernet(builtin)
• Optional external universal AC power
supply: 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 serverbased 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
IEEE 802.3af Power over Ethernet
version ships with
• Telephone Console
• 25 foot Ethernet 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 de