A POLYCOM DATA SHEET

Polycom® SoundStation® IP 7000

IP Conference Phone

Astounding voice quality and clarity from the world’s most advanced IP conference phone

The Polycom® SoundStation® IP 7000 is a breakthrough conference phone that

delivers outstanding performance and robust capabilities for SIP-based VoIP platforms.
Ideal for boardrooms, conference rooms, auditoriums and executive offices, the
SoundStation IP 7000 is the most advanced and expandable conference phone
ever developed. Connect two units together or add optional expansion microphones
for increased room coverage and voice pickup. Connect the SoundStation IP 7000
to a Polycom HDX® video conferencing system for a complete, integrated voice and
video conferencing solution.
Benefits
The SoundStation IP 7000 features Polycom® HD Voice™ that delivers high-fidelity
audio from 160Hz to 22kHz, boosting productivity and reducing listener fatigue by  olycom HD Voice – unparalleled
P
turning ordinary conference calls into crystal-clear interactive conversations. From clarity to make your conference calls
full-duplex audio that eliminates distracting drop-outs to the latest echo cancellation more efficient and productive
advancements, only Polycom can deliver a conference phone experience with no Polycom’s
 patented Acoustic
compromises. Clarity Technology – delivers the best
conference phone experience with no
The SoundStation IP 7000 shares the same UC software with Polycom’s award-winning
compromises
SoundPoint® IP products—the most comprehensive, reliable and feature-rich SIP
products in the industry, with proven interoperability across a broad array of IP PBX and F
 lexible configuration options –
hosted platforms—making it the clear choice for SIP environments. A large multi-line multi-unit connectivity, expansion
high-resolution LCD display with multi-language support offers call information and microphones and integration with
context sensitive call functions. Polycom HDX to meet the needs of
many different types of rooms
About Polycom
S
 trong, robust SIP software –
Polycom is the global leader in standards-based unified communications (UC) solutions leverages the most advanced SIP
for telepresence, video, and voice powered by the Polycom® RealPresence® Platform. endpoint software in the industry, with
The RealPresence Platform interoperates with the broadest range of business, mobile, advanced call handing, security, and
and social applications and devices. More than 400,000 organizations trust Polycom provisioning features
solutions to collaborate and meet face-to-face from any location for more productive
R
 obust interoperability – compatible
and effective engagement with colleagues, partners, customers, specialists, and
with a broad array of SIP call platforms
prospects. Polycom, together with its broad partner ecosystem, provides customers
to maximize voice quality and
with the best TCO, scalability, and security for video collaboration, whether on-premises,
feature availability while simplifying
hosted, or cloud-delivered. Visit www.polycom.com or connect with Polycom on Twitter,
management and administration
Facebook, and LinkedIn.
L
 arge high-resolution display with
XHTML microbrowser –
enables new applications that make
conference calling easier and more
functional
Polycom® SoundStation® IP 7000 Conference Phone
Features and Specifications
Additional SoundStation IP 7000 features/benefits
• Equipped with built-in Power over Ethernet (PoE). An optional A/C power kit also available.
• 20-foot microphone pickup, and even more with optional expansion microphones or multi-unit connectivity, reaching all corners of the room.
• Automatic Gain Control intelligently adjusts the microphone sensitivity based on where participants are seated in the conference room.
• Features technology that resists interference from mobile phones and other wireless devices, delivering clear communications without distractions.
• Built-in 2.5mm applications port allows you to connect the conference phone to a mobile phone for productive calls even where no network connection is available, or to
a computer for calls using PC-based soft phone clients.

Power (missed, placed, and received) • VCCI Class A

• IEEE 802.3af Power over Ethernet (Class 3) • Corporate Directory (LDAP) support • EN55024
• User selectable ringer tones • RoHS compliant
• Optional external universal AC power supply:100-
• Wave file support for call progress tones Telecom
240V, 1.5A, 48V/50W • Unicode UTF-8 character support • AS/ACIF S004
Display • Multilingual user interface encompassing Simplified • Telepermit
• Size (pixels): 255 x 128 (W x H) Chinese, Traditional Chinese, Danish, Dutch, • KC
• White LED backlight with custom intensity control English (Canada /US/UK), French, German, Italian, • TRA
Japanese, Korean, Norwegian, Polish, Portuguese, • ANATEL
Keypad Russian, Slovenian, Spanish, Swedish • GOST-R
• Standard 12-key keypad • Called, calling, connected party information
• Context-dependent soft keys: 4 Protocol Support
• On-hook/off-hook, redial, mute, volume up/down, Interfaces • IETF SIP (RFC 3261 and companion RFCs)
menu, navigation keys • Ethernet 10/100 Base-T
• 2.5mm connection port (audio in/out) for PCs and Power over Ethernet version ships with
Audio Features mobile phones • Conference Phone Console
• 3 cardioid microphones 200 to 22,000 Hz • 25 foot (7.6m) Ethernet cable
• 2 Walta ports for wired expansion microphones,
• Loudspeaker frequency response: 160-22,000 Hz • Quick Start Guide
daisy-chaining and Polycom HDX video integration
• 20ft (6.1m) microphone pickup AC Power version ships with
• Volume: Adjustable to 88 dB at 0.5 meter peak Network and Provisioning
• Conference Phone Console
volume • IP Address Configuration: DHCP and Static IP
• 25 foot (7.6m) Ethernet cable
• Full-duplex: Type 1 compliant with IEEE 1329 • Time synchronization with SNTP server
• Universal Power Supply 48V, 1.5A
• Individual volume settings with visual feedback for • FTP/TFTP/FTPS/HTTP/HTTPS server-based
• 7 foot (2.1m) region-specific power cord
each audio path central provisioning for mass deployments.
• Power Insertion Cable
• Automatic Gain Control Provisioning server redundancy supported.
• Web portal for individual unit configuration and • Quick Start Guide
• Voice activity detection
• Comfort noise fill online software upgrade HDX Ready version ships with
• DTMF tone generation / DTMF event RTP payload • QoS Support: IEEE 802.1p/Q tagging (VLAN), • Conference Phone Console
• Low-delay audio packet transmission Layer 3 TOS and DSCP • 25 foot (7.6m) Ethernet cable
• Adaptive jitter buffers • Network Address Translation (NAT) support - static • 10 foot (3m) CLink2 cable for connection
• Packet loss concealment • RTCP support (RFC 1889) to HDX
• Acoustic echo cancellation • Configuration import/export • Quick Start Guide
• Background noise suppression • Local digit map (dialing plan)
• Supported Codecs • Hardware diagnostics Accessories
--G.711 (A-law and Mu-law) • Status and statistics • 2 wired or wireless expansion microphones
--G.729a (Annex B) • Reset to factory settings • Lapel pin microphone (limited availability)
--G.722, G722.1 • Multi Interface Module for daisy-chaining, lapel pin
Security microphone and external speaker connectivity
--G.722.1C
• Transport Layer Security (TLS)
--Siren 14 Environmental Conditions
• Encrypted configuration files
--Siren 22 • Operating temperature: 32 - 104°F (0 - 40°C)
• Digest authentication
--iLBC 13.33 and 15.2kbps • Password login • Relative humidity: 20%-85% (noncondensing)
SIP Call Handling Features • Support for URL syntax with password for boot • Storage temperature: -22 - 131°F (-30 - 55°C
• Call transfer, hold, divert (forward) and pickup server
• HTTPS secure provisioning Warranty
• Distinctive incoming call treatment/call waiting • 1 year
• Advanced Local three-way conferencing • Support for signed software executables
(conference, join, split, hold, resume) • IEEE 802.1x Network Access Control Country of Origin
• One-touch speed dial, redial Safety • Thailand
• Remote missed call notification • UL60950-1 Phone Dimensions
• Automatic off-hook call placement • IEC60950-1 • 15.5 x 14.6 x 2.9 in (39.4 x 37.2 x 7.3 cm) (L x W
• SIP URI dialing • EN60950-1
x H)
• Do not disturb function • CE Mark
• Shared call/bridged line appearance • CAN/CSA C22.2, No. 60950-1-03 Phone Console Weight
• Busy Lamp Field (BLF) • AS/NZS60950-1 • 2.4 lb (1.08 kg)
• Multicast Group Paging and Push-to-Talk EMC
Box Dimensions
Other Features • FCC (47 CFR Part 15) Class A • 19.1 x 17.0 x 5.1 (48.4 x 43.3 x 13 cm) (L x W x H)
• Integration with Polycom HDX video systems • ICES-003 Class A
• SIP Server Redundancy • EN55022 Class A Box Weight
• Time and date display/call timer • CISPR22 Class A • 5.4 lb (2.43 kg)
• User-configurable contact directory and call history • AS/NZS CISPR22 Class A

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4750 Willow Road, Pleasanton, CA 94588
1.800.POLYCOM or +1.925.924.6000
www.polycom.com

© 2012 Polycom, Inc. All rights reserved. POLYCOM®, the Polycom “Triangles” logo and the names and marks associated with Polycom’s products are trademarks and/or service marks of Polycom, Inc. and are registered
and/or common law marks in the United States and various other countries. All other trademarks are property of their respective owners. No portion hereof may be reproduced or transmitted in any form or by any means,
for any purpose other than the recipient’s personal use, without the express written permission of Polycom.
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