DATA SHEET

Polycom

®

SoundStructure

®

C-Series Models

Truly immersive audio experience for voice and video conferencing

Polycom ® SoundStructure ® installed audio solutions deliver high definition clarity to voice and video conferences for improved productivity and faster decision making.

Advanced design and custom configuration capabilities allow you to quickly and efficiently deliver world-class audio quality to a variety of room environments. Seamless integration with Polycom video and voice conferencing solutions creates life-like experiences, with Polycom immersive and room telepresence solutions, Polycom ®

HDX ® microphones, and Polycom Ceiling Microphone Arrays delivering unmatched video and voice collaboration quality. Controlling SoundStructure is easier than ever with Polycom Touch Control, an intuitive touchscreen controller that allows you to dial calls and precisely adjust audio levels in the room.

SoundStructure C-Series installed audio solutions set a new standard in performance for immersive conferencing. Full-stereo acoustic echo cancellation, with 22kHz bandwidth and powerful feedback elimination, allows meeting participants to focus on the conversation without distraction. Comprehensive input, output and submix processing provides configuration flexibility and integrates all core audio processing into a single solution, eliminating the need and costs of additional equipment. Optional modular telephony interface cards expand functionality and provide investment protection.

The innovative Polycom OBAM™ matrix architecture in SoundStructure enables up to eight SoundStructure units to appear as one system for unparalleled scale and flexibility. Advanced signal grouping, channel labeling and submixing let you leverage existing designs and configurations for future installations, saving time and money, while exclusive SoundStructure Studio software simplifies configuration and enables advanced system designs for existing and evolving needs.

About Polycom

Polycom is the global leader in standards-based unified communications (UC) solutions for telepresence, video, and voice powered by the Polycom ® RealPresence ® Platform.

The RealPresence Platform interoperates with the broadest range of business, mobile, and social applications and devices. More than 400,000 organizations trust Polycom solutions to collaborate and meet face-to-face from any location for more productive and effective engagement with colleagues, partners, customers, specialists, and prospects. Polycom, together with its broad partner ecosystem, provides customers with the best TCO, scalability, and security for video collaboration, whether on-premises, hosted, or cloud-delivered. Visit www.polycom.com or connect with Polycom on

Twitter, Facebook, and LinkedIn.

Benefits

• Breakthrough audio conferencing

performance – Life-like voice conferencing with no compromises.

The most powerful and flexible installed audio solution available.

• Easy installation and configuration, even for very large or complex

systems – Innovations such as

SoundStructure Studio configuration software and OBAM ™ make it easier than ever to deliver crisp, clear, life-like sound.

• Seamless integration with Polycom

HDX ® solutions – Delivers the true

Polycom UltimateHD™ experience for natural, life-like voice and video conferencing that raises productivity to a whole new level.

Polycom SoundStructure C-Series Models Specifications

Features and Benefits

• Unrivaled Stereo Acoustic Echo Cancellation (AEC) –

The only product with both monaural and stereo echo cancellation from 20Hz to 22 kHz – AEC with no compromises for more immersive meetings

• Breakthrough feedback elimination – Enables more flexible microphone, talker, loudspeaker placements and speech reinforcement delivering audio where you want it

• Polycom OBAM™ matrix architecture – Connect multiple units together to create larger systems without the limitations of traditional bussing

• All conferencing features available on all inputs –

Eliminate the worries of running out of microphone inputs, since every feature is available on all inputs

• Gain sharing automatic microphone mixer – Improved automixer experience ensures smoother transitions and robust performance in a variety of operating environments

• Provide expanded functionality and investment

protection – Use modular single or dual-line PSTN, or VoIP telephony cards

• Fully digital Polycom HDX integration – Add powerful, configurable audio to your HD video solution

• Polycom Touch Control – Provides a simple-to-use interface for volume control, muting, and dialing of SoundStructure.

• Use HDX digital mic arrays – Compatible with both HDX digital mic arrays and traditional analog microphones (each

HDX digital microphone requires the processing of three analog inputs)

•  Push to talk microphones and other applications –

Gain flexible support with logic inputs (contact closures) and outputs

• Powerful events programming – Customize system behavior with the Polycom Touch Control, HDX telepresence solutions, HDX IR remotes, and more

• Password protection – Prevents unauthorized users from accessing, controlling or making changes to the system

SoundStructure Studio

SoundStructure Studio is Polycom’s next-generation installed audio design and configuration software. Using SoundStructure Studio makes it possible to get your installed audio solutions up and running faster than you ever thought possible. It also makes it easier than ever to add and remove input and output signals through a single matrix view on one screen, and to provide default signal routings and gains based on the type of input or output signal.

The matrix page (above) shows the signal routing from inputs to outputs for all audio channels in the system. Groups of multiple signals can be collapsed and configured as a single channel or expanded and configured individually.

SoundStructure Studio channels page (left) allows for easy access to the settings for all the inputs and outputs for the entire system.

Polycom, Inc.

1.800.POLYCOM

www.polycom.com

© 2012 POLYCOM, INC. ALL RIGHTS RESERVED. POLYCOM®, 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.

805_0612

DATA SHEET

Polycom

®

SoundStructure SR12

Truly immersive audio experience for voice and video conferencing

The Polycom SoundStructure SR12 installed audio solution delivers powerful and flexible audio processing for any commercial sound application that does not require conferencing capabilities. Either alone or linked with up to seven additional systems, the SoundStructure SR12 system is an ideal solution for any sound application, including houses of worship, stadiums, conference centers, hotels, night clubs, and restaurants. A powerful gain sharing automixer with automatic gain control on all inputs ensures consistent microphone pickup in all environments and flexible equalization, dynamics, and cross-over processing enable a broad range of applications. Feedback elimination on all inputs prevents embarrassing acoustic feedback and provides for flexible microphone and loudspeaker placement, and unrivaled noise cancellation technology removes the broadest range of background noises. A central matrix mixer that seamlessly scales from 12 up to 96 inputs and outputs makes setting up multi-zone audio systems a snap.

In addition, the SoundStructure model SR12 features the same flexibility and ease of installation as the SoundStructure C-Series conferencing products, and is a perfect way of adding 12 additional nonconferencing inputs (such as other line level audio sources) and outputs to a SoundStructure conferencing system. An innovative OBAM matrix architecture enables multiple SoundStructure units to work together as one large system for unparalleled scalability and flexibility. All SoundStructure products leverage advanced signal grouping, labeling and submixing that let you leverage yesterday’s work during today’s installation, saving time and money. Plus, exclusive SoundStructure

Studio software from Polycom makes configuration easy, and is powerful enough to handle the most challenging acoustic designs.

About Polycom

Polycom is the global leader in standards-based unified communications (UC) solutions for telepresence, video, and voice powered by the Polycom ® RealPresence ® Platform.

The RealPresence Platform interoperates with the broadest range of business, mobile, and social applications and devices. More than 400,000 organizations trust Polycom solutions to collaborate and meet face-to-face from any location for more productive and effective engagement with colleagues, partners, customers, specialists, and prospects. Polycom, together with its broad partner ecosystem, provides customers with the best TCO, scalability, and security for video collaboration, whether onpremises, hosted, or cloud-delivered. Visit www.polycom.com or connect with Polycom on Twitter, Facebook, and LinkedIn.

Benefits

• Breakthrough feedback elimination –

Enables more flexible microphone, talker, and loudspeaker placements and speech reinforcement

• Unrivaled noise cancellation

technology – Removes the broadest range of background noises from your audio inputs

• Gain sharing automatic microphone

mixer – Improved automixer experience ensures smoother transitions and robust performance in a variety of operating environments

• OBAM™ matrix architecture – Connect multiple units together to create larger systems without the limitations of traditional bussing

• SoundStructure Studio – Powerful

Windows-based software for efficient design and configuration; use it to easily set up a basic system, or use its powerful customization tools for more complex environments

• Integrated Ethernet – Control and manage the system from anywhere on the network

Polycom SoundStructure SR12 Specifications

Block Processing Diagram

Submix Input from Matrix

Submix Processing

Dynamics

Processing

Parametric

Equalisation

Fader Delay

Submix output to Matrix

Analogue s

Back of Unit To Telco from Matrix s

Telephony Processing

To Telco from Matrix

Delay

Tone

Generator

Fader s

Analogue

Dynamics

Processing

Parametric

Equalisation

Delay Fader

Parametric

Equalisation

Dynamics

Processing

Automatic

Gain Control

Call Progress

Detection

Noise

Cancellation

D/A

Converter

Analogue

Gain

Output to

PSTN Line

Line Echo

Cancellation

A/D

Converter

Analogue

Gain

Intput from

PSTN Line

Dimensions

•  19” x 13.5” x 1.75” in (483 x 1343 x 45 mm)

(W x L x H) (one rack unit)

Weight

•  12 lbs. (5.5 kg) dry, 14 lbs. (6.4 kg) shipping

•  Connectors

•  RS-232: DB9F

•  OBAM In/Out: IEEE 1394B

•  CLINK2 : RJ45

•  LAN: RJ45

•  Control /Status: DB25F

•  Audio: Mini (3.5mm) quick connect terminal blocks

•  IR Receive: Mini (3.5mm) quick connect terminal block

Power and Thermal

•  Internal supply

•  Input voltage of 90-250 VAC; 50-60 Hz; line power requirements (including 0.6 PF):

105VA (SR12)

•  Thermal dissipation (Btu/hr): 215 Btu/hr

•  0 to 40°C operating temperature

Inputs

•  Phantom power: 48 V DC through

6.8kOhm series resistor per leg, 7.5mA per channel, software selectable

Polycom, Inc.

1.800.POLYCOM

www.polycom.com

•  Analog input Gain: -20 to 64 dB on all inputs in 0.5 dB teps, software adjustable

•  Maximum input amplitude: +20.4 dBu,

1% THD + N

•  Nominal level: 0 dBu (0.775V rms)

•  Equivalent input noise: <-122 dBu, 20 -

20,000 Hz, Rs=150 Ohms (1%)

•  Input Impedance: 10 kOhms

•  Input EMI Filter: Pi filter on all audio inputs

Outputs

•  Output Gain: -100 to 20 dBu in 1 dB steps, software adjustable

•  Maximum output amplitude: +23 dBu,

1% THD + N

•  Nominal output level: 0 dBu (0.775 V rms)

•  Output impedance: 50 Ohm, each leg to ground, designed to drive loads > 600 Ohms

•  Output EMI Filter: Pi filter on all audio outputs

System 1

•  Frequency response:

20-22,000 Hz, + 0.1 /- 0.3 dB

•  Idle channel noise:

<-109 dB FS no weighting, 20 - 20,000 Hz,

-60dB FS, 0.997 kHz input signal,

0 dB gain

•  Dynamic range:

>109 dB FS no weighting, 20 - 20,000 Hz,

-60 dB FS, 0.997 kHz input signal, 0 dB gain

•  Linearity: 0 dB FS to -122 dB FS +/- 1 dB

•  THD+N: < 0.005%, -20 dB FS input signal

•  Common Mode Rejection Ratio:

<-61 dB, 20 - 20,000 Hz, no weighting

•  Cross talk: <-110 dB, 20-20,000 Hz, 1kHz, channel-to-channel

•  Latency: Mic/Line inputs to outputs:

5msec, 20 msec with NC processing enabled

•  Noise cancellation: 0-20 dB, software selectable

•  Control Inputs: Contact closure

•  Status Outputs: Open collector 60V and

500mA maximum per output

•  All signal ground pins connected to chassis ground through low impedance planes

1 Unless noted, all values are valid for all channels at 0dB input gain

© 2012 POLYCOM, INC. ALL RIGHTS RESERVED. POLYCOM®, 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.

809_0612

Polycom

®

Vortex

®

EF2201

DSP-based phone add for installed conferencing applications

The ideal “phone add” for installed conferencing applications, the EF2201 makes telephone interfacing easy. With a variety of control options, built-in DTMF dialer, and one-cable connection to other Vortex products, the EF2201 simplifies the addition of telephone calls to your audio or video conference.

When participants can’t make it to a video conference, or can only drop in for a short time, it’s easy to accommodate their needs. The Vortex EF2201 links directly to an analog (PSTN) telephone line, connecting outside callers to your meeting. In addition to providing crystal-clear telephone audio to all participants, the Vortex EF2201 significantly reduces background noise on the phone line, keeping unwanted sounds from interfering with your conference. As the next generation of digital hybrid telephone interfaces, the Vortex EF2201 interfaces with other Vortex products through a digital expansion bus.

Telephony signals are communicated with a Vortex EF2280 over the digital

P-bus channel of the EF Bus, without the need for analog input or output on the EF2280. The EF2201 can be controlled via RS232, Vortex Expansion bus, and front panel controls. Additional features are audible ring indications, user-configurable entry and exit tones, up to 256 speed dial memories, and controllable levels of DTMF and dial tone signals. The full cross-point matrix allows users to determine how the input and output signals are sent to and from the telephony interface. Settings are configured with Conference

Composer software (provided), and can be stored as user presets.

The direct-link audio choice for The Polycom Office™

With integrated video, voice, data, and web capabilities, The Polycom Office is the only solution that offers you an easy way to connect, conference, and collaborate any way you want. The Polycom Office is our commitment to making distance communications as natural and interactive as being there.

Work faster, smarter, and better with the Polycom Vortex EF2201 and The

Polycom Office.

Easily add calls to conferences – Utilise telephone hybrid with noise cancellation to add telephone callers

Simple, fully digital connection – Between

Vortex EF2200 series products with EF Bus, no analogue input / output is required

Productivity-enhancing features – 256 speed-dial memory, selectable auto hang-up via loop drop and progress tone detection

Use it anywhere in the world – Country-specific telephone configuration settings

Consistent audio quality –

Automatic gain control on the telephone audio receive path

Polycom

®

Vortex

®

EF2201

System Block Diagram and Rear Panel

EF Bus Matrixing

PEQ

Tone

Gen.

LEC

D/A

A/D

TLP

TLP (Telephone Line Processing)

PEQ

Parameters

AGC

On Off

Noise

Canceller

Parameters

Call Prog

Detect

Parameters

Mute

On Off

RS-232

Control/Status

Telephone Line

Phone

Polycom Vortex EF2201 Technical

Specifications

Microphone Input

• -30 dBu to +0 dBu/ -66 dBu to -33 dBu, nominal; software selectable

Dimensions

• 483 mm (19”) W x 244 mm (9.6”) L x

45 mm (1.75”) H (one rack unit)

Control Inputs

• Contact closure

Weight

• 1.8 kg (4 lbs.) dry, 2.5 kg (5.5 lbs.) shipping

Connectors

• RS-232: DB9F

• EF Bus In/Out: RJ45

• Control /Status: DB25F

• Telephone Line/Set: RJ11 (2)

Power

• External supply (provided)

• Input voltage of 110-240 VAC;

47-63 Hz; power consumption 15 W

(typical)

Status Outputs

• 5 V, 20mA each

Headroom

• 20 dB, nominal

Frequency Response

• 250 Hz - 3.6 kHz on telephony

Interface Line Echo Canceller

(telephone hybrid)

• Echo cancellation 40 dB, total 60 dB

• Convergence 30 dB/second

• Cancellation span 32 msec

Typical EF2201 Installation

EF Bus Telephone Line

©2008 Polycom, Inc. All rights reserved.

Polycom, the Polycom logo, and Vortex are registered trademarks and The Polycom

Office is a trademark of Polycom, Inc. in the U.S. and various countries. All other trademarks are the property of their respective companies. Specifications subject to change without notice.

www.polycom.com

Polycom EMEA

270 Bath Road, Slough,

Berkshire SL1 4DX, UK

(T) +44 (0)1753 723000

(F) +44 (0)1753 723010

Polycom Headquarters

4750 Willow Road,

Pleasanton, CA 94588

(T) 1.800.POLYCOM

(765.9266) for North America only

Polycom Asia Pacific

8 Shenton Way

#11-01 Temasek Tower

Singapore 068811

+65.6389.9200

For your local Polycom office please visit www.polycom.com

Part No. 3726-82201-002 Rev. 03/03

Polycom

®

Vortex

®

EF2280

Acoustic echo/noise canceller with automatic microphone/matrix mixer

Benefits

Industry leading audio quality – Design enhancements improve the audio experience for even the most demanding listening environments; support for the broadest range of microphones in the industry

Hear every word with Polycom’s industry-leading acoustic echo and

noise cancellation – Ensures the highest possible audio quality between sites with

20 kHz bandwidth, 270 ms tail length, more than 10 dB of room gain, and a convergence rate of over 40 dB per second on each mic/line channel

Integrated voice and video

solution – May be linked with additional

Vortex Installed Voice products including

Vortex EF2201, Vortex EF2210, Vortex

EF2211, the Vortex EF2241 and Vortex

EF2280; Polycom’s SoundStation VTX

1000™ wideband conference phone and

Polycom’s video conferencing systems including the VSX™ 7000 and 8000 product lines

Easy setup – Conference Composer™ and

Polycom Instant Designer™ software make configuration fast and simple

Easy to operate – Controllable via

Crestron ® , AMX ® or other room control systems, RS-232, the Vortex EF-IR11 infrared controller, or from the unit’s front panel; 16 factory presets and 32 user configurable presets allow quick selection of operating parameters

Adaptable to user requirements – Userprogrammable delays on all outputs, 5-band parametric EQ on all inputs and outputs, intelligent Automatic Gain Control, 256 macros, system diagnostics, and other tools ensure the best performance for all room conditions

Enjoy clear, understandable conversation in all conferencing applications.

The Vortex EF2280 improves audio quality for conferencing, distance learning, and other installed applications through industry-leading acoustic echo and noise cancellation - with support for up to 8 microphones.

The Vortex EF2280 automatically mixes microphones and other audio sources while cancelling acoustic echoes and annoying background noise. It is typically installed at each site in a multi-site network and is commonly used in applications such as boardrooms, courtrooms, distance learning, sound reinforcement, and room combining. It connects easily to other equipment such as codecs, VCRs, or other A/V products.

The unit can be programmed from the front panel, or through Conference Composer™ software (included).

Conference Composer's Instant Designer™ wizard ensures fast, accurate setup for a variety of applications.

A single Vortex EF2280 unit provides automatic mixing of up to eight microphones plus four auxiliary audio sources. If more microphones are needed, additional Vortex EF2280 or Vortex EF2241 units can be linked, up to a total of eight units. Number of Open Microphones (NOM) information can be specified across all channels in the linked units. The microphone channels feature industry-leading acoustic echo cancellation to prevent retransmission of signals to their original locations. A neural network Automatic Gain Control

(AGC) reacts only to valid speech patterns, bringing voices within desired levels. AGC controls are user adjustable, as are settings for the five-band parametric EQ offered on all input and output channels and output delay controls.

The high-quality audio choice to access the power of Polycom unified collaborative communications solutions.

With the greatest breadth and depth of integrated video, voice, and Web solutions, only Polycom delivers the ultimate communications experience. Our market-leading conferencing and collaboration technologies, supported by world-class service, enable people and organizations to maximize their effectiveness and productivity. Add to that the most experience and proven best-practices in the industry, and it’s clear why

Polycom has become the smart choice for organizations seeking a strategic advantage in a real-time world.

Connect. Any Way You Want.

Polycom Vortex EF2280 Specifications

System Block Diagram and Rear Panel

Dimensions

• 19” (483 mm) W x 9.6” (244 mm) L x 1.75” (45 mm) H (one rack unit)

Weight

• 4 lbs. (1.8 kg) dry, 5.5 lbs. (2.5 kg) shipping

Connectors

• RS-232: DB9F

• EF Bus In/Out: RJ45

• Control /Status: DB25F

• Audio: Mini (3.5mm) quick connect terminal blocks

Power

• External supply (provided)

• Input voltage of 110-240 VAC; 47-63 Hz; power consumption 30 W

Polycom Headquarters: www.polycom.com

Polycom EMEA:

Polycom Asia Pacific:

Inputs

• Phantom power: 48 V DC, software selectable

• Analog input Gain: 0 to 30 dB Mic/line inputs in 1 dB steps, software adjustable

• Mic/Line switch gain: 33 dB

• Maximum input amplitude: +19 dBu, 1% THD + N

• Nominal level: 0 dBu (0.775V rms)

• Equivalent input noise: <-124 dBu, 20 - 20,000 Hz

• Input Impedance: 10 kOhms

• Input EMI Filter: Pi filter on all audio inputs

Outputs

• Output Gain: -100 to 20 dBu in 1 dB steps, software adjustable

• Maximum output amplitude: +23 dBu, 1% THD + N

• Nominal output level: 0 dBu (0.775 V rms)

• Output impedance: 33 Ohm, each leg to ground

• Output EMI Filter: Pi filter on all audio outputs

System*

• Frequency response: 20-20,000 Hz, + 0.2 /- 0.3 dB

• Idle channel noise: <-100 dB FS "A" weighted, 20 - 20,000

Hz, 0 dB gain

• Dynamic range: >100 dB FS "A" weighted, 20 - 20,000 Hz,

0 dB gain

• Linearity: 0 dB FS to -110 dB FS +/- 1 dB

• THD+N: < -90 dB FS

• Common Mode Rejection Ratio: <-61 dB, 20 - 20,000 Hz, no weighting

• Cross talk: <-104 dB, 20-20,000 Hz, channel-to-channel

• Latency: Mic/Line inputs to outputs: 13 ms, processing enabled

• Acoustic Echo Cancellation Span: 270 ms

• Total Cancellation: >65 dB

• Convergence Rate: 40 dB/second

• Noise cancellation: 0-15 dB, software selectable

• Operating Temperature: 0° - 40° C

• Control Inputs: Contact closure

• Status Outputs: 5 V, 20 mA each

*Unless noted, all values are valid for all channels at line level

©2004 Polycom, Inc. All rights reserved.

Polycom, the Polycom logo and Vortex are registered trademarks and SoundStation VTX 1000, Conference Composer, InstantDesigner and VSX are trademarks of Polycom, Inc. in the U.S. and various countries. All other trademarks are the property of their respective companies. Specifications subject to change without notice. For Architect's and Engineer's

Specifications, please visit Polycom.com, or contact the Installed Voice Business Group at 1.770.350.4140.

4750 Willow Road, Pleasanton, CA 94588 (T) 1.800.POLYCOM (765.9266) for North America only.

For North America, Latin America and Caribbean (T) +1.925.924.6000, (F) +1.925.924.6100

270 Bath Road, Slough, Berkshire SL1 4DX, (T) +44 (0)1753 723000, (F) +44 (0)1753 723010

Polycom Hong Kong Ltd., Rm 1101 MassMutual Tower, 38 Gloucester Road, Wanchai, Hong Kong, (T) +852.2861.3113, (F)+852.2866.8028

Part No. 3726-82280-001 Rev. 10/04

Polycom VSX 8000

The ultimate in an easy-to-install, high performance video conferencing system

Polycom VSX 8000 Benefits

Form Factor – Sleek 1U rack mount design that's perfect for integrated environments

Professional-Grade Connectivity –

Phoenix-style audio connectors for balanced, line-level input and output as well as S-Video, serial and VGA input and output for video

Solutions Oriented Interoperability –

Designed to seamlessly integrate with

Polycom Vortex Installed Voice Products,

SoundStation VTX 1000™ conference phone and Polycom MGC™ audio and video bridge products

Versatility in Peripheral Choice – Add on a touch panel control device, teletype device for closed-captioning, medical control device or secure encryption devices

Modular Network Interfaces – Use the

10/100 Ethernet connection built-in or plug in a QBRI or PRI ISDN network module or a Serial (V.35/RS-530/RS-449) network module

The Polycom Solution – Everything you need to deploy and manage a complete video conferencing network with Polycom

Global Management System™, Polycom

PathNavigator™, Polycom Conference Suite and the Polycom MGC™

A sleek rack mount design, professional grade connectivity and versatile interoperability make this product the perfect choice for integrated and custom environments.

If you are looking for a video conferencing system that combines ease of installation with the highest quality audio and video, the Polycom VSX 8000 offers the perfect marriage of form and function.

The VSX 8000 is specially designed with professional integrators in mind. Starting with the slim, highly integrated,

1U design including: professional grade connectivity, direct VGA input supporting People+Content™ collaboration and versatile slide-in trays to support circuit switched connectivity options (QBRI, PRI, Serial (V.35/RS4449/RS530).

Audio integration flexibility is unparalleled with Phoenix-style connectors, true balanced signals along with the ability to support mixers, direct microphone inputs with phantom power as well the Polycom Digital Tabletop VSX

Microphones. Polycom’s patent pending Siren™ 14 Stereo Audio is an integral part of every system and is easily configured for any room. Video participants can hear far-end participant’s voices project in stereo through left and right speakers for a truly life-like meeting experience.

The Polycom VSX 8000 delivers video performance that will impress even the most demanding customers using the industry’s most advanced video technologies, H.264 and Pro-Motion™. Standards based H.264 video offers outstanding video quality at call speeds up to 512 kbps, providing superior video without weighing down the network.

For higher bandwidth calls, Pro-Motion, high-resolution, television-like video is an excellent choice for the absolute highest quality picture.

The Polycom VSX 8000 offers a rich feature set including an easy to use, customizable user interface, AES encryption software, built-in quality of service and quality of experience with Polycom’s implementation of audio and video error concealment, SNMP support and Gatekeeper and Firewall utilities. Control of the VSX 8000 is accomplished over multiple serial ports for complete API control by external control systems, a full range of IR options as well as control passing with the popular Polycom Vortex Installed Voice Products. All in all, a complete package for integrators to design into a wide range of video conferencing applications.

The ultimate choice for professional integrators. The power of Polycom conferencing and collaboration solutions.

With integrated video, voice, data, and Web capabilities, only Polycom solutions let your customers connect, conference, and collaborate anytime, anywhere and any way they want. It’s our commitment to making distance communications as natural and interactive as being there. Install systems faster, smarter, and more efficiently with Polycom VSX 8000 and other Polycom conferencing and collaboration solutions.

Connect. Any Way You Want.

ITU H.323 and H.320 compliant

Bandwidth

• Maximum Data Rate IP and Serial/V.35:

Up to 2 Mbps

• Maximum Data Rate ISDN: Up to 2 Mbps

Video Standards & Protocols

• H.261, Annex D

• H.263+ Annexes: F, I, J, L, N, T

• H.263++ Annexes: W

• H.264

• ITU 60-fps full screen – Pro-Motion™

Frame Rates (Point-to-Point)

• Intelligently selects frame rate for best performance video

• 30 fps at 56 kbps up to 2 Mbps

• 60 fields per second up to 2 Mbps

Video Inputs: 4 Connectors

• 1 x S-Video; Professional Y/C BNCs (main video camera)

- 1 x DB15; PowerCam™ and PowerCam Plus power, PTZ control, IR, mic input

• 1 x S-Video; 4-pin mini DIN

(second camera with PTZ control)

• 1 x S-Video; 4-pin mini DIN (VCR or DVD player)

• 1 x VGA (Content input from laptop)

Serial Data Port: 2 Connectors

• 2 x DB9

• Control port for custom integration with remote devices such as Crestron and AMX control systems

• Integration with Polycom Vortex® Installed

Voice Products

• Communication port for transmission of serial data

(i.e. medical devices) over ISDN calls

• Auxiliary camera control

Video Outputs: 4 Connectors

• 1 x S-Video; Professional Y/C BNCs (main display)

• 1 x S-Video; 4-pin mini DIN (second display)

• 1 x S-Video; 4-pin mini DIN (VCR or DVD player)

• 1 x VGA (Content display)

Video Formats

• NTSC/PAL

• Graphics: XGA, SVGA, VGA

People Video Resolution

• Pro-Motion interlaced video (60/50 fields full-screen video for NTSC/PAL)

• 4SIF (704 x 480)

• 4CIF (704 x 576)

• SIF (352 x 240)

• CIF (352 x 288)

• Choice of 4:3 or 16:9 display aspect ratios

• Display People on VGA Second Monitor

Content Video Resolution

• XGA (1024 x 768), SVGA (800 x 600), VGA (640 x

480) for Content on VGA displays

• People video support for 4CIF and SIF on

VGA Display

• Up to 4CIF for Content on NTSC/PAL displays

Audio Standards & Protocols

• Polycom Siren 14 Stereo ready

• 14 kHz bandwidth with Siren™ 14 on IP, ISDN, and

IP/ISDN mixed calls

• 7 kHz bandwidth with G.722, G.722.1

• 3.4 kHz bandwidth with G.711, G.728, G.729A

Audio Features

• Seamless integration with Polycom Vortex Installed

Voice Products

• Audio add-in using SoundStation VTX 1000™ conference phone

• Audio add-in over ISDN

• Audio add-in over POTS

• Full-duplex digital audio

• Instant Adaptation Echo Cancellation

• Automatic Gain Control (AGC) – Voice activated

• Automatic Noise Suppression (ANS)

• Ability to turn off Echo Cancellation when external audio equipment is used

Polycom VSX 8000 Specifications

• Audio Mixer (Mic, VCR, line-in)

• Built-in tonal speaker test

• Real-time audio level meter for local and far-end microphones

• Microphone and VCR input audio mixing

• Ability to talk over VCR audio

Audio Inputs: 6 Connectors

• 1 x Conference link

- Supports up to (3) microphones

- Supports SoundStation VTX 1000 conference phone

• 2 x RCA/Phono, line level input for VCR, DVD player or audio mixer

• 2 x Phoenix connectors; balanced line level or direct microphone inputs with 24 V Phantom power

• 1 x RJ-11 for analog speaker telephone

Audio Outputs: 6 Connectors

• 2 x Phoenix connectors; balanced line level output

• 2 x RCA/Phono, Line Level output for VCR record

• 2 x RCA/Phono, Line Level output for speakers

Other ITU-Supported Standards

• H.221 communications

• H.224/H.281 far-end camera control

• Annex Q standard for FECC in H.323 calls

• H.225, H.245, H.239, H.241

• H.231 in multipoint calls

• H.243 MCU password

• H.233, H.234, H.235V3 encryption standards

• Bonding, Mode 1

Network Interfaces Supported

• IP (LAN, DSL, cable modem)

• Single 10/100 Ethernet port (10 bps/100Mbps/Auto)

• Optional ISDN QBRI (Basic Rate Interface) Module

• Optional ISDN PRI (Primary Rate Interface) Module

T1/E1

• Optional Serial Module (V.35/RS-530/RS-449 with

RS-366 dialing)

Network Features

• Integration with Cisco Systems' CallManager

Version 4.0

• Automatic IP/ISDN calling

• Down speeding over IP and ISDN

• Audio & Video Error Concealment over IP and ISDN, mixed calls

• IP address conflict warning

• Fast Connect IP for quick video connections

• Maximum call length digital timer

• Auto SPID detection and line number configuration

• MGC Click&View™ for individual screen layouts

• Polycom OneDial™ intelligent call management attempts call on preferred network (IP or ISDN) and automatically rolls over to secondary network if needed

• Polycom PathNavigator™ support for easy call placement and network cost optimization

• TCP/IP, DNS, WINS, DHCP, ARP, HTTP, FTP, Telnet

• Basic SIP* (Session Initiation Protocol) implementation

• Chair control through API command or Integrated

Web Interface

• Software Upgradeable Inverse Multiplexer (IMUX)

Conference on Demand

• Initiates unscheduled MGC calls from the endpoint

• Utilizes Polycom Office (PathNavigator and MGC)

• Auto selects either the internal or external bridge

• Dials all participants simultaneously

Security

• COMSEC tested by Titan Systems, Information

Security Systems Division, and independently validated to operate with approved government encryption technologies

• KG-194/KIV-7 encryptor support with on screen and address book dialing

• Enhanced integration for independently certified, classified encryption devices

• Account validation number entry

• Secure password authentication

• Unique factory default passwords

• Administrator password

• Dial-in meeting password

• Encrypted password for VSX Web access

• Ability to disable remote interfaces (FTP, Telnet,

HTTP, SNMP)

• Ability to disable mixed protocol multipoint calls

• Auto-Answer (On/Off)

• Allow access to user settings (On/Off)

Embedded Encryption

• Advanced Encryption Standard (AES)

• FIPS validated by National Institute of Standards

& Technology (NIST) certified agency

• 128-bit key length

• AES software encryption on ISDN, IP and Serial/V.35

up to 2 Mbps

• Standards-based H.235V3 (IP)

• Standards-based H.233/H.234 (ISDN/Serial)

• Automatic key generation and exchange

• Supported in People+Content™

• Supported in Point-to-Point IP, ISDN and mixed network calls

User Interface

• User-friendly graphical interface

• Customizeable home screen and color themes

• Kiosk mode with scrolling marquee

• User-selectable camera icons and ring tones

• Up to (99) user defined camera presets

• Speed Dial List on home page

• Persistent Preview (Far-site PIP) on all screens

• Picture-In-Picture (PIP) (On/Off, Moveable)

• Dual-Monitor Emulation

• Numerical menu navigation (Similar to mobile phone)

• Date, Time Server accessibility

• Calendar and Conference scheduling

• Alert Signal on home page

• Do Not Disturb (On/Off)

• VSX Web for remote monitoring

Directory Services

• 4,000+ number global directory

• 1,000+ number local directory

• Limitless multipoint entries

• Live address book with Polycom Global Directory

Services automatically and quickly updates directory with address changes or new endpoints

• Live address book with Polycom Global Directory

Services automatically and quickly removes endpoints from directory if they are turned off

• Polycom Global Directory Services integrates with

Active Directory/LDAP

• Directory Server backup in the event Polycom Global

Directory is not accessible

• Automatic ISDN localization of calls

System Management

• SNMP for enterprise management

• Diagnostics and software upgrades via PC, LAN

• Integrated VSX Web management tool

• Web Director: Remote administrator video monitoring and control from VSX Web (enabled/disabled from endpoint for security)

• Out-of-box setup from VSX Web

• Place a call from VSX Web

• Language independence between set-top interface and VSX Web

• System configuration from VSX Web

• Recent Calls Log – Records last 99 incoming and outgoing calls

• Call Detail Record (CDR) – Reports all incoming and outbound calls along with call statistics

• CDR Feature On/Off

• Downloadable CDR data for processing requires no external management system

• Account number validation at call initiation integrated with Polycom Global Management System™ for billing purposes

• Administrator-configurable dialing speeds

• Complete support for The Polycom Office™ including:

- Polycom Global Management System

- Polycom OneDial

- Polycom PathNavigator

- Polycom Conference Suite

- Polycom MGC

- Polycom SoundStation VTX 1000

- Polycom Vortex®

Quality of Service and Experience – iPriority™

• Video Error Concealment

• Audio Error Concealment

• Universal Plug and Play (UPnP)

• IP Precedence (ToS)

• DiffServ (DSCP) (COS)

• Dynamic Bandwidth Allocation

• Proactive Network Monitoring

• Packet and jitter control

• Network Address Translation (NAT) support

• Automatic NAT discovery

• Configurable video/audio/FECC service value

• Asymmetric speed control

• Alternate Gatekeeper support

• TCP/UDP fixed-port firewall support

• Lip synchronization

• Echo cancellation

• Echo suppression

• Auto gatekeeper discovery

• Automatic gateway dialing profiles

• Specify outbound call routing for gateway/ISDN

Collaboration Solutions

• Closed captioning support

• Web Streaming in and out of a call, RTP based, suitable with QuickTime players

System Options

• People+Content (using ImageShare II)

• PowerCam™ Plus Camera

• PowerCam Camera

• Polycom Digital Tabletop Microphone

• Internal Multipoint Feature

Language Support (12 languages)

Chinese (Simplified), Chinese (Traditional), English,

French, German, Italian, Japanese, Korean,

Norwegian, Portuguese, Russian, Spanish

- Documentation translations in all languages

- User interface translations in all languages

- Keypad audio dialing confirmation in all languages

- VSX Web translations in all languages

- Remote controls labeled in all languages

Electrical

• Auto sensing power supply

• Operating voltage/power 90-250 VAC, 47-63 Hz/

80 watts

Environmental Specifications

• Operating Temperature: 0-40˚ C

• Operating Humidity: 15-80%

• Non-Operating Temperature: -40-70˚ C

• Non-Operating Humidity (Non-condensing): 10-90%

Physical Characteristics

• Video Base Unit (W/H/D): 17.25"/1.73"/9.68";

438.15 mm/43.83 mm/245.85 mm

• Video Base Unit Weight: 8.4 lbs; 3.8 kg

Polycom VSX 800 Warranty

• One-year return to factory parts and labor

• One-year software updates and upgrades

* SIP video extensions are in the process of being standards ratified.

VSX 8000 shown with optional QBRI network module installed

©2004 Polycom, Inc. All rights reserved.

Polycom, the Polycom logo and Vortex are registered trademarks and VSX, Polycom PathNavigator, Global Management System, Siren, Pro-Motion, MGC, Click&View, Polycom

OneDial, People+Content, The Polycom Office, iPriority, PowerCam and SoundStation VTX 1000 are trademarks of Polycom, Inc.

All other trademarks are the property of their respective owners. Information in this document is subject to change without notice.

www.

ivci.com 1-800-224-7083

Part No. 3726-17107-001 Rev. 6/04

Polycom

®

Vortex

®

Products

Frequently Asked Questions

What is the Vortex line?

Polycom Vortex products provide advanced acoustic echo and noise cancellation, automatic microphone mixing, room zoning, and other functions for installed conferencing and collaboration applications.

Designed specifically for system integration, the Vortex line installs easily with a wide range of video codecs, room audio components, and controllers such as AMX® and Crestron®.

What products are available in the Vortex line?

Vortex EF2280 - provides 8 mic/line level inputs, 4 line level inputs,

12 line level outputs

Vortex EF2241 - provides 4 mic/line level inputs, 4 line level inputs,

8 line level outputs, a 10 Watt power amplifier, and a DSP-based telephone hybrid for phone adds

Vortex EF2211 - provides 1 mic/line level input, 2 line level inputs, 3 line level outputs, a 10 Watt power amplifier, and a DSP-based telephone hybrid

Vortex EF2210 - provides 1 mic/line level input, 2 line level inputs, and 3 line level outputs

Vortex EF2201 - a DSP based telephone hybrid for phone adds, designed to work in conjunction wih other Vortex products

What do the Vortex model numbers mean?

The Vortex products are numbered EF22XY where:

EF = Echo Free (a Polycom trademark)

22 = 22kHz wideband audio

X = the number of microphone inputs

Y = the number of phone adds (analog telephone lines)

For example, the Vortex EF2241 provides 22kHz wideband audio, permits direct connection of 4 microphones, and has an integrated phone add.

When should I use a Vortex product vs. tabletop?

A Vortex solution should be selected when you need to have multiple microphones around the room, including wireless and ceiling microphones, or distributed loudspeakers to provide full room coverage.

In larger rooms, 20'x20' and above, or when there are many participants, additional microphones and loudspeakers may be required to allow all participants to hear and be heard clearly. Also, if multiple telephone lines are required to allow multiple simultaneous callers, the Vortex products should be used as they allow you to have up to

8 simultaneous telephone callers. If there are requirements for wireless microphones, podium microphones, or multiple video codecs, a Vortex should be used.

When using video conferencing, the Vortex will improve the audio quality of your installations and can be used with nearly all Polycom video codecs.

If the highest audio quality is a requirement, the Vortex will give you the highest quality, the most flexibility, and the expandability to support your future audio and conferencing needs.

How do I know which Vortex product to specify for my application?

If you are designing a room with several microphones, plan on using the Vortex EF2241 or Vortex EF2280. These products provide individual channel acoustic echo and noise cancellation, giving you the best possible voice quality for multi-mic applications.

If you have a mixer already and would like to add teleconferencing, the Vortex EF2211 (with built-in telephony interface) or Vortex

EF2210 will do the job nicely.

To add PSTN (analog) telephone calls to a voice or video conference, you can use the Vortex EF2241, Vortex EF2211, or the Vortex

EF2201. Vortex products can also interface with the Polycom

SoundStation VTX 1000 TM , which gives users a familiar telephone controller and can provide wideband (7kHz) telephone audio when connected to another SoundStation VTX 1000 on the far end.

How many Vortex products can be linked together?

Up to 8 units* can be linked. This means you can use up to 64 microphones in a system using 8 Vortex EF2280 units. NOM (number of open microphones) information can be specified across all linked Vortex units.

*The Vortex EF2280 and Vortex EF2201 phone add can share the same device ID, allowing you to have up to 8 Vortex EF2280 units

AND 8 Vortex EF2201 units, if desired.

Connect. Any Way You Want.

Polycom Vortex

Frequently Asked Questions

What other Polycom products can be interfaced to Vortex products?

The Polycom SoundStation VTX 1000 wideband conference phone can be used to add wideband telephony audio to a Vortex. The SoundStation VTX 1000 user interface can be used to dial phone calls, mute the audio, and change the volume of the incoming wideband telephone audio.

In addition, the new Polycom VSX™ 8000 video conferencing codec interfaces easily to the Vortex, with its infra-red remote control capable of executing commands within the Vortex for volume control and muting Vortex microphones.

All Polycom video codecs, as well as those made by other companies, interface easily with Vortex products, and Vortex can also be easily controlled with AMX or Crestron control systems.

Are Vortex products difficult to install and set up?

Not at all! All connections use industry-standard connectors (Phoenix, DB9, etc.), and our Instant Designer™ setup wizard makes it possible to configure even the most complex systems in a short time.

How does the Polycom Instant Designer wizard work?

A recent addition to our Conference Composer™ design software for

Windows®, the patent pending Polycom Instant Designer guides the system designer through the steps required to create high performance Vortex audio conferencing and sound reinforcement solutions.

The A/V integrator or consultant simply chooses the necessary inputs, outputs

(including the new Polycom VSX 7000 or VSX 8000 video codecs) and optionally the sound reinforcement zones needed for the desired system, and Polycom

Instant Designer automatically selects the appropriate Vortex devices necessary to implement the system, maps the inputs and outputs to the devices, and creates the Conference Composer design files required to implement the design. It is easy to upload these design files to the Vortex devices to complete the configuration. All of this can be done within a matter of minutes.

The Polycom Instant Designer handles all the details of creating zone to zone gains for sound reinforcement, configuring the acoustic echo cancellers, interfacing to common audio and video equipment, bussing between devices, configuring presets, creating volume control macros, and preconfiguring logic ports for push to talk microphones.

For more information, see the Instant Designer application note (it may be downloaded in PDF format via the Polycom Resource Center on the internet).

How do I get a copy of the Instant Designer software?

The Instant Designer software is included in the Conference Composer software and can be downloaded from the Polycom Resource Center at http://extranet.polycom.com.

Is a remote control available for Vortex products?

Yes, the optional Vortex EF-IR11 infrared controller provides direct control of the

Vortex EF2280, EF2201, EF2211 and EF2241. It can be used to adjust volume, mute and unmute audio, activate a phone line, dial out or answer calls, or execute any of 35 macros that are user-programmed into the Vortex units. And, as noted earlier, Vortex products are easily controlled with external room controllers such as Crestron or AMX.

What makes the Vortex Acoustic Echo Cancellation (AEC) better than competing units?

The Vortex AEC has been designed to get the best trade-off between full-duplex audio (quick interaction of both sides of a conversation) and removal of echo when both sides of the conversation are moving about their rooms while both sides are speaking. This is a measure of the doubletalk performance. Vortex products also have the fastest convergence speed of any acoustic echo canceller on the market.

What is convergence speed and why is it important?

Convergence speed is how fast the AEC can adapt and "lock" onto to the local echo and effectively begin removing it. You can think of convergence speed as the rate at which audio from the remote site played into the local room will be attenuated by the AEC before being sent back to the remote site. This is related to convergence time, which is the amount of time it takes the AEC to reduce the echo to some level. With most echo cancellers, before the AEC locks into the echo, there is some amount of clipping that happens to the out going audio to prevent an echo from being sent back to the remote side. This clipping makes it difficult to have a natural full-duplex conversation. Because participants in a room are constantly moving, or the microphones are being moved or muted, or the volume is being changed, the echo path is always changing.

Faster convergence means the AEC can lock onto changes in the room faster and sound better faster.

Why does Polycom's Vortex sound better than other units?

There are several reasons including the full bandwidth of the signal (22kHz), the digital trim pots that adjust the analog gain before the A/D converter (this maximizes the signal to noise ratio), and our patented ambient noise cancellation.

The Vortex's neural network-based AGC also helps the AEC performance by bringing all signals to the proper level. Vortex units also provide better room gain performance than competing products.

What is room gain and why is it important?

Room gain is the difference in level (in dB) between the outgoing level from the room device to the incoming level to the room device. Factors that affect room gain are the amount of amplification in the loudspeaker, the amount of gain applied to the microphone inputs, and how closely coupled the microphones and loudspeakers are.

Polycom Vortex

Frequently Asked Questions

Room gain is important because the acoustic echo canceller compares the signal that is sent to the amplifier/loudspeaker to the signal picked up by the microphones. If the signal picked up by the microphones is much larger than the signal sent to the amplifier/loudspeaker (which is the case if the room gain is positive), many AECs have a tougher time removing the echo and may become less full duplex.

For more information on room gain, see the Polycom application note on the

Vortex EF2280.

How is Polycom's noise canceller different from other products that say they reduce noise?

Many products remove noise by filtering. When this happens, they will also remove any signal in the same frequency band as the noise. The patented

Polycom noise canceller is different from other products because it uses an adaptive frequency selective algorithm to remove the noise only where there is noise and to leave the desired signals (speech, audio, etc.) alone.

What type of automixers do the Vortex EF2280 and Vortex EF2241 use?

The automixer -- or more accurately, both automixers, since you can split the mixer in each product-- in the Vortex EF2280 and Vortex EF2241 use a gating style of automixing. The automixers make a determination of which signals should be gated on and this decision gets translated into a decision of what gain to apply to the input signal as it gets mixed with other signals to create an output signal in the matrix. The automixer takes in up to 8 microphone/line inputs in the Vortex EF2280 (4 in the Vortex EF2241) and produces the same number of line output signals and gain information to be applied to each of these gated outputs as they are used in the matrix. If you select the gated version of an input at a cross point in the matrix then you will get the signal scaled by the gain that the automixer has determined should be applied to that signal.

What types of filters and other parameters do the Vortex products provide?

User-programmable delays on all outputs, 5-band parametric EQ on all inputs and outputs, 256 macros, 16 factory presets, 32 user-configurable presets, system diagnostics and other tools are provided to ensure the best performance for all room conditions.

The Polycom noise canceller works best with steady state signals, i.e., signals that are random, but their randomness can be thought of as fixed. Example of this are white or pink noise - the signal is random but the type of randomness is known; periodic signals such as tones - the signal is steady state; and crowd noise - a large collection of voices, when mixed together, sound like random background noise. Combinations of these types of signals can also be effectively removed such as the noise from power drills - there is a combination of periodic signals and random noise. The types of noise that the Polycom noise canceller cannot remove include short impulsive noises, like a hammer blow, which are not present over a long enough time window to have the algorithm adapt and remove it.

How does the Polycom AGC (Automatic Gain Control) work, and how is it different from competing products?

Polycom's AGC algorithm uses a neural network to help make decisions about what is speech and what is not speech. A neural network is an artificial intelligence concept that is a form of nonlinear adaptive filter with a built-in state machine. The neural net helps make a much better judgment on whether the input data is speech-like every couple of hundred milliseconds. The Polycom algorithm looks at the input signal and makes a determination as to whether the signal is a valid speech input or not. Once it finds valid speech (and it is always looking for valid speech), it will estimate the nominal level of the incoming signal. It will then determine what level to apply to the signal to get it to a

0dBu nominal level. Other AGC algorithms may simply use the envelope of the signal to determine if the signal should be increased or decreased. This has the unintended consequence of training on background noise and making the signal too loud if there is prolonged silence.

Polycom Headquarters: www.polycom.com

Polycom EMEA:

Polycom Asia Pacific:

© 2004 Polycom,Inc. All rights reserved.

Polycom, the Polycom logo design and Vortex are registered trademarks and Conference Composer, InstantDesigner, VSX, and SoundStation VTX 1000 are trademarks of Polycom, Inc. in the U.S. and various countries.

All other trademarks are the property of their respective companies. Specifications are subject to change without notice.

4750 Willow Road, Pleasanton, CA 94588 (T) 1.800.POLYCOM (765.9266) for North America only.

For North America, Latin America and Caribbean (T) +1.925.924.6000, (F) +1.925.924.6100

270 Bath Road, Slough, Berkshire SL1 4DX, (T) +44 (0)1753 723000, (F) +44 (0)1753 723010

Polycom Hong Kong Ltd., Rm 1101 MassMutual Tower, 38 Gloucester Road, Wanchai, Hong Kong, (T) +852.2861.3113, (F)+852.2866.8028

Rev. 08/04