Polycom Soundstation Duo Conference Phone

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$ 933.99 Inc-Gst

1 Year Nationwide Warranty

Extended warranties available

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Polycom Soundstation Duo, Dual Mode Conference Phone

The only dual-mode conference speaker phone for small and midsize rooms.

The Polycom SoundStation Duo is a dual-mode conference phone with both analogue and VoIP capabilities. This conference speakerphone provides exceptional deployment flexibility and best-in-class investment protection. Designed for small to midsize conference rooms, SoundStation Duo is easy to set up and use.

Also, this unified communications system delivers Polycom’s legendary HD audio performance. In VoIP environments, the SoundStation Duo conference phone delivers the most robust, standards-based interoperability in the industry.

Polycom HD Voice technology. Full-duplex audio. The latest in echo cancellation and resistance to interference from mobile phones and other wireless devices. These features and more help the SoundStation Duo conference phone deliver unrivalled group-conferencing experiences without distractions.

The Polycom SoundStation Duo provides:

Best-in-class investment protection

Operates in analogue telephony environments and supports the migration to VoIP.

The broadest VoIP inter-operable

Compatible with a broad array of SIP call platforms to maximize voice quality and feature availability while simplifying management and administration.

Strong, robust SIP software

Leverages the most advanced SIP endpoint software in the industry, with advanced call handing, security, and provisioning features.]

Lower deployment and administration costs

Web-configuration tool makes setup simple and eliminates the need for a boot server.

Technical Overview

Support for analogue and IP telephony platforms
Interoperable with leading SIP-based PBX and softswitch platforms
24×7 reliability with automatic failover from IP to analogue
Easy administration with no boot server required
Up to 10-foot (3-meter) microphone pickup range.
Optional expansion microphones for even larger room coverage
Polycom HD Voice technology
2.5mm port for connection to mobile phones and PCs for Internet dialling

The obvious choice for crystal clear group audio conferencing

Large organization or small, thousands of conference rooms or just one, you have a need to bring dispersed teams, business partners, and customers together to communicate and collaborate. Conference phones from Polycom have become the de facto standard for connecting groups of people across multiple locations. With the Polycom SoundStation Duo conference phone, Polycom has taken the concepts of group productivity tool and standard office workhorse to a new level for small to midsize rooms, delivering the ultimate in deployment flexibility, ease of use, and audio quality.

Unrivalled investment protection with the broadest connection options

Whether you currently have a traditional analog connection or have already migrated to Voice over IP (VoIP) telephony, the Polycom SoundStation Duo conference phone works. In VoIP environments, the SoundStation Duo delivers the most robust standards-based inter-operable in the industry.

Lower cost of deployment and administration

Setting up the SoundStation Duo for analogue operation is as simple as plugging it in. In open SIP-based VoIP environments, a web-based tool assists with setup and facilitates online software upgrades. A large backlit display with broad multi-language support offers call information and context sensitive call functions.

Crystal-clear voice conferencing with no compromises

Backed by Polycom’s legendary audio technology, the SoundStation Duo phone delivers remarkably clear voice quality. From Polycom® HD Voice™ technology and full-duplex audio to the latest in echo cancellation and resistance to mobile phone and wireless device interference, the SoundStation Duo conference phone delivers unrivalled group conferencing experiences without distractions.

Built-in investment protection

Use in analogue or IP mode and keep it up-to-date with simple online software upgrades.

Robust interoperability

Compatible with a broad array of IP call platforms to maximize voice quality and feature availability while simplifying management and administration.

Business continuity

Auto failover from IP to analogue and failback for continuous operation in case of a network failure.

Unparalleled voice clarity

Polycom® HD Voice™ technology makes your IP conference calls more effective and productive.

Easy to deploy and administer

Web configuration tool eliminates the need for a boot server.

Superior call handing, security, and provisioning

Leveraging the most advanced IP endpoint software in the industry.

Unmatched flexibility

Connect to mobile phones and PCs for Internet dialling.

Product specifications

Power

IEEE 802.3af Power over Ethernet

External universal AC power supply: 100–240 V, 24 V, 0.5 A, 2.5 mm DC plug

Display

Size (W x H): 248 x 68 pixels

White LED backlight with custom intensity control

Keypad

Standard 12-key keypad

Context-dependent soft keys: 4

On-hook/Off-hook, conference, redial,

mute, volume up/down, menu, 5-way navigation keys

Audio features

3 cardioid microphones: 200–7000 Hz

Loudspeaker frequency response: 220–7000 Hz

10 ft (3 m) microphone pickup

Volume – Adjustable to 86 dB at 0.5 meter vpeak volume

Full-duplex – Type 1 compliant with IEEE 1329

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 –

iLBC 13.33 and 15.2kbps

SIP call handling features

Call hold*

Call transfer, divert (forward) and pickup

Distinctive incoming call treatment/call waiting

Advanced Local three-way conferencing (conference, join, split, hold, resume)

One-touch speed dial, redial*

Remote missed call notification

Automatic off-hook call placement

SIP URI dialling

Do not disturb function

Shared call/bridged line appearance

Busy Lamp Field (BLF)

Multicast Group Paging and Push-to-Talk

Other features

Automated failover (SIP to PSTN)

SIP Server Redundancy

Time and date display/call timer

User-configurable contact directory and call history (missed, placed, and received)

Corporate Directory (LDAP) support

User selectable ringer tones

Wave file support for call progress tones

Unicode UTF-8 character support

Called, connected party information

Support for multiple Caller ID standards – Bellcore Type 1 – ETSI – DTMF

Multilingual user interface encompassing

Simplified Chinese, Traditional Chinese, Danish, Dutch, English (Canada /US/UK), French, German,

Italian, Japanese, Korean, Norwegian, Polish, Portuguese, Russian, Slovenian, Spanish, Swedish

Interfaces

Ethernet 10/100 Base-T

Two-wire RJ-11 analogue PBX or PSTN interface

2.5 mm connection port

2 RJ9 ports for wired expansion microphones

Network and provisioning

IP Address Configuration – DHCP and Static IP

Time synchronization with SNTP server

FTP/TFTP/FTPS/HTTP/HTTPS server based central provisioning for mass deployments.

Provisioning server redundancy supported.

Web portal for individual unit configuration and online software upgrade

QoS Support—IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP

Telchemy VQmon support

Network Address Translation (NAT) support—static

RTCP support (RFC 1889)

Configuration import/export

Local digit map (dialling plan)

Hardware diagnostics

Status and statistics

Reset to factory settings

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 executable

IEEE 802.1x Network Access Control

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

FCC Part 68

AS/ACIF S002/S004

Telepermit

GOST-R

TRA

Protocol support

IETF SIP (RFC 3261 and companion RFCs)

SoundStation Duo ships with

Conference phone console

21 ft (6.4 m) combined analogue

Ethernet cable with power injection module

Universal power supply 24 V, 0.5 A

7 ft (2.1 m) region-specific power cord

7 ft (2.1 m) Ethernet cable

7 ft (2.1 m) telephony cable (RJ11)

Quick Start Guide

Accessories

2 x expansion microphones 200–7000 Hz

Environmental conditions

Operating temperature – 32–104° F (0–40° C)

Relative humidity – 20–85% (non-condensing)

Storage temperature: -22–131°F (-30–55°C)

Warranty

1 Year Warranty

Country of origin

2200-19000-001 – Made in China

G2200-19000-001 – Assembled in USA

Phone dimensions (L x W x H)

15.6 x 12.9 x 2.5 in (34.6 x 32.7 x 6.4 cm)

Phone console weight

1.62 lb (0.74 kg)

Box dimensions (L x W x H)

13.9 x 18 x 3.7 in (35.3 x 45.8 x 9.4 cm)

Box weight

4.56 lb (2.06 kg)