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.
Unrivaled 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 interoperability in the industry.
Lower cost of deployment and administration
Note:
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
peak 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 dialing
• 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
• 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
• Called, connected party information
• Support for multiple Caller ID standards**
- Bellcore Type 1
- ETSI
- DTMF
Interfaces
• Ethernet 10/100 Base-T
• Two-wire RJ-11 analog 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 serverbased
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 (dialing 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 executables
• 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
• KC
• 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 analog
and 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
Country of origin
• 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)