The Polycom SoundStation Duo conference phone offers flexibility for small to medium-sized rooms, enabling you to use existing analog lines and VoIP lines. Failover also allows you to still use your analog connection if your internet connection fails as a result of technical issues with your provider.
Polycom SoundStation Duo Key Features
- Support for both analog and IP telephony platforms
- Interoperable with leading SIP-based PBX and softswitch platforms
- 24x7 reliability with automatic failover from IP to analog
- Easy administration with no boot server required
- Up to 10ft (3 meter) microphone pickup range.
Best-in-class Investment Protection
The Polycom SoundStation Duo conference phone operates in analog telephony environments, however it also supports the migration to VoIP.
Mobile Compatibility
Connects to mobile phones for dialing without the need for an analog phone line. This device can also be connected to a PC or tablet to become a high-quality conference phone for Internet calling.
Polycom SoundStation Duo Conference Phone Technical Specifications
Power
- IEEE 802.3af Power over Ethernet
- Optional external universal AC power supply: 100-240V, 24V, 0.5A, 2.5mm DC plug
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: 4
- On-hook/Off-hook, conference, redial, mute, volume up/down, menu, navigation keys
Audio Features
- 3 cardioid microphones: 200-7000 Hz
- Loudspeaker frequency response: 220-7000 Hz
- 10ft (3m) microphone pickup
- Volume: Adjustable to 86 dB at 0.5 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- 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 ἀle support for call progress tones
- Unicode UTF-8 character support
- Multilingual user interface encompassing Simpliἀed 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 public switched telephone network interface
- 2.5mm connection port***
- 2 RJ9 expansion microphone ports
Network and Provisioning
- IP Address Configuration: DHCP and Static IP
- Time synchronisation 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
- 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 ἀles
- 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
- AS/ACIF S004
- ANATEL
- Telepermit
- KC
- GOST-R
- TRA
Protocol Support
- IETF SIP (RFC 3261 and companion RFCs)
SoundStation Duo ships with the following:
- Telephone Console
- 21-ft (6.4-m) combined analog and Ethernet cable with Power Injection Module
- Universal Power Supply 24V, 0.5A
- 7-ft (2.1-m) region-speciἀc power cord
- 7-ft (2.1-m) Ethernet cable
- 7-ft (2.1-m) telephony cable (RJ11)
- Quick Start Guide
Accessories
- 2 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
Country of Origin
Phone Dimensions
- 15.6 x 12.9 x 2.5 in (34.6 x 32.7 x 6.4 cm) (L x W x H)
Phone Console Weight
Box Dimensions
- 13.9 x 18 x 3.7 in (35.3 x 45.8 x 9.4 cm) (L x W x H)
Box Weight
* Also available in PSTN mode
** Due to the diversity of Caller ID standards, some features may not be available in all areas. In addition, the quality of the telephone line connection may affect Caller ID functionality. Caller ID service may require a subscription from a service provider in your area.
*** SoundStation Duo uses a cable that connects to a standard 2.5mm headset connector. If your mobile phone model or PC does not support this type of connection you will need an adapter (not included).
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