Polycom SoundStation Duo - Dual Mode (Analog or IP mode)

 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.

Benefits

• Built-in investment protection – Use in analog 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 analog 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 dialing

Power

• IEEE 802.3af Power over Ethernet
• 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, 5-way 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
• 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
• 2.5mm 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

Protocol Support

• IETF SIP (RFC 3261 and companion RFCs)

* 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.