Polycom Soundstation Duo

Polycom Soundstation Duo

SKU: 03-0050

Polycom Soundstation Duo

The Polycom SoundStation Duo is an extremely flexible dual mode conference phone that works in both analog and IP telephony environments and includes all the outstanding audio capabilities you’ve come to expect from Polycom’s industry-leading voice conferencing solutions. In addition, SoundStation Duo provides the following key benefits:

Best-in-class investment protection

  • Circuit switched telephony support and migration to SIP-based platforms
  • Broadest interoperability with leading SIP-based IP PBX and Softswitch platforms
  • 24X7 reliability with automatic failover from SIP to circuit switched

Lower cost of deployment and administration

  • Easy administration with no boot server required
  • Simple web-based configuration tool
  • Outstanding performance at an affordable price

Legendary audio conferencing experience and performance

  • Clearer conversations with HD voice in IP mode
  • Exceptional room coverage and mic pickup
  • Expandable coverage area with optional expansion microphones
  • Easy to use with proven UI and industrial design
  • Tech Specs

    Polycom Soundstation Duo Specifications

    • IEEE 802.3af Power over Ethernet
    • Optional external universal AC power supply: 100-240V, 24V, 0.5A, 2.5mm DC plug
    • Size (pixels): 248 x 68 (W x H)
    • White LED backlight with custom intensity control
    • 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 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
    • 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 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
    • 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
    • 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
    • CE Mark
    • EN60950-1
    • IEC60950-1
    • UL60950-1
    • CAN/CSA C22.2 No.60950-1-03
    • AS/NZS60950-1
    • RoHS Compliant
    • 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
    • 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-specific power cord
    • 7-ft (2.1-m) Ethernet cable
    • 7-ft (2.1-m) telephony cable (RJ11)
    • Quick Start Guide
    • 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)