Polycom Soundstation Duo SIP/Analogue Conference Phone

The latest in a long line of Conferencing Products by the class leading Polycom Label.

Conference units can be seen as quite an outlay, so Polycom have designed the Soundstation Duo, to give you the comfort of knowing that your investment can grow and change with your business. This new unit can plug into a standard analogue telephone socket and if you move up to using a VOIP solution it can also upgrade with you at no additional cost. Using SIP standard it will work on the majority of SIP enabled systems.

Features and Benefits

• Best-in-class investment protection – The Polycom SoundStation Duo conference phone operates in analog telephony environments and supports the migration to VoIP
• Broadest VoIP interoperability – 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 – Leveraging the most advanced SIP endpoint software in the industry, with advanced call handing, security, and provisioning features
• Lower cost of deployment and administrative – Web configuration tool makes setup simple and eliminates the need for a boot server
• Applications Port – Connects to mobile phones for dialing from rooms without an analog phone line, and connects to a PC or tablet to become a high-quality conference phone for Internet calling
• Resists interference from mobile phones and other wireless devices while delivering clear voice conferencing with no distractions

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 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

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

• 1 year