Polycom SoundStation Duo IP & Analog Conference Phone has Broad Connection Options

• Manufacture Discontinued 01/22 – no direct replacement
• Use in analog or IP mode and keep it up-to-date with simple online software upgrades
• Compatible with a broad array of IP call platforms
• Circuit switched telephony support and migration to SIP-based platforms
• Broadest interoperability with leading SIP-based IP PBX and Softswitch platforms
• 24/7 reliability with automatic failover from SIP to circuit switched
• Lower cost of deployment and administration
• Outstanding performance at an affordable price
• Legendary audio conferencing experience and performance
• Crystal clear group audio conferencing 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
• Auto failover from IP to analog and failback for continuous operation in case of a network failure
• Unparalleled voice clarity via Polycom HD Voice technology
• Easy to deploy and administer with a web configuration tool that eliminates the need for a boot server
• Superior call handing, security, and provisioning, the SSDUO leverages the most advanced IP endpoint software in the industry
• Connect to mobile phones and PCs for internet dialing
• The phone’s large backlit display offers easily visible call information, telephone functions, and broad multi-language support
• IEEE 802.3af Power over Ethernet or 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
• Standard 12-key keypad, 4 context-dependent soft keys, on-hook/off-hook, conference, redial, mute, volume up/down, menu, navigation keys
• 3 cardioid microphones: 200-7000 Hz, loudspeaker frequency response: 220-7000 Hz, 10ft (3m) microphone pickup
• Adjust volume 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
• SIP call handling features: call hold, call transfer, divert (forward) and pickup, distinctive incoming call treatment/call waiting, advanced local three-way conferencing, one-touch speed dial, redial remote missed call notification, and automatic off-hook call placement
• 1 year manufacture warranty