Polycom SoundStation Duo is a professional conference phone that features both IP and analog connections. SoundStation Duo lets you connect the conference phone to the PSTN now then migrate to a SIP-based VoIP system later. Or you could set it up to use the analog connection as a failover in case of network failure or disruption.
For the full range of both analogue and SiP Polycom conference phones, please click here
Polycom SoundStation Duo features HD Voice technology, providing incredible wideband audio with acoustic echo cancellation and background noise reduction. Deploy this conference phone in your small or midsize room: it has a 360° microphone pickup range of 10ft using a three cardioid microphone array. Let more participants join the conversation with expansion microphones (not included).
Polycom SoundStation Duo Key Features
- Support for both analog and IP telephony platforms
- Interoperable with leading SIP-based PBX and softswitch platforms
- 24×7 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 dialling 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.
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
