The Cisco Unified IP Conference Phone 8831 enhances people-centric communications, combining superior high-definition (HD) audio performance and 360-degree coverage for all sizes of conference rooms and executive offices. It provides an audiophile sound experience with a full-duplex two-way wideband (G.722) audio hands-free speaker.
The IP Conference Phone 8831 is a simple, scalable solution that meets the challenges of the most diverse rooms. It provides flexible deployment options and expansion by using extension microphones that can be wired or wireless (Digital Equipment Cordless Telephone) with a daisy-chain configuration of two units.
The IP Conference Phone 8831 has an industrial design with enhanced ergonomics that puts the user first. It offers a detached control panel so that the display may be easily viewed without having to move the entire unit. It also provides easy view of device mute status from all sides. Supported on Cisco Unified Communications Manager and Business Edition systems, the IP Conference Phone 8831 delivers a more productive, acoustically pleasing, and secure communications experience across sites and participants.
The Cisco Unified IP Conference Phone 8831 identifies incoming messages and categorizes them for users on the screen. This feature allows you to quickly and effectively return calls using direct dial-back capability. The corporate directory integrates with the Lightweight Directory Access Protocol Version 3 (LDAPv3) standard directory.
The Settings feature key allows you to adjust display contrast, select background images (if available), and select from a large number of unique ringer sounds through the User Preference menu. Network configuration preferences also can be set up (usually by the system administrator). Configuration can be either automatically or manually set up for Dynamic Host Control Protocol (DHCP), Trivial File Transfer Protocol (TFTP), Cisco Unified Communications Manager, and backup Cisco Unified Communications Manager instances.
The online help feature gives you information about the phone keys, buttons, and features. The pixel display allows for more flexible feature navigation and significantly expands the information viewed when using features such as services, information, messages, and directory. For example, the directory button can show local and server-based directory information.
Hold, mute, and redial keys
The Mute key is a fixed key. When it is active, the LED flashes on the Cisco Unified IP Conference Phone 8831 as well as on the optional wired and wireless microphone kits. Hold and Redial are provided as soft keys associated with the screen, and are always at the same position for easy access.
The conference station has a large high-resolution, graphical 3.5-inch backlit display (396 x 162 pixels).
The Cisco Unified IP Conference Phone 8831 offers full-duplex high-quality wideband speakerphone technology. Included are Automatic Gain Control, comfort-noise generation, silence suppression and voice activity detection, Echo Suppression, and dynamic noise reduction, which reduce noise by up to 9 dB from constant noise sources such as fans or heating, ventilation, and air conditioning (HVAC) systems. A two-way high-fidelity loud speaker system provides superior speech clarity versus traditional conferencing systems.
The convenient volume control buttons on Cisco Unified IP Conference Phone 8831 provide for easy, decibel-level adjustments for the speakerphone and ringer.
Quality-of-service (QoS) optionsProduct Highlights
The Cisco Unified IP Conference Phone supports DHCP and 802.1Q/p standards. The conference station can also be configured with an 801.1Q VLAN header containing the VLAN ID overrides configured by the Admin VLAN ID.
- Superior wideband acoustics with the first two-element speaker in a conference phone; this feature allows the phone to capture the full voice spectrum without having to compromise with a single-element speaker
- Expanded room coverage with support for daisy chaining two units
- Support for DECT wireless extension microphones
- Session Initiation Protocol (SIP) signaling
- Device authentication and signaling encryption using Transport Layer Security (TLS) with Advanced Encryption Standard 128 (AES-128)
- Media encryption using Secure Real-Time Transport Protocol (SRTP) with AES-128