Overview
Overview of Polycom SoundStation IP 7000 SIP Conference Phone (2200-40000-001) | Like New
The Polycom SoundStation IP 7000 SIP Conference Phone (2200-40000-001) is a game-changing conference phone that provides exceptional performance and a broad feature set for SIP-based VoIP platforms. It is the most technologically advanced conference phone ever created, and it is suitable for executive offices, conference rooms, and boardrooms.
The Polycom HD Voice technology in the SoundStation IP 7000 increases productivity and reduces listener fatigue by transforming conventional conference calls into crystal-clear interactive conversations. It provides high-fidelity audio from 160 Hz to 22 kHz, capturing both the lower and higher frequencies of the human voice for conference conversations that sound as natural as if you were present..
The Polycom SoundStation IP 7000 SIP Conference Phone provides improved audio performance that considerably outperforms prior generations of conference phones for all conference calls. Only Polycom can provide a conference phone experience without compromises, from full-duplex technology that prevents irritating drop-outs to the most recent echo cancellation innovations.
The Polycom SoundStation IP 7000 SIP Conference Phone conference phone is the most adaptable and extendable conference phone ever created. Connect two units together in the conference room for greater loudness and microphone pickup, as well as numerous call control interfaces. Connect up to two extra expansion microphones to a single phone to ensure everyone in the room is in close proximity. In addition, the SoundStation IP 7000 can be linked to Polycom® HDX® group video systems for a fully integrated voice and video conferencing solution.
Polycom has combined its long history in voice conferencing and VoIP technology to create the Polycom SoundStation IP 7000 SIP Conference Phone, a pioneering new conference phone that is the clear choice for SIP-enabled applications. It uses the same SIP phone software as Polycom's award-winning Polycom® SoundPoint IP desktop phones, which are the industry's most comprehensive, dependable, and feature-rich SIP solutions, with proven interoperability with a wide range of IP PBX and hosted platforms.
Furthermore, the Polycom SoundStation IP 7000 SIP Conference Phone has a big multi-line high-resolution LCD display with a full XHTML microbrowser, transforming your conference phone into a powerful applications platform for your conference room. Advanced three-party conference functionality and LDAP corporate directory integration are included in the package.
Key Features of Polycom SoundStation IP 7000 SIP Conference Phone
Polycom HD Voice
Unparalleled clarity to make your conference calls more efficient and productive
Polycom Acoustic Clarity technology
Delivers the best conference phone experience with no compromises
Flexible configuration options
Multi-unit connectivity, expansion microphones and integration with Polycom HDX room telepresence solutions to meet the needs of many different types of rooms
Strong, robust SIP software
Leverages the most advanced SIP endpoint software in the industry, with advanced call handing, security, and provisioning features
Robust interoperability
Compatible with a broad array of SIP call platforms to maximize voice quality and feature availability while simplifying management and administration
Large high-resolution display with XHTML microbrowser
Enables new applications that make conference calling easier and more functional
Specifications
Specification for Polycom SoundStation IP 7000 SIP Conference Phone (2200-40000-001) | Like New
Manufacturer | Polycom |
Manufacturer Part Number | 2200-40000-001 |
Product Name | Polycom SoundStation IP 7000 SIP Conference Phone |
Product Type | Conference Phone |
Power | • IEEE 802.3af Power over Ethernet (built in) |
• Optional external universal AC power supply: 100-240V, 1.3A, 48V/50W | |
Display | • Size (W x H): 255 x 128 pixels |
• White LED backlight with custom intensity control | |
Keypad | • Standard 12-key keypad |
• Context-dependent soft keys: 4 | |
• On-hook/Off-hook, redial, mute, volume up/down | |
• Directional navigation wheel | |
Audio features | • Loudspeaker |
- Frequency: 160–22,000 Hz | |
- Volume: Adjustable to 88 dB at 1/2 meter peak volume | |
• Full-duplex: Type 1 compliant with IEEE 1329 full duplex standards | |
• 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, G.722.1 | |
G.722.1C | |
- Polycom® Siren™ 14 | |
- Polycom® Siren™ 22 | |
Call handling features | • Shared call/bridged line appearance |
• Busy Lamp Field (BLF) | |
• Distinctive incoming call treatment/call waiting | |
• Call timer | |
• Call transfer, hold, divert (forward), pickup | |
• Called, calling, connected party information | |
• Local three-way conferencing | |
• One-touch speed dial, redial | |
• Call waiting | |
• Remote missed call notification | |
• Automatic off-hook call placement | |
• Do not disturb function | |
Other features | • Local feature-rich GUI |
• Time and date display | |
• User-configurable contact directory and call history (missed, placed, and received) | |
• Customizable call progress tones | |
• Wave file support for call progress tones | |
• Unicode UTF-8 character support. | |
Multilingual user interface encompassing Chinese, Danish, Dutch, English (Canada/US/UK), French, German, Italian, Japanese, Korean, Norwegian, Portuguese, Russian, Spanish, Swedish | |
Network and provisioning | • Ethernet 10/100 Base-T |
• 2.5 mm connection port | |
• EX mic ports: Two Walta ports | |
• IP Address Configuration: DHCP and Static IP | |
• Time synchronization with SNTP server | |
• FTP/TFTP/HTTP/HTTPS serverbased central provisioning for mass deployments. provisioning server redundancy supported. | |
• Web portal for individual unit configuration | |
• QoS Support—IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP | |
• Network Address Translation (NAT) support—static | |
• RTCP support (RFC 1889) | |
• Event logging | |
• Local digit map | |
• Hardware diagnostics | |
• Status and statistics | |
• User selectable ringer tones | |
• Convenient volume adjustment keys | |
• Field upgradeable | |
Security | • 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 | |
Safety | • UL60950-1 |
• IEC60950-1 | |
• EN60950-1 | |
• CE Mark | |
• CSA C22.2, No. 60950-1-03 | |
• AS/NZS60950-1 | |
EMC | • FCC (47 CFR Part 15) Class A |
• ICES-003 Class A | |
• EN55022 Class A | |
• CISPR22 Class A | |
• AS/NZS CISPR22 Class A | |
• VCCI Class A | |
• EN55024 | |
• RoHS compliant | |
Protocol support | • IETF SIP (RFC 3261 and companion RFCs) |
• IEEE 802.3af Power over Ethernet version | |
AC Power version ships with | • Telephone console |
• 25 ft (7.6 m) Ethernet cable | |
• Universal power supply | |
• 7 ft (2.1 m) region-specific power cord | |
• Power insertion cable | |
• Quick Start Guide | |
• Quick User Guide | |
HDX room telepresence systems ready version ships with | • Telephone console |
• 25 ft (7.6 m) Ethernet cable | |
• 15 ft (4.6) C-link cable for connection to HDX group video systems | |
• Quick Start Guide | |
• Quick User Guide | |
Environmental conditions | • Operating temperature: 32–104° F (0–40° C) |
• Relative humidity: 20–85% (non-condensing) | |
• Storage temperature: -22–131° F (-30–55° C) | |
Country of origin | • Thailand |
Phone dimensions (L x W x H) | • 15.5 x 14.6 x 2.9 in (39.4 x 37.2 x 7.3 cm) |
Phone console weight | • 2.4 lb (1.08 kg) |