Polycom SoundStation IP 6000 Conference VoIP Phone ( 2200-15600-001) | Like New
SKU: 2200-15600-001-LN
This Phone Supports Webex Calling
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Soundstation IP 6000 Conference Phone
The SoundStation IP 6000 is an advanced IP conference phone that delivers superior performance for small to midsize conference rooms. With advanced features, broad SIP interoperability and remarkable voice quality, the SoundStation IP 6000 is a price/performance breakthrough for SIP-enabled IP environments.
Features
- Polycom HD Voice - Unparalleled clarity to make your conference calls more efficient and productive
- 12-foot microphone pickup - Combined with Automatic Gain Control for performance far beyond older SoundStation IP conference phones
- Ability to increase coverage - Optional extended microphones expand coverage to 20 x 30 feet for larger conference rooms
- Backlit display provides ease of use - 248 x 68 pixel resolution LCD makes menu navigation easy; Support for foreign languages including Asian characters
- Industry-leading SIP software - Leveraging the most advanced SIP endpoint in the industry, with advanced call handling, security, and provisioning features
- Robust interoperability - Compatible with a broad array of SIP call platforms
- Network interface - Ethernet10/100BaseT; supports Static IP and DHCP
- Codec support - G.711, G.729a, G.722, G.722.1, G.722.1C, Siren 14
- Protocol support - IETF SIP (RFC3261 and companion RFCs)
- Management support - Web-base configuration (HTTP), FTP/TFTP/HTTP/HTTPS support for downloading of software updates and configuration files
- Power - Can be powered via 802.3af Power over Ethernet (PoE) or local PoE injector
Specifications
Technical Information
| Manufacturer | Polycom |
| Manufacturer Part Number | 2200-15600-001 |
| Product Name | Polycom SoundStation IP 6000 Conference VoIP Phone |
| Product Type | Conference Phone |
| Power | IEEE 802.3af Power over Ethernet (built in) |
| Optional external universal AC power supply: 100-240V, 0.4A, 48V/19W | |
| 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: 3 | |
| On-hook/Off-hook, redial, mute, | |
| volume up/down | |
| Audio Features | Loudspeaker |
| Frequency: 220-14,000 Hz | |
| Volume: Adjustable to 86 dB at 1/2 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, G.722.1 | |
| G.722.1C | |
| Siren 14 | |
| 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 Simplified Chinese, Danish, Dutch, English (Canada / US / UK), French, German, Italian, Japanese, Korean, Norwegian, Polish, Portuguese, Russian, Slovenian, Spanish, Swedish | |
| Network and Provisioning | Ethernet 10/100 Base-T |
| 2.5mm connection port | |
| EX mic ports: Two RJ-9 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 |