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 Product >> Cisco >> ALL
 
Product_Id:
72410344716
ProductName:
CP-6821-3PCC-K9
Specification:
Product Notes:
Product Category:
Cisco
 
   Product Description

Cisco CP-6821-3PCC-K9 Full English Product Description

Listing Title

Cisco CP-6821-3PCC-K9 IP Phone 6821 Multiplatform (3PCC) 2-Line Business VoIP Desk Phone, 2.5-inch 240×120 Backlit Grayscale LCD Display, Dual 10/100BASE-TX Switched Ethernet Ports with PC Pass-Through, IEEE 802.3af Class 2 PoE Supported, Full-Duplex Wideband HD Audio Speakerphone, RJ-9 Wideband Headset Port, Third-Party Call Control (3PCC) Open SIP Firmware, SRTP/TLS Encrypted Secure Voice, XML Application Support, Desktop & Wall Mountable, Ideal for Small-to-Mid Office Staff, Remote Workers, Retail Back Offices and General Business Multi-Line Voice Deploymentsshoprefres...

Full Commercial Overview

The Cisco CP-6821-3PCC-K9 is a compact entry-level multi-platform IP desk phone belonging to the Cisco 6800 Series MPP (Multiplatform Phone) VoIP endpoint family. The suffix3PCCstands for Third-Party Call Control, meaning this model runs open-standard SIP firmware and is not locked to Cisco CUCM; it fully interoperates with third-party hosted PBX, Asterisk, Broadsoft, Webex Calling and all industry-standard SIP telephony platformsshoprefres....

Designed for cost-conscious small and medium businesses, remote office workers and light-to-medium call-volume staff, the 6821 supports up to two independent SIP registrations with multi-call-per-line appearance functionality, allowing users to handle multiple concurrent calls on each line button. It delivers enterprise-grade HD wideband audio, integrated Ethernet switching for shared PC wall jacks, native PoE power delivery and robust secure voice encryption, all within a space-saving ergonomic chassis.

All device provisioning, dial plan configuration, unattended bulk firmware upgrades and remote diagnostics are supported via TFTP, HTTP and HTTPS secure auto-provisioning. Users can complete basic local configuration adjustments directly through the backlit LCD graphical menu navigation without requiring a PC web portal. The phone also supports customizable XML enterprise applications for directory access, corporate news and custom service workflowsCisco.

Hardware, Display & Audio Architecture

1. Two Independent Programmable SIP Line Buttons

Two physical illuminated line keys on the left side of the display support up to two unique SIP registrations. Each line key can handle multiple simultaneous call appearances (call waiting, hold, shared line monitoring) with color-coded LED status indicators (green for idle, red for active/held calls, flashing red for incoming ringing).

2. Graphical LCD Display & Physical Control Layout

  • Display Panel: 2.5-inch (6.35cm) backlit grayscale LCD screen with 240 × 120 pixel resolution. The screen renders caller ID name/number, real-time call duration timestamps, local call history logs, system menu navigation prompts, network time/date, XML application content and status icons for active headset, mute and DND modes.
  • Hard Key Layout & Navigation: Standard 12-key tactile numeric dial pad; dedicated fixed-function illuminated hard keys: Mute, Headset, Speakerphone, Volume Toggle, Navigation Cluster (four-way directional rocker with central select button), Service, Directory, Hold, Transfer, Conference and Exit softkey buttons below the LCD for context-sensitive call functions.
  • LED Status Indicators: Backlight status LEDs on all line and function keys; dedicated vertical red Message Waiting Indicator (MWI) LED on the top-right corner of the chassis to notify users of unheard voicemail messages.

3. Complete Wideband HD Audio Subsystem

  • Ergonomic wired analog handset with wideband HD voice support, connected via a standard RJ-9 modular port.
  • Built-in full-duplex hands-free speakerphone with acoustic echo cancellation, supporting simultaneous talk-and-listen conference-style calling without audio cut-off; DTMF tone masking is enabled automatically during speakerphone operation for privacy.
  • Dedicated RJ-9 wideband headset audio port compatible with Cisco 500 Series wired headsets and industry-standard office headsets.
  • Supported Voice Codecs: G.711 A-law / µ-law, G.726, G.729A/B/AB, G.722 HD wideband voice, G.722.2, iSAC, ILBC, OPUS wideband audio codecsCisco.
  • Audio Optimization Technologies: Adaptive dynamic jitter buffer, Voice Activity Detection (VAD) silence suppression, Comfort Noise Generation (CNG); DTMF signaling transmitted via in-band audio or RFC 2833 out-of-band SIP INFO signaling.

4. Chassis & Mounting Specifications

Compact matte black ABS plastic desktop chassis with a removable single-position foldable rear foot stand for stable desk or countertop placement. The foot stand can be detached entirely to enable wall mounting, with pre-drilled mounting holes located on the rear panel of the phone.

  • Physical Dimensions (excluding foot stand, handset and cabling): 207 × 206 × 33 mm (8.2 × 8.1 × 1.3 inches)
  • Unit Weight (including foot stand, excluding handset): 512 g (18 oz)

Network, Power & Hardware Interfaces

1. Dual Switched 10/100BASE-TX Fast Ethernet RJ45 Ports

Two auto-sensing Fast Ethernet ports with an integrated mini network switch, a core deployment advantage for office workstations with attached desktop PCs:

  • Primary WAN Port: Connects the IP phone directly to corporate LAN switches, VoIP gateways or SIP PBX appliances for SIP control signaling and RTP voice media traffic.
  • Secondary PC Pass-Through Port: Enables direct connection of desktop PCs, laptops or retail POS terminals to the corporate network through the phone, eliminating the requirement for extra wall-mounted network jacks and cutting office cabling infrastructure costsCisco.

Supported networking standards: Full IPv4 RFC compliance suite, DHCP client for automatic IP addressing, DNS SRV record lookup, ARP, ICMP, TCP/UDP, RTP/RTCP voice media quality reporting, Layer 2 802.1p QoS traffic marking, voice VLAN tagging, STUN NAT traversal for remote/home office NAT network environments, SNTP network time synchronization, CDP and LLDP network device discovery protocols.

2. Dual Power Supply Options

  1. Native IEEE 802.3af Class 2 PoE Support (Primary Deployment Method): The phone draws electrical power directly from PoE-enabled network switches via Ethernet cabling, eliminating separate AC power adapters for standard office workstation deployments.
  2. Auxiliary AC Power Supply Backup: Optional universal AC-to-DC power adapter (Cisco country-specific power cube series, sold separately), rated for 100–240 VAC 50/60Hz input, output 5V DC for phone operation in non-PoE network environmentsCisco.

3. Additional Physical Interfaces

  • RJ-9 modular wired handset connector
  • Dedicated RJ-9 wideband headset audio jack
  • Recessed pinhole password-protected factory reset button for restoring default device configurations.

Core Business Telephony Feature Set

All calling functions can be configured via auto-provisioning templates, on-device LCD menu or triggered via standard SIP star feature access codes:

  1. Two-Line Multi-Call Handling: Multi-call-per-line appearance support, per-line call waiting, call hold, music-on-hold, attended & blind call transfer, local three-way ad-hoc audio conferencing, call swap between active lines, automatic last number redial, on-hook dialing, global Do Not Disturb (DND) mode to suppress incoming ring alerts, call park and unpark functionality, shared line appearance monitoring support.
  2. Caller ID & Personalization: Caller ID name and number display on LCD, outbound caller ID blocking, anonymous call rejection, customizable distinctive ringtones per line, programmable speed dial assignments and local stored personal phone directory with BLF (Busy Lamp Field) status monitoring for extensions.
  3. Utility & Group Calling Features: One-touch intercom calling with auto-answer speakerphone mode, selective and group call pickup, call back on busy, system broadcast paging support.
  4. Local Call Log Storage: Onboard storage for dialed, received and missed call records with timestamps and call duration data, fully viewable on the phone’s LCD screen.
  5. Voicemail Integration: Native MWI voicemail notification via SIP SUBSCRIBE/NOTIFY messaging, dedicated one-touch Messages hard key for instant voicemail retrieval.

Security & Remote Management Capabilities

  1. Voice & Signaling Encryption: SRTP secure AES encryption for VoIP voice media streams, TLS encrypted SIP control signaling to prevent network call eavesdropping and malicious signaling interception.
  2. Multi-Tier Device Access Security: Separate administrator and end-user permission levels protected by unique passwords; password-locked factory reset function to block unauthorized full configuration erasure.
  3. Automated Provisioning & System Monitoring: Secure unattended remote firmware upgrades without interrupting active voice calls, centralized syslog event logging, built-in network diagnostic tools for IT administrators and small business telecom service providers; customizable XML enterprise application support for corporate directories and internal services.

Environmental & Regulatory Specifications

  • Regulatory Compliance Certifications: FCC Part 15 Class B electromagnetic interference compliance, UL / C-UL electrical safety certification, CE marking for European Union market compliance, A-Tick Australian telecommunications certification, HAC hearing aid compatibility certification.
  • Operating Environment: Indoor use only Operating Temperature: 0°C ~ +40°C (32°F ~ 104°F) Storage Temperature: -20°C ~ +70°C (-13°F ~ 158°F) Operating Humidity: 10% – 90% non-condensing Storage Humidity: 10% – 95% non-condensing
  • Memory Architecture: Onboard flash memory for persistent firmware and configuration storage, dedicated SDRAM runtime memory for real-time multi-line call processing and LCD menu operation.
  • Lifecycle Status: Current active Cisco MPP multi-platform VoIP hardware, supported with regular firmware updates, not end-of-sale.
  • Warranty: Standard one-year limited Cisco hardware warranty for commercial business end-user deployments.

Key Specification Bullet Points

  • Model Number: Cisco CP-6821-3PCC-K9
  • Device Type: Multiplatform 2-Line Open SIP VoIP Desk IP Phone (3PCC Third-Party Call Control Firmware)
  • Supported Telephony Protocols: Industry Standard SIP v2 (RFC 3261 Suite), MPP Multiplatform Firmware for Third-Party PBX Compatibility
  • Supported Voice Codecs: G.711 A/µ, G.726, G.729A/B/AB, G.722 HD Wideband Voice, G.722.2, iSAC, ILBC, OPUS
  • Display: 2.5-inch backlit grayscale LCD screen, 240 × 120 pixel resolution, XML application support
  • Physical Buttons & Indicators: 2 programmable illuminated line keys, 12-key numeric dial pad, dedicated illuminated hard keys for Mute, Headset, Speakerphone, Volume, Navigation Cluster, Directory, Hold, Transfer, Conference; standalone red vertical MWI voicemail waiting LED
  • Physical Audio Interfaces: RJ-9 handset port, dedicated RJ-9 wideband headset jack, built-in full-duplex echo-canceling HD speakerphone, HAC hearing-aid compatible handset
  • Ethernet Interfaces: Dual auto-sensing 10/100BASE-TX RJ45 switched LAN ports (WAN uplink + PC pass-through secondary port)
  • Power: Native IEEE 802.3af Class 2 PoE support; Optional separate 5V DC AC power adapter sold individually
  • Mounting Method: Removable foldable desktop foot stand as standard; wall-mount capable via rear chassis mounting holes
  • Supported QoS & Networking: 802.1p Layer 2 QoS marking, voice VLAN tagging, DHCP client, STUN NAT traversal, SNTP time sync, CDP/LLDP network discovery
  • Security Features: TLS encrypted SIP signaling, SRTP AES encrypted voice media, dual-level admin/user password lock, HTTPS web management, password-protected factory reset
  • Core Call Features: 2-line multi-call-per-line concurrent call support, call waiting, hold, attended/blind transfer, three-way conference, BLF extension monitoring, caller ID blocking, anonymous call rejection, intercom, group call pickup, speed dial, LCD-accessible local call logs, DND silent mode, voicemail MWI notification, call park/unpark, shared line appearance
  • Management Tools: Embedded web admin interface, local LCD menu configuration, TFTP/HTTP/HTTPS auto remote provisioning, syslog event logging, unattended remote in-service firmware upgrades, customizable XML enterprise application support
  • Deployment Scenarios: Small and medium business office staff desks, retail back-office workstations, remote home office telephony setups, low-to-medium call-volume administrative staff, general business multi-line voice deployments using third-party SIP PBX platforms
  • Lifecycle Status: Active production multi-platform MPP IP phone, fully supported by Cisco firmware development
  • Warranty: One-year limited Cisco hardware warranty

Model Quick Reference

  • CP-6821-3PCC-K9 is the entry-level 2-line multi-platform SIP phone of Cisco 6800 MPP Series; the 3PCC suffix denotes open third-party SIP call control firmware (distinct from CUCM-locked SCCP versions).
  • Core positioning: Cost-effective compact VoIP endpoint built exclusively for open SIP PBX environments (Asterisk, Broadsoft, Webex Calling, hosted SIP services) without reliance on Cisco Unified Communications Manager.
  • Primary deployment advantage: Dual switched Fast Ethernet ports eliminate redundant wall network jacks for combined PC + phone workstation setups, while native IEEE 802.3af PoE reduces separate power adapter cabling costs for mass office rollouts.
  • Series internal comparison: 6821 = 2-line compact entry model; 6841 = 4-line mid-tier; 6851 = 8-line high-capacity; all support MPP 3PCC open SIP firmware variants.

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