Condition: New

Cisco ATA-186-I1 VoIP Adapter
Supported Protocols: SCCP, SIP, MGCP, H.323
Protocol Installed: SCCP
Includes Free Power Supply (ATA186-PWR-CUBE)


Product Description

The Cisco ATA186 VoIP Adapter is a handset-to-Ethernet adapter that turns traditional telephone devices into IP devices. Customers can take advantage of the many new and exciting IP telephony applications by connecting their analog devices to Cisco ATAs. The Cisco VoIP Adapter products are standards-based communication devices that deliver true, next generation voice-over-IP (VoIP) terminations to businesses and residences worldwide.


Protects Legacy Telephone Investment

The Cisco ATA186 supports two voice ports, each with its own independent telephone number, and a single 10BaseT Ethernet port. This adapter can make use of existing Ethernet LANs, in addition to broadband pipes such as digital subscriber line (DSL), fixed wireless, and cable modem deployments.


Cost Effective

The Cisco ATA 186 helps customers turn their analog phone devices into IP devices cost-effectively and is the preferred solution to address the needs of customers who connect to either enterprise networks, small-office environments, or the emerging VoIP managed voice services and local services market. Enterprise customers are using the Cisco ATA 186 to connect analog phones and FAX machines to their VoIP network. Service providers are taking advantage of emerging telephony applications and the ease of deploying second-line services using the Cisco ATA 186.


Features

• Two voice ports support legacy (analog) touch tone telephones
• RJ 45 connection to 10BaseT Ethernet hub or switch
• Auto-provisioning with Trivial File Transfer Protocol (TFTP) provisioning servers
• Automatic assignment of IP address, network route IP, and subnet mask via Dynamic Host Configuration Protocol (DHCP)
• Web configuration through built-in Web server
• Touch-tone telephone keypad configuration with voice prompt
• Administration password to protect configuration and access
• Remote upgrades through network
• Advanced pre-processing to optimize full-duplex voice compression
• High performance line-echo cancellation eliminates noise and echo
• Voice activity detection (VAD) and comfort noise generation (CNG) save bandwidth by delivering voice, not silence
• Dynamic network monitoring to reduce jitter artifacts such a packet loss
• Session Initiation Protocol (SIP)
• Skinny Client Control Protocol (SCCP)-Cisco CallManager technology

• G.729, G.729A, G.729AB2
• G.723.1
• G.711a-law
• G.711�-law

• DHCP (RFC 2131)
• Web configuration via built-in Web server
• Touch-tone telephone keypad configuration with voice prompt
• Basic boot provisioning (RFC 1350 TFTP Profiling)
• Dial plan provisioning
• Cisco Discovery Protocol for SCCP

• RC4 encryption for TFTP configuration profiles

• DTMF tone detection and generation

• FC 2833 AVT tones for SIP, MGCP, SCCP

• Configurable for two sets of frequencies and single set of on/off cadence

• Echo canceller for each port
• 8 ms echo length
• Nonlinear echo suppression (ERL greater than 28 dB for f = 300 to 3400 Hz)
• Convergence time = 250 ms
• ERLE = 10 to 20 dB
• Double-talk detection

• Voice activity detection (VAD)
• Comfort noise generation (CNG)
• Dynamic jitter buffer (adaptive)

• G.711 fax pass-through
• G.711 fax mode