Product Overview

The Cisco ATA186 Analog Telephone Adaptor is a handset-to-Ethernet adaptor that interfaces regular analog phones with IP-based telephony networks. The Cisco ATA 186 is installed at the subscriber`s premises and supports two voice ports, each with its own independent phone number. This adaptor takes advantage of broadband pipes being deployed through digital subscriber line (xDSL), fixed wireless, cable modems, and other Ethernet connections.

The Cisco ATA 186 is the ideal solution for service providers deploying IP telephony services in the residential market while taking advantage of the installed base of handsets. By deploying IP-based telephones as a second-line, service providers can now offer additional revenue-generating services for emerging telephony applications in their residential services portfolio. Service providers can also realize a rapid return on investment (ROI) by utilizing their existing networks and move to converged network architectures. Thus, saving capital costs along with operational and administrative costs.

Key Features and Benefits

• Interfaces legacy telephones to IP-based networks
  Two voice ports support legacy (analog) touch-tone telephones
  RJ-45 connection to 10/100Base-T Ethernet hub/switch
  
• Flexible configuration and provisioning options
  Auto-provisioning with 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
  Voice prompt configuration via touch-tone telephone keypad (IVR menu)
  Administration password to protect configuration and access
  Remote upgrades through network
  
• Clear, natural-sounding voice quality
  Advanced pre-processing to optimize full-duplex voice compression
  High performance line-echo cancellation eliminates noise and feedback
  Voice activity detection (VAD) saves bandwidth by delivering voice, not silence
  Regular telephone call experience with comfort noise generation (CNG) and virtual dial-tone
  Dynamic network monitoring to reduce jitter artifacts
  
• Supports standard protocols for interoperability and deployment flexibility
  H.323
  SIP
• Small form-factor design to fit in all environments

Specifications

Voice-over-IP (VoIP) protocols
• H.323 v2
• H.323 v4
• SIP (RFC 2543 bis)
• MGCP 1.0 (RFC 2705)
• MGCP 1.0/network-based call signaling (NCS) 1.0 Profile
• MGCP 0.1
• SCCP
Voice codecs
• G.729, G.729A, G.729AB2
• G.723.1
• G.711a-law
• G.711µ-law
Provisioning and configuration
• 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
Security
• H.235 for H.323
• RC4 encryption for TFTP configuration profiles
Dual-tone multi-frequency (DTMF)
• DTMF tone detection and generation
Out-of-band DTMF
• H.245 out-of-band DTMF for H.323
• RFC 2833 AVT tones for SIP, MGCP, SCCP
Call progress tones
• Configurable for two sets of frequencies and single set of on/off cadence
Line-echo cancellation
• 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 features
• Voice activity detection (VAD)
• Comfort noise generation (CNG)
• Dynamic jitter buffer (adaptive)
Fax
• G.711 fax pass-through
• G.711 fax mode