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   Cisco ATA 186 Analog Telephone Adaptor

               --- Brand New with User License and 3-month Warranty





$159.00
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The Cisco ATA 186 Analog Telephone Adaptor is a handset-to-Ethernet adaptor 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 Analog Telephone Adaptor 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 ATA 186 supports two voice ports, each with its own independent telephone number, and a single 10/100BaseT Ethernet port. This adaptor 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.


Figure 1  Cisco ATA 186—Endpoint for an end-to-end broadband system




The Cisco ATA 186 allows you to connect analog telephones and faxes to an IP telephony network.


Table 1: Features and Benefits
Features Benefits
  • Two voice ports support legacy (analog) touch-tone telephones

  • RJ-45 connection to 10/100Base-T Ethernet hub or switch

Connects legacy telephones to IP-based networks
  • 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

Flexible configuration and provisioning options
  • 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

Clear, natural-sounding voice quality
  • H.323

  • Session Initiation Protocol (SIP)

  • Media Gateway Control Protocol (MGCP)

  • Skinny Client Control Protocol (SCCP)—Cisco CallManager technology

Supports multiple protocols for interoperability and deployment flexibility
  • Fits in most environments

Small form-factor design
  • Passwords displayed as asterisks instead of readable text.

Enhanced security
  • Network status page

Track packet input, output and errors


System Requirements:
A

Regular analog, touch-tone telephones
B

10/100Base-T category-3 cable or better (access to an IP network)
C

Power for AC/DC power adaptor




Software 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 codecs1

  • 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)

Fax2

  • G.711 fax pass-through

  • G.711 fax mode




1In simultaneous dual-port operation, the second port is limited to G.711 when using G.729.
2Success of fax transmissions up to 14.4 kbps depends on network conditions and fax modem/fax machine tolerance to those conditions. Network must have reasonably low network jitter, network delay, and packet loss rate.


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