Cisco SPA100 Series Administration Manual

Getting Started

Thank you for choosing the Cisco SPA100 Series Phone Adapters. This chapter
provides more information about the features of the product and the web-based
configuration utility.

Feature Overview

Cisco SPA100 Series Analog Telephone Adapters (ATAs) provide your standard
analog phones with access to Internet phone services through two standard
telephone RJ-11 phone ports. The ATA connects to the Internet through a
broadband (DSL or cable) modem or router. The ATA can be used with an on-site
call-control system or an Internet-based call-control system.

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The ATA is an intelligent low-density Voice over IP (VoIP) gateway that enables
carrier-class residential and business IP Telephony services delivered over
broadband or high-speed Internet connections. An ATA maintains the state of each
call it terminates and reacts appropriately to user input events (such as on/off hook
or hook flash). The ATAs use the Session Initiation Protocol (SIP) open standard so
there is little or no involvement by a “middle-man” server or media gateway
controller. SIP allows inter-operation with all ITSPs that support SIP.

The system supports four simultaneous calls, including “active” calls and “on-hold”
calls. A phone can handle one on-hold call and one active call simultaneously.

Understanding Voice Service Operations

The ATA allows calls to be made by using SIP-based Voice-over-IP (VoIP) services
and traditional telephone Public Switched Telephone Network (PSTN) services.
Calls can be placed and received by using an analog phone or fax machine.

The ATA maintains the state of each call and makes the proper reaction to user
input events (such as on/off hook or hook flash). Because the ATA uses the Session
Initiation Protocol (SIP), it is compatible with most Internet Telephony Service
Provider (ITSP) offerings.

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Getting Started

Feature Overview

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ATA Voice Fe ature s

The ATA can be custom provisioned within a wide range of configuration
parameters. The following sections describe the factors that contribute to voice
quality:

• Supported Codecs

• SIP Proxy Redundancy

• Other ATA Voice Features

Supported Codecs

The ATA supports the codecs listed below. You can use the default settings or
configure the codec settings in the Audio Configuration section of the Line 1 and

Line 2 Settings (PHONE Port1 and PHONE2) page.

Codec Description

G.711 (A-law and mu-law) Very low complexity codecs that support

uncompressed 64 kbps digitized voice
transmissions at one through ten 5 ms voice frames
per packet. These codecs provide the highest
narrow-band voice quality and uses the most
bandwidth of any of the available codecs.

G.726-32 Low complexity codec that supports compressed

32 kbps digitized voice transmission at one through
ten 10 ms voice frames per packet. This codec
provides high voice quality.

G.729a ITU G.729 voice coding algorithm used to

compress digitized speech. G.729a is a reduced
complexity version of G.729 requiring about half
the processing power of G.729. The G.729 and
G.729a bit streams are compatible and
interoperable, but not identical.

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