Operation Manual

Getting Started

Feature Overview

Cisco SPA100 Series Administration Guide 2

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.

ATA Voice Features

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 1

and PHONE 2) 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.