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Getting Started
Feature Overview
Cisco SPA100 Series Administration Guide 2
1
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.
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