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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Getting Started
Feature Overview
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SIP Proxy Redundancy
In typical commercial IP Telephony deployments, all calls are established through
a SIP proxy server. A typical SIP proxy server can handle thousands of
subscribers. It is important that a backup server be available so that an active
server can be temporarily switched out for maintenance. The ATA supports the
use of backup SIP proxy servers (through DNS SRV) so that service disruption is
minimized.
An easy way to support proxy redundancy is to configure your DNS server with a
list of SIP proxy addresses. The ATA can be instructed to contact a SIP proxy
server in a domain named in the SIP message. The ATA consults the DNS server to
get a list of hosts in the given domain that provide SIP services. If an entry exists,
the DNS server returns an SRV record that contains a list of SIP proxy servers for
the domain, with their host names, priority, listening ports, and so on. The ATA tries
to contact the list of hosts in the order of their stated priority.
If the ATA is currently using a lower priority proxy server, it periodically probes the
higher priority proxy to see whether it is online, and switches back to the higher
priority proxy when possible. You can use the default settings or configure the
Proxy Redundancy Method in the Proxy and Registration section of the Line 1
Settings (PHONE Port) page
Other ATA Voice Features
• Silence Suppression and Comfort Noise Generation
Voice Activity Detection (VAD) with Silence Suppression is a means of
increasing the number of calls supported by the network by reducing the
average bandwidth required for a single call. VAD distinguishes between
speech and non-speech signals, and Silence Suppression removes the
natural silences that occur in a conversation. Therefore the IP bandwidth is
used only to transmit speech. Comfort Noise Generation provides
artificially-generated background white noise (sounds) to reassure callers
that their calls are still connected during the silent periods. You can enable
this feature in the Audio Configuration section of the Line 1 and Line 2
Settings (PHONE Port1 and PHONE2) page.
• Modem and Fax Pass-Through
- Modem pass-through mode can be triggered by predialing the Vertical
Service Activation Code for the Modem Line Toggle Code. You can
configure this setting in the Vertical Service Activation Codes section
of the Regional page.
- FAX pass-through mode is triggered by the detection of a CED/CNG
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tone or an NSE event.
Getting Started
Feature Overview
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- Echo canceller is automatically disabled for Modem passthrough mode.
- Echo canceller is disabled for FAX pass-through if the parameter FAX
Disable ECAN (Line 1 or 2 tab) is set to “yes” for that line (in that case
FAX pass-through is the same as Modem pass-through)
- Call waiting and silence suppression are automatically disabled for both
FAX and Modem pass-through. In addition, out-of-band DTMF
transmission is disabled during modem or fax passthrough.
• Adaptive Jitter Buffer
The ATA can buffer incoming voice packets to minimize the impact of
variable network delays. This process is known as jitter buffering. The size
of the jitter buffer adjusts to changing network conditions. The ATA has a
Network Jitter Level control setting for each line of service. The jitter level
determines how aggressively the ATA tries to shrink the jitter buffer over
time to achieve a lower overall delay. If the jitter level is higher, it shrinks
more gradually. If jitter level is lower, it shrinks more quickly. You can use the
default settings or configure this feature in the Network Settings section of
the Line 1 and Line 2 Settings (PHONE Port1 and PHONE2) page.
• Adjustable Audio Frames Per Packet
This feature allows the user to set the number of audio frames contained in
one RTP packet. Packets can be adjusted to contain from 1–10 audio
frames. Increasing the number of packets decreases the bandwidth utilized,
but it also increases delay and may affect voice quality. You can configure
this setting in the RTP Parameters section of the SIP page.
• DTMF Relay
The ATA may relay DTMF digits as out-of-band events to preserve the
fidelity of the digits. This can enhance the reliability of DTMF transmission
required by many IVR applications such as dial-up banking and airline
information. You can configure this setting in the RTP Parameters section of
the SIP page.
• Call Progress Tones
The ATA has configurable call progress tones. Call progress tones are
generated locally on the ATA so that an end user is advised of status (such
as ringback) Parameters for each type of tone (for instance a dial tone
played back to an end user) may include frequency and amplitude of each
component, and cadence information. You can keep the default settings or
configure these tones in the Call Progress Tones section of the Regional
page.
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Getting Started
Feature Overview
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• Call Progress Tone Pass Through
This feature allows the user to hear the call progress tones (such as ringing)
that are generated from the far-end network.
• Echo Cancellation
Impedance mismatch between the telephone and the IP Telephony
gateway phone port can lead to near-end echo. The ATA has a near-end
echo canceller that compensates for impedance mismatch. The ATA also
implements an echo suppressor with Comfort Noise Generator (CNG) so
that any residual echo is not noticeable. This feature is enabled by default.
You can configure this setting in the Audio Configuration of the Line 1 and
Line 2 Settings (PHONE Port1 and PHONE2) page.
• Hook Flash Events
The ATA can signal hook flash events to the proxy during a connected call.
This feature can be used to provide advanced mid-call services with thirdparty-call control.
- Depending on the features that the service provider offers using third-
party-call-control, you may need to disable Call Waiting Service, Three
Way Conference Service, or Three Way Call Service to correctly signal a
hook flash event to the softswitch. You can configure these settings in
the Supplementary Service Subscription section of the Line 1 and
Line 2 Settings (PHONE Port1 and PHONE2) page.
- You can configure the length of time allowed for detection of a hook flash
by adjusting the Hook Flash Timer parameter in the Control Timer
Values section of the SIP page.
• Configurable Dial Plan with Interdigit Timers
The ATA has three configurable interdigit timers: an initial timeout signaling
that a phone is taken off hook, a long timeout signaling the end of a dialed
string, and a short timeout, signaling that more digits are expected. For more
information, see Configuring Dial Plans, page 138.
• Polarity Control
The ATA allows the polarity to be set when a call is connected and when a
call is disconnected. This feature is required to support some pay phone
system and answering machines. You can configure these settings in the
FXS Port Polarity Configuration section of the Line 1 and Line 2 Settings
(PHONE Port1 and PHONE2) page.
• Calling Party Control
Calling Party Control (CPC) signals to the called party equipment that the
calling party has hung up during a connected call by momentarily removing
the voltage between the tip and the ring. This feature is useful for
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Getting Started
Feature Overview
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auto-answer equipment. You can configure these settings in the Control
Timer Values section of the Regional page.
• Event Logging
You can enable logging and select the relative priority of events to be
logged. The information can be sent to a Syslog Server. You can configure
the syslog and debug settings in the Miscellaneous Settings section of the
System page.
• Encryption of SIP messages using SIP over TLS
You can enable SIP over Transport Layer Security (TLS) to encrypt the SIP
messages between the service provider and the your business. SIP over
TLS relies on the widely-deployed and standardized TLS protocol to
encrypt the signaling messages. You can configure the SIP Transport
parameter in the SIP Settings section of the Line 1 and Line 2 Settings
(PHONE Port1 and PHONE2) page.
• Secure Calling using SRTP
Voice packets are encrypted by using Secure Real-Time Transport Protocol
(SRTP). This function is implemented on a standards basis (RFC4568).
Secure call service (Secure Call Serv) is enabled by default in the
Supplementary Service Subscription section of the Line 1 and Line 2
Settings (PHONE Port1 and PHONE2) page. When this service is enabled,
users can activate secure calling by pressing the star (*) key before dialing a
phone number. Alternatively, you can enable the Secure Call Setting to
encrypt all calls from a user’s phone. See the Supplementary Service
Settings section of the User 1 and User 2 page.
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