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Page 16 / 186 Scroll up to view Page 11 - 15
1-2
Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 1
Introducing Linksys Analog Telephone Adapters
Overview
Figure 1-1
illustrates how the different ATAs provide voice connectivity in a VoIP network, including
the SPA3102, which acts as a SIP-PSTN gateway. As shown, the following devices also provide
QoS-enabled IP routers in addition to ports for connecting analog telephone devices:
•
WRP400
•
RTP300
•
WRTP54G
•
WRTP54GP2
Figure 1-1
Linksys ATAs in the VoIP Network and PSTN
The AG310 and SPA3102 provide full PSTN connectivity in addition to a single FXS port. In addition,
the AG310 provides an ADSL2+ gateway.
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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 1
Introducing Linksys Analog Telephone Adapters
Ensuring Voice Quality
Each Linksys 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. Linksys ATAs maintain the states of all the calls it terminates and makes the proper
reaction to user input events (such as on/off hook or hook flash). Because the ATAs use the SIP standard,
there is little or no involvement by a “middle-man” server or media gateway controller.
The response of a Linksys ATA includes playing a dial tone, collecting DTMF digits and comparing
them against a dial plan, or terminating the call.
Note
The information contained in this guide is not a warranty from Linksys, a division of Cisco Systems, Inc.
Customers planning to use Linksys ATAs in a VoIP service deployment are advised to test all
functionality they plan to support before putting the ATA in service.
By implementing Linksys ATAs with the SIP protocol, intelligent endpoints at the edges of a network
perform the bulk of the call processing. This allows the deployment of a large network with thousands
of subscribers without complicated, expensive servers.
The ATA is a key element in the end-to-end IP Telephony solution. It provides one or more standard
telephone RJ-11 phone ports (identical to the telephone phone wall jacks) to which the subscriber
connects standard analog telephone equipment to access phone services. The ATA connects to a wide
area IP network, such as the Internet, through a broadband (DSL or cable) modem or router.
Ensuring Voice Quality
Voice quality, as perceived by the subscribers of the IP Telephony service, should be equivalent (or
better) compared to the PSTN. Voice quality can be measured with such methods as Perceptual Speech
Quality Measurement (PSQM), with a scale of 1–5, in which lower is better; and Mean Opinion Score
(MOS), with a scale of 1–5, in which higher is better.
Table 1-2
displays speech quality metrics associated with various audio compression algorithms.
Note
The Linksys ATA supports all the above voice coding algorithms.
The following sections describe the factors that contribute to voice quality.
Table 1-2
Speech Quality Metrics
Algorithm
Bandwidth
Complexity
MOS Score
G.711
64 kbps
Very low
4.5
G.726
16, 24, 32, 40 kbps
Low
4.1 (32 kbps)
G.729a
8 kbps
Low–medium
4
G.729
8 kbps
Medium
4
G.723.1
6.3, 5.3 kbps
High
3.8
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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 1
Introducing Linksys Analog Telephone Adapters
Ensuring Voice Quality
Audio Compression Algorithm
Speech signals are sampled, quantized, and compressed before they are packetized and transmitted to
the other end. For IP Telephony, speech signals are usually sampled at 8000 samples per second with
12–16 bits per sample. The compression algorithm plays a large role in determining the voice quality of
the reconstructed speech signal at the other end. The Linksys ATA supports the most popular audio
compression algorithms for IP Telephony: G.711 a-law and μ-law, G.726, G.729a, and G.723.1.
The encoder and decoder pair in a compression algorithm is known as a codec. The compression ratio
of a codec is expressed in terms of the bit rate of the compressed speech. The lower the bit rate, the
smaller the bandwidth required to transmit the audio packets. Although voice quality is usually lower
with a lower bit rate, it is usually higher as the complexity of the codec gets higher at the same bit rate.
Silence Suppression
The Linksys ATA applies silence suppression so that silence packets are not sent to the other end to
conserve more transmission bandwidth. Instead, a noise level measurement can be sent periodically
during silence suppressed intervals so that the other end can generate artificial comfort noise that mimics
the noise at the other end (using a CNG or comfort noise generator).
Packet Loss
Audio packets are transported by UDP, which does not guarantee the delivery of the packets. Packets
may be lost or contain errors that can lead to audio sample drop-outs and distortions and lowers the
perceived voice quality. The Linksys ATA applies an error concealment algorithm to alleviate the effect
of packet loss.
Network Jitter
The IP network can induce varying delay of the received packets. The RTP receiver in the Linksys ATA
keeps a reserve of samples to absorb the network jitter, instead of playing out all the samples as soon as
they arrive. This reserve is known as a jitter buffer. The bigger the jitter buffer, the more jitter it can
absorb, but this also introduces bigger delay. Therefore, the jitter buffer size should be kept to a
relatively small size whenever possible. If the jitter buffer size is too small, late packets may be
considered lost and this lowers the voice quality. The Linksys ATA can dynamically adjust the size of
the jitter buffer according to the network conditions that exist during a call.
Echo
Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead to
near-end echo. The Linksys ATA has a near-end echo canceller with at least 8 ms tail length to
compensate for impedance match. The Linksys ATA also implements an echo suppressor with comfort
noise generator (CNG) so that any residual echo is not noticeable.
Page 19 / 186
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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 1
Introducing Linksys Analog Telephone Adapters
Feature Descriptions
Hardware Noise
Certain levels of noise can be coupled into the conversational audio signals because of the hardware
design. The source can be ambient noise or 60 Hz noise from the power adaptor. The Linksys ATA
hardware design minimizes noise coupling.
End-to-End Delay
End-to-end delay does not affect voice quality directly but is an important factor in determining whether
subscribers can interact normally in a conversation taking place over an IP network. Reasonable delay
figure should be about 50–100 ms. End-to-end delay larger than 300
ms is unacceptable to most callers.
The Linksys ATA supports end-to-end delays well within acceptable thresholds.
Feature Descriptions
The Linksys ATA is a full featured, fully programmable phone adapter that can be custom provisioned
within a wide range of configuration parameters. This chapter contains a high-level overview of features
to provide a basic understanding of the feature breadth and capabilities of the Linksys ATA.
•
SIP Proxy Redundancy, page 1-5
•
Supported Codecs, page 1-6
•
Streaming Audio Server and Music on Hold, page 1-6
•
Silence Suppression and Comfort Noise Generation, page 1-7
•
Modem and Fax Pass-Through, page 1-7
•
Adaptive Jitter Buffer, page 1-7
•
Other Features, page 1-8
SIP Proxy Redundancy
In typical commercial IP Telephony deployments, all calls are established through a SIP proxy server.
An average SIP proxy server may handle tens of 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 Linksys
ATA supports the use of backup SIP proxy servers so that service disruption should be nearly eliminated.
A relatively simple way to support proxy redundancy is to configure a static list of SIP proxy servers to
the Linksys ATA in its configuration profile where the list is arranged in some order of priority. The
Linksys ATA attempts to contact the highest priority proxy server whenever possible. When the
currently selected proxy server is not responding, the Linksys ATA automatically retries the next proxy
server in the list.
The dynamic nature of SIP message routing makes the use of a static list of proxy servers inadequate in
some scenarios. In deployments where user agents are served by different domains it is not feasible to
configure a static list of proxy servers for each domain.
One solution in this situation is through the use of DNS SRV records. The Linksys ATA can be
instructed to contact a SIP proxy server in a domain named in the SIP message. The Linksys ATA
consults the DNS server to get a list of hosts in the given domain that provides SIP services. If an entry
Page 20 / 186
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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 1
Introducing Linksys Analog Telephone Adapters
Feature Descriptions
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 Linksys ATA tries to contact the list of
hosts in the order of their stated priority.
If the Linksys ATA is currently using a lower priority proxy server, it periodically probes the higher
priority proxy to see whether it is back on line, and switches back to the higher priority proxy when
possible.
Supported Codecs
Negotiation of the optimal voice codec sometimes depends on the ability of the Linksys ATA device to
match a codec name with the codec used by the far-end device. The Linksys ATA allows the network
administrator to individually name the various codecs that are supported so that the Linksys ATA can
successfully negotiate the codec with the far-end equipment. The administrator can select which
low-bit-rate codec is to be used for each line. G.711a and G.711u are always enabled.
Note
When no static payload value is assigned per RFC 1890, the Linksys ATA can support dynamic payloads
for G.726.
Streaming Audio Server and Music on Hold
This feature allows you to attach an audio source to one of the Linksys ATA FXS ports and use it as a
streaming audio source device. The corresponding Line (1 or 2) can be configured as a streaming audio
server (SAS) such that when the Line is called, the Linksys ATA answers the call automatically and
Table 1-3
Codecs Supported by Linksys ATAs
Codec (Voice Compression
Algorithm)
Description
G.711 (A-law and mμ-law)
This very low complexity codec supports uncompressed 64 kbps
digitized voice transmission at one through ten 5 ms voice frames
per packet. This codec provides the highest voice quality and uses
the most bandwidth of any of the available codecs.
G.726
This low complexity codec supports compressed 16, 24, 32, and
40 kbps digitized voice transmission at one through ten 10 ms
voice frames per packet. This codec provides high voice quality.
G.729A
The ITU G.729 voice coding algorithm is used to compress
digitized speech. Linksys supports G.729. G.729A is a reduced
complexity version of G.729. It requires about half the processing
power to code G.729. The G.729 and G.729A bit streams are
compatible and interoperable, but not identical.
G.723.1
The Linksys ATA supports the use of ITU G.723.1 audio codec at
6.4 kbps. Up to two channels of G.723.1 can be used
simultaneously. For example, Line 1 and Line 2 can be using
G.723.1 simultaneously, or Line 1 or Line 2 can initiate a
three-way conference with both call legs using G.723.1.

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