Linksys ATA

Administrator Guide

Document Version 3.1

Chapter 1

Introducing Linksys Analog Telephone Adapters

Feature Descriptions

starts streaming audio to the calling party provided the FXS port is off-hook. If the FXS port is on-hook

when the incoming call arrives, the Linksys ATA replies with a SIP 503 response code to indicate

“Service Not Available.”

If an incoming call is auto-answered, but later the FXS port becomes on-hook, the Linksys ATA does

not terminate the call but continues to stream silence packets to the caller. If an incoming call arrives

when the SAS line has reached full capacity, the Linksys ATA replies with a SIP 486 response code to

indicate “Busy Here”. The SAS line can be set up to refresh each streaming audio session periodically

(via SIP re-INVITE) to detect whether the connection to the caller is down. If the caller does not respond

to the refresh message, the SAS line terminates the call so that the streaming resource can be used for

other callers.

On a connected call, the Linksys ATA may place the remote party on call. The only way to do this on

the SPA2102 is to perform a hook-flash to initiate a three-way call, or to swap two calls during

call-waiting. If the remote party indicates that they can still receive audio while the call is holding, the

SPA2102 can be set up to contact an auto-answering SAS as described in and have it stream audio to the

holding party. When used this way, the SAS is referred to as a MOH Server.

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 required bidirectional bandwidth for a single call. VAD uses

a very sophisticated algorithm to distinguish between speech and non-speech signals. Based on the

current and past statistics, the VAD algorithm decides whether or not speech is present. If the VAD

algorithm decides speech is not present, the silence suppression and comfort noise generation is

activated. This is accomplished by removing and not transmitting the natural silence that occurs in

normal two-way connection. The IP bandwidth is used only when someone is speaking. During the silent

periods of a telephone call, additional bandwidth is available for other voice calls or data traffic because

the silence packets are not being transmitted across the network.

Comfort Noise Generation provides artificially-generated background white noise (sounds), designed to

reassure callers that their calls are still connected during silent periods. If Comfort Noise Generation is

not used, the caller may think the call has been disconnected because of the “dead silence” periods

created by the VAD and Silence Suppression feature.

Modem and Fax Pass-Through

Modem Pass-through Mode can be triggered only by predialing the <Modem Line Toggle Code>. FAX

Pass-through Mode is triggered by CED/CNG tone or NSE events. Echo canceller is automatically

disabled for Modem Pass-through Mode only. Echo canceller is automatically disabled only if <FAX

Disable ECAN> (Line 1/2) 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 is automatically disabled for both FAX and

Modem pass-through as before. In addition, out-of-band DTMF Tx is disabled during modem or fax

pass-through.

Adaptive Jitter Buffer

The Linksys ATA can buffer incoming voice packets to minimize out-of-order packet arrival. This

process is known as jitter buffering. The jitter buffer size proactively adjusts or adapts in size, depending

on changing network conditions.

Linksys ATA

Administrator Guide

Document Version 3.1

Chapter 1

Introducing Linksys Analog Telephone Adapters

Feature Descriptions

The Linksys ATA has a Network Jitter Level control setting for each line of service. The jitter level

decides how aggressively the Linksys 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.

Other Features

The following table summarizes other features provided by Linksys ATAs.

Table 1-4

Linksys ATA Features

Feature

Description

International Caller ID

Delivery

In addition to support of the Bellcore (FSK) and Swedish/Danish (DTMF)

methods of Caller ID (CID) delivery, release 2.0 adds a large subset of

ETSI-compliant methods to support international CID equipment.

Different types of CID delivery method can be obtained by

mixing-and-matching some of the steps as shown.

Secure Calls

A user (if enabled by service provider or administrator) has the option to

make an outbound call secure in the sense that the audio packets in both

directions are encrypted.

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.

DTMF

In-Band and Out-of-Band (RFC 2833) (SIP INFO *) The Linksys 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.

Call Progress Tone

Generation

The Linksys ATA has configurable call progress tones. Parameters for each

type of tone may include number of frequency components, frequency and

amplitude of each component, and cadence information.

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.

Full Duplex Audio

Full-duplex is the ability to communicate in two directions simultaneously

so that more than one person can speak at a time. Half-duplex means that

only one person can talk at a time, like a CB radio or walkie-talkie, which

is unnatural in normal free-flowing two-way communications. The Linksys

ATA supports full-duplex audio.

Echo

Cancellation—Up to 8

ms Echo Tail

The SPA3102 supports hybrid line echo cancellation. This feature uses the

G.165 echo canceller to eliminate up to 8 ms of line echo. This feature does

not provide acoustic echo cancellation on endpoint devices; that is, an end

user speakerphone.

Linksys ATA

Administrator Guide

Document Version 3.1

Chapter 1

Introducing Linksys Analog Telephone Adapters

Technology Background

Technology Background

This section provides background information about the technology and protocols used by the ATA. It

includes the following topics:

•

Session Initiation Protocol, page 1-10

•

Network Address Translation, page 1-10

Signaling Hook Flash

Event

The Linksys ATA can signal hook flash events to the remote party on a

connected call. This feature can be used to provide advanced mid-call

services with third-party-call-control. Depending on the features that the

service provider offers using third-party-call-control, the following three

Linksys ATA features may be disabled to correctly signal a hook-flash

event to the softswitch:

•

Call Waiting Serv

•

Three Way Call Serv

•

Three Way Conf Serv

You can configure the length of time allowed for detection of a hook flash

using the Hook Flash Timer parameter on the Regional tab of the

administration web server.

Configurable Dial

Plan with Interdigit

Timers

The Linksys ATA has three configurable interdigit timers:

•

Initial timeout (T)—Handset off hook, no digit pressed yet.

•

Long timeout (L)—One or more digits pressed, more digits needed to

reach a valid number (as per the dial plan).

•

Short timeout (S)—Current dialed number is valid, but more digits

would also lead to a valid number.

Polarity Control

The Linksys 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.

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 removing the voltage

between the tip and ring momentarily. This feature is useful for

auto-answer equipment, which then knows when to disengage.

Report Generation and

Event Logging

The Linksys ATA reports a variety of status and error reports to assist

service providers to diagnose problems and evaluate the performance of

their services. The information can be queried by an authorized agent,

using HTTP with digested authentication, for instance. The information

may be organized as an XML page or HTML page.

Syslog and Debug

Server Records

The Linksys ATA supports detailed logging of all activities for further

debugging. The debug information may be sent to a configured Syslog

server. Via the configuration parameters, the Linksys ATA allows some

settings to select which type of activity/events should be logged, as for

instance, a debug level setting.

Table 1-4

Linksys ATA Features

Feature

Description

Linksys ATA

Administrator Guide

Document Version 3.1

Chapter 1

Introducing Linksys Analog Telephone Adapters

Technology Background

Session Initiation Protocol

Linksys ATAs are implemented using open standards, such as Session Initiation Protocol (SIP),

allowing interoperation with all ITSPs supporting SIP.

Figure 1-2

illustrates a SIP request for connection

to another subscriber in the network. The requestor is called the user agent server (UAS), while the

recipient is called the user agent client (UAC).

Figure 1-2

SIP Requests and Responses

In a SIP VoIP network, when the SIP proxy receives a request from a UAS for a connection and it does

not know the location of the UAC, it forwards the message to another SIP proxy in the network. Once

the UAC is located and the response is routed back to the UAS, a direct peer-to-peer session is

established between the two UAs. The actual voice traffic is transmitted between UAs over dynamically

assigned ports using the Real-time Protocol (RTP).

Network Address Translation

This section describes issues that arise when using a Linksys ATA on a network behind a network

address translation (NAT) device. It includes the following topics:

•

NAT Overview, page 1-10

•

NAT Types, page 1-11

•

Simple Traversal of UDP Through NAT, page 1-12

•

SIP-NAT Interoperation, page 1-12

NAT Overview

Network Address Translation (NAT) allows multiple devices to share the same public, routable, IP

address for establishing connections over the Internet. NAT is typically performed by a router that

forwards packets between the Internet and the internal, private network.

SIP UA

SIP UA

SIP Proxy

SIP Proxy

RTP

1

2

3

4

SIP Proxy

Linksys ATA

Administrator Guide

Document Version 3.1

Chapter 1

Introducing Linksys Analog Telephone Adapters

Technology Background

A typical application of a NAT is to allow all the devices in a subscriber home network to access the

Internet through a router with a single public IP address assigned by an ISP. The IP header of the packets

sent from the private network to the public network is substituted by NAT with the public IP address and

a port assigned by the router. The receiver of the packets on the public network sees the packets as

coming from the external address instead of the private address of the device.

The association between a private address and port and a public address and port is called a NAT

mapping

. This mapping is maintained for a short period of time, that varies from a few seconds to several

minutes. The expiration time is extended whenever the mapping is used to send a packet from the source

device.

Figure 1-3

NAT Support with Session Border Controller Provided by ITSP

The ITSP may support NAT mapping using a Session Border Controller (see

Figure 1-3

). This is the

preferred option because it eliminates the need for managing NAT on the Linksys ATA. If this is not

available, you will need to discuss with the ITSP how to use the NAT Support Parameters provided by

the Linksys ATA, such as <Outbound Proxy> and <STUN Server Enable>.

NAT Types

The different ways that NAT is implemented is sometimes divided into the following categories:

•

Full cone NAT—Also known as one-to-one NAT. All requests from the same internal IP address and

port are mapped to the same external IP address and port. An external host can send a packet to the

internal host, by sending a packet to the mapped external address

•

Restricted cone NAT—All requests from the same internal IP address and port are mapped to the

same external IP address and port. Unlike a full cone NAT, an external host can send a packet to the

internal host only if the internal host had previously sent a packet to it.

•

Port restricted cone NAT/symmetric NAT—Port restricted cone NAT or symmetric NAT is like a

restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can

send a packet to a particular port on the internal host only if the internal host had previously sent a

packet from that port to the external host.

Internet

ITSP

NAT Device

Linksys ATA

ISP

DHCP

server

192.168.1.100

Session Border

Controller

Private IP address

192.168.1.1

External IP address

assigned by ISP