LINKSYS ATA Administrator Guide

Linksys ATA Administrator Guide

Document Version 3.2

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Linksys ATA Administrator Guide

Copyright ©2007 Cisco Systems, Inc. All rights reserved.Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and product names are trademarks or registered trademarks of their respective holders.

Preface

CHAPTER

Document Audience

i-xi

Linksys Analog Telephone Adapters i-xi

How This Document is Organized i-xii

Document Conventions i-xii

Preface

This guide describes administration and use of the Linksys Analog Telephone Adapters (ATAs). It contains the following sections:

• Document Audience, page xi

• Linksys Analog Telephone Adapters, page xi

• How This Document is Organized, page xii

• Document Conventions, page xii

• Related Documentation, page xiii

• Technical Support, page xiii

Document Audience

This document is written for the following audience:

• Service providers offering services using Linksys VoIP products

• VARs and resellers who need configuration information for Linksys VoIP products

• System administrators or anyone who performs Linksys VoIP product installation and

administration

Note This guide does not provide the configuration information required by specific service

providers. Please consult with the service provider for specific service parameters.

Linksys Analog Telephone Adapters

The following summarizes the ports and features provided by the Linksys ATAs described in this document.

• PAP2T—Voice adapter with two FXS ports

• SPA1001—Small VoIP adapter

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• SPA2102—Voice adapter with router

Linksys ATA Administrator Guide

How This Document is Organized

• SPA3102—Voice adapter with router and PSTN connectivity

• SPA8000—Voice adapter supporting up to eight FXS connections

• AG310—ADSL2+ gateway with VoIP and PSTN connectivity

• RTP300—IP router with two FXS ports

• WRP400—Wireless-G IP router with FXS ports

• WRTP54G—Wireless-G IP router with two FXS ports

• WRT54GP2—Wireless-G IP router with two FXS ports

How This Document is Organized

This document is divided into the following chapters and appendices.

Chapter Contents

Chapter 1, “Introducing Linksys Analog Telephone Adapters”

Chapter 2, “Getting Started” This chapter describes how to use the different administration and

Chapter 3, “Configuring Linksys ATA s ”

Chapter 4, “Configuring the PSTN Gateway (FXO)”

Chapter 5, “Linksys ATA Field Reference”

Appendix A, “Acronyms” This appendix provides the expansion of acronyms used in this

Appendix B, “Glossary” This appendix defines the terms used in this document.

Appendix C, “User Guidelines” This appendix summarizes the operations of ATA user features.

This chapter introduces the Linksys Analog Telephone Adapters (ATAs).

configuration tools provided for managing Linksys ATAs.

This chapter describes how to complete the basic configuration of a Linksys ATA.

This chapter describes how to configure the Linksys SPA3102 and AG310 for providing PSTN connectivity.

This chapter lists the function and usage for each field or parameter on the ATA administration web server pages.

document.

Preface

Document Conventions

The following are the typographic conventions used in this document.

Typographic Element Meaning

Boldface Indicates an option on a menu or a literal value to be entered in a field.

<parameter> Angle brackets (<>) are used to identify parameters that appear on the

Italic Indicates a variable that should be replaced with a literal value.

Monospaced Font Indicates code samples or system output.

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configuration pages of the Linksys ATA administration web server. The index at the end of this document contains an alphabetical listing of each parameter, hyperlinked to the appropriate table in

Field Reference”

Chapter 5, “Linksys ATA

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Preface

Technical Support

Introducing Linksys Analog Telephone Adapters

This guide describes the administration and use of Linksys analog telephone adapters (ATAs). This chapter introduces the functionality of the Linksys ATAs and includes the following sections:

• Overview, page 1-1

• Ensuring Voice Quality, page 1-3

• Feature Descriptions, page 1-5

• Technology Background, page 1-9

• Where to Go From Here, page 1-13

Overview

Table 1-1 summarizes the ports and features provided by the Linksys ATAs described in this document.

Table 1-1 Linksys ATAs

CHA PTER

FXS Product Name

PAP2T Two (2) — One (1) — Two (2) Voice adapter with two FXS ports

SPA1001 One (1) — One (1) — One (1) Small VoIP adapter

SPA2102 Two (2) — One (1) — One (1) Voice adapter with router

SPA3102 One (1) One (1) One (1) One (1) Two (2) Voice adapter with router and PSTN

SPA8000 Eight (8) — One (1) Maintenance

RTP300 Two (2) — One (1) Four (4) Two (2) IP router with two FXS ports

WRP400 WRTP54G WRT54GP2

AG310 One (1) One (1) One (1) One (1) One (1) ADSL2+ gateway with VoIP and

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(Analog

Phone)

Two (2) — One (1) Four (4) Two (2) Wireless-G IP router with two FXS

FXO PSTN Connection

RJ-45 Internet (WAN)

RJ-45 Ethernet (LAN)

only

Configurable Voice Lines

Four (4) Voice adapter with support for up to

Description

connectivity

eight FXS devices

ports

PSTN connectivity

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Overview

Chapter 1 Introducing Linksys Analog Telephone Adapters

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

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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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Chapter 1 Introducing Linksys Analog Telephone Adapters

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.

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

Note The Linksys ATA supports all the above voice coding algorithms.

The following sections describe the factors that contribute to voice quality.

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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).

Chapter 1 Introducing Linksys Analog Telephone Adapters

Packet Loss

Network Jitter

Echo

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.

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.

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.

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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 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 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

Feature Descriptions

Hz noise from the power adaptor. The Linksys ATA

ms is unacceptable to most callers.

• 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

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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.

Table 1-3 Codecs Supported by Linksys ATAs

Codec (Voice Compression Algorithm)

G.711 (A-law and mµ-law) This very low complexity codec supports uncompressed 64 kbps

G.726 This low complexity codec supports compressed 16, 24, 32, and

G.729A The ITU G.729 voice coding algorithm is used to compress

G.723.1 The Linksys ATA supports the use of ITU G.723.1 audio codec at

Chapter 1 Introducing Linksys Analog Telephone Adapters

Description

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.

40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet. This codec provides high voice quality.

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.

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.

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

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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

Feature Descriptions

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.

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Feature Descriptions

Other Features

Chapter 1 Introducing Linksys Analog Telephone Adapters

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.

The following table summarizes other features provided by Linksys ATAs.

Table 1-4 Linksys ATA Features

Feature Description

International Caller ID Delivery

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

Adjustable Audio Frames Per Packet

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

Call Progress Tone Generation

Call Progress Tone Pass Through

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

Echo Cancellation—Up to 8 ms Echo Tail

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.

make an outbound call secure in the sense that the audio packets in both directions are encrypted.

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.

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.

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.

This feature allows the user to hear the call progress tones (such as ringing) that are generated from the far-end network.

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.

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.

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Table 1-4 Linksys ATA Features

Feature Description

Signaling Hook Flash Event

Configurable Dial Plan with Interdigit Timers

Technology Background

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.

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).

Polarity Control The Linksys ATA allows the polarity to be set when a call is connected and

Calling Party Control Calling Party Control (CPC) signals to the called party equipment that the

Report Generation and Event Logging

Syslog and Debug Server Records

Technology Background

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

would also lead to a valid number.

when a call is disconnected. This feature is required to support some pay phone system and answering machines.

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.

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.

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.

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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

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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. 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

SIP UA

RTP

Chapter 1 Introducing Linksys Analog Telephone Adapters

Figure 1-2 illustrates a SIP request for connection

SIP Proxy

SIP UA

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

SIP Proxy

NAT Overview

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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.

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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

Technology Background

NAT Types

Private IP address

192.168.1.1

Linksys ATA

192.168.1.100

NAT Device

DHCP server

External IP address assigned by ISP

ISP

Internet

ITSP

Session Border

Controller

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>.

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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.

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Technology Background

With symmetric NAT all requests from the same internal IP address and port to a specific destination IP address and port are mapped to a unique external source IP address and port. If the same internal host sends a packet with the same source address and port to a different destination, a different mapping is used. Only an external host that receives a packet can send a UDP packet back to the internal host.

Simple Traversal of UDP Through NAT

Simple Traversal of UDP through NATs (STUN) is a protocol defined by RFC 3489, that allows a client behind a NAT device to find out its public address, the type of NAT it is behind, and the port associated on the Internet connection with a particular local port. This information is used to set up UDP communication between two hosts that are both behind NAT routers. Open source STUN software can be obtained at the following website:

http://www.voip-info.org/wiki-Open+Source+VOIP+Software

STUN does not work with a symmetric NAT router. To determine the type of NAT your router uses, complete the following steps:

Step 1 Enable debugging on the Linksys ATA:

1. Make sure you do not have firewall running on your PC that could block the syslog port (by default

this is 514).

2. On the administration web server, System tab, set <Debug Server> to the IP address and port number

of your syslog server.

Chapter 1 Introducing Linksys Analog Telephone Adapters

Note that this address and port number has to be reachable from the Linksys ATA.

3. Set <Debug level> to 3 but you do not need to change the value of the <syslog server> parameter.

4. To capture SIP signaling messages, under the Line tab, set <SIP Debug Option> to Full. The output

is named syslog.514.log.

Step 2 To determine the type of NAT your router is using set <STUN Test Enable> to yes.

Step 3 View the syslog messages to determine if your network uses symmetric NAT or not.

SIP-NAT Interoperation

In the case of SIP, the addresses where messages/data should be sent to a Linksys ATA system are embedded in the SIP messages sent by the device. If the Linksys ATA system is sitting behind a NAT device, the private IP address assigned to it is not usable for communications with the SIP entities outside the private network.

Note If the ITSP offers an outbound NAT-Aware proxy, this discovers the public IP address from the remote

endpoint and eliminates the need to modify the SIP message from the UAC.

The Linksys ATA system must substitute the private IP address information with the proper external IP address/port in the mapping chosen by the underlying NAT to communicate with a particular public peer address/port. For this, the Linksys ATA system needs to perform the following tasks:

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• Discover the NAT mappings used to communicate with the peer.

This can be done with the help of an external device, such as a STUN server. A STUN server

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responds to a special NAT-Mapping-Discovery request by sending back a message to the source IP address/port of the request, where the message contains the source IP address/port of the original request. The Linksys ATA system can send this request when it first attempts to communicate with a SIP entity over the Internet. It then stores the mapping discovery results returned by the server.

• Communicate the NAT mapping information to the external SIP entities.

If the entity is a SIP Registrar, the information should be carried in the Contact header that overwrites the private address/port information. If the entity is another SIP UA when establishing a call, the information should be carried in the Contact header as well as in the SDP embedded in SIP message bodies. The VIA header in outbound SIP requests might also need to be substituted with the public address if the UAS relies on it to route back responses.

• Extend the discovered NAT mappings by sending keep-alive packets.

Because the mapping is alive only for a short period, the Linksys ATA system continues to send periodic keep-alive packets through the mapping to extend its validity as necessary.

Where to Go From Here

To Refer to

Use the different administration and

Chapter 2, “Getting Started”

configuration tools provided for managing Linksys ATAs.

Complete the basic configuration of a Linksys

Chapter 3, “Configuring Linksys ATAs”

ATA.

Configure the Linksys SPA3102 or AG310 for providing PSTN connectivity.

Look up the function and usage for each field or

Chapter 4, “Configuring the PSTN Gateway (FXO)”

Chapter 5, “Linksys ATA Field Reference”

parameter on the ATA administration web server pages.

Look up the expansion of acronyms used in this

Appendix A, “Acronyms”

document.

Define the terms used in this document. Appendix B, “Glossary”

Understand the operations of ATA user features. Appendix C, “User Guidelines”

The following documentation provides additional information about features and functionality of Linksys ATAs:

• AA Quick Guide

• IVR Quick Guide

• SPA Provisioning Guide

The following documentation describes how to use and configure other Linksys VoIP products:

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• SPA9000 Administrator Guide

• LVS CTI Integration Guide

• LVS Integration with ITSP Hosted Voicemail Guide

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Where to Go From Here

Chapter 1 Introducing Linksys Analog Telephone Adapters

• SPA900 Series IP Phones Administrator Guide

• Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment

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CHA PTER

Getting Started

This chapter provides a brief description of each Linksys ATA and describes the tools and utilities available for administration. It includes the following sections:

• Linksys Analog Telephone Adapters (ATAs), page 2-1

• Establishing Connectivity, page 2-14

• Connecting the SPA8000, page 2-16

• Using the Interactive Voice Response Interface, page 2-18

• Using the Administration Web Server, page 2-22

The following ATAs provide Layer 2 (data link) and Layer 3 (IP) configuration options:

• AG310

• WRP400

• WRTP54/54G

• RTP300

For information about configuring these options, including private virtual circuits (PVCs), PPPoE, and IP routing, refer to the user guide for the specific device.

For information about configuring the Public Switch Telephone Network (PSTN) connectivity options for the SPA3102 and AG310 refer to

Chapter 3, “Configuring Linksys ATAs.”

Linksys Analog Telephone Adapters (ATAs)

Linksys ATAs convert voice traffic into data packets for transmission over an IP network. This section illustrates and summarizes the ports and LEDs provided by each model. It includes the following topics:

• Caring for Your Hardware, page 2-2

• PAP2T, page 2-3

• SPA2102, page 2-6

• SPA3102, page 2-7

• SPA8000, page 2-9

• AG310, page 2-2

• WRP400, page 2-10

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Linksys Analog Telephone Adapters (ATAs)

• WRTP54G, page 2-11

• WRT54GP2, page 2-13

• RTP300, page 2-4

Caring for Your Hardware

The Linksys ATA is an electronic device that should not be exposed to excessive heat, sun, cold or water. To clean the equipment, use a slightly moistened paper or cloth towel. Do not spray or pour cleaning solution directly onto the hardware unit.

AG310

The AG310 provides a high-speed ADSL2/+ modem along with a four-port Ethernet switch and a PSTN gateway (see

Note Througout this document, when references are made to the software configuration of the SPA3102, the

information also applies to the AG310, which provides the same functionality as the SPA3102, as well as an ADSL modem and a four-port Ethernet switch.

Figure 1-4).

Chapter 2 Getting Started

Figure 1-4 AG310

The following tables describe the LEDS on the front panel and the ports on the back panel of the device.

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