LINKSYS 900 Administrator Guide
Linksys 900 Series IP Phone Administrator Guide
Document Version 3.0
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Fax: 949 823-1100
Linksys 900 Series IP Phone Administrator Guide
Copyright ©2007 Cisco Systems, Inc. All rights reserved.Speci fications are subject to change without notice. Linksys is a registered trademark or trademar k 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.
Compliance and Safety Information
This equipment has been tested and found to comply with the limits for a Class B digital de vice in accordance with the sp ecifications in p art 15 of the FCC rules. This product
bears the CE Marking indicating compliance with the 89/336/EEC directive. Standards to which conformity is Declared: EN 61000-4-2:1995, EN 61000-4-3:1997, EN
61000-4-4:1995, EN 61000-4-5:1995, EN 61000-4-6:1996, EN 61000-4-8:1994, EN 61000-4-11:1994, EN 61000-3-2:2001, EN 61000-3-3:1995 & EN 55022:1998
Class B Modifications to this product not authorized by Linksys could void FCC approval, thereby terminating end user authority to use this product. For indoor use only.
Read installation instructions before connecting to a po wer source. The electric pl ug and so cket must be accessib le at all times as this is the main method to disc onnect power
from the device.
Shock Hazard: Do not operate near water or similar fluid. Do not work with this device during periods of lightning activity. Do not touch wires at the end of cables or within
sockets.
One Year Limited Hardware Warranty
Linksys provides a one (1) year limited hardware warranty. Linksys warrants to customer that this product conforms to its published specifications and will be free from
defects in material and workmanship at the time of delivery and for a period of one year thereafter. Without limiting the foregoing, this warranty does not cover any defect
resulting from (i) any design or specificatio n supplied by an ent ity other t han Links ys, (ii) non-observ ance of techn ical opera ti ng parameters (e.g., exceeding limiti ng values),
or (iii) misuse, abuse, abnormal conditions or alteration by anyone other than Linksys. Replacement, Repair, Refund: After the receipt of an RMA (Return Materials
Authorization) request, Linksys will attempt to refund, repair or replace this device. To receive an RMA number for this device, contact the party from whom it.
Preface xi
Document Audience xi
Linksys 900 Series IP Telephones xi
How This Document is Organized xii
Document Conventions xii
Related Documentation xiii
Technical Support xiii
CONTENTS
CHAPTER
1 Introducing Linksys 900 Series IP Phones 1-1
Overview 1-1
SPA900 Series Features 1-2
SPA901 Features 1-4
SPA92x, SPA94x, and SPA962 Features 1-4
Ensuring Voice Quality 1-4
Feature Descriptions 1-6
SIP Proxy Redundancy 1-6
Supported Codecs 1-6
Other Features 1-7
Technology Background 1-10
Session Initiation Protocol 1-10
Using 900 Series Phones with a Firewall or Router 1-11
Network Address Translation 1-11
NAT Overview 1-12
NAT Types 1-12
Simple Traversal of UDP Through NAT 1-13
SIP-NAT Interoperation 1-13
CHAPTER
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2 Getting Started 2-1
Where to Go From Here 1-14
Linksys 900 Series IP Phones 2-1
Caring for Your Hardware 2-2
SPA901 2-2
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Contents
Front Panel and Side of Phone 2-2
Back Panel 2-3
SPA92x, SPA94x, and SPA962 Hardware Features 2-3
SPA921 2-4
Front Panel 2-4
Back Panel 2-5
SPA922 2-5
SPA941 2-5
Front Panel 2-6
Back Panel 2-6
SPA942 2-7
SPA962 2-7
Front Panel 2-8
Back Panel 2-8
Establishing Connectivity 2-8
Bandwidth Requirements 2-8
Installing the SPA900 Series IP Phone 2-9
Assembling the Phone and Connecting to the Network 2-9
Attaching the Desk Stand 2-10
Mounting the Phone to the Wall 2-10
Turning on the Phone 2-11
CHAPTER
Using the Administration Web Server 2-11
Connecting to the Administration Web Server 2-11
Administrator Account Privileges 2-12
Web Interface URLs 2-13
Upgrade URL 2-13
Resync URL 2-13
Reboot URL 2-14
Provisioning 2-14
Provisioning Capabilities 2-14
Configuration Profile 2-14
Using the Interactive Voice Response Interface 2-15
Using the IVR Menu 2-15
IVR Options 2-16
Entering a Password through the IVR 2-18
3 Managing Linksys 900 Series IP Phones 3-1
Using the LCD Display 3-1
LCD Display Controls 3-1
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Using Soft Keys 3-3
Entering and Saving Settings 3-4
Localization 3-5
Changing the Display Background (SPA962) 3-7
Call Appearances and Extensions 3-8
Line Key LEDs 3-9
Using Call Features 3-10
Selecting the Audio I/O Device and Line 3-11
Making Calls 3-11
Answering and Ending Calls 3-12
Hold and Resume 3-12
Call Waiting 3-12
Speed Dialing 3-13
Three-Way Conferencing 3-13
Attended Call Transfer 3-13
Blind Call Transfer 3-14
Call Back 3-14
Message Waiting Indication (MWI) 3-14
Accessing Voicemail 3-15
Muting Calls 3-15
Shared Call Appearances 3-15
Personal Directory 3-15
Caller and Called Name Matching 3-16
Dialing Assistance 3-16
Supplementary Services 3-16
Call Logs 3-17
Audio Volume Adjustment 3-18
Managing Ring Tones 3-19
Contents
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Configuring a Dial Plan 3-20
Dial Plan Digit Sequences 3-20
Dial Plan Rules 3-21
Digit Sequence Syntax 3-21
Element Repetition 3-21
Sub-sequence Substitution 3-21
Intersequence Tones 3-22
Number Barring 3-22
Interdigit Timer Master Override 3-22
Local Timer Overrides 3-22
Pause 3-22
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Contents
Dial Plan Examples 3-23
Dial Plan Timers 3-23
Interdigit Long Timer 3-24
Interdigit Short Timer 3-24
Dial Plans 3-24
System Administration 3-24
Reboot and Restart 3-25
Factory Reset 3-25
Password Protection 3-25
Managing the Time/Date 3-25
Daylight Saving Time 3-25
Using Star Codes to Activate/Deactivate Services 3-26
Disabling Services 3-28
Error and Log Reporting 3-29
Troubleshooting FAQ 3-29
CHAPTER
4 LCD Command Reference Guide 4-1
1 Directory 4-2
Entering Names and Numbers into the Directory 4-2
Entering Directory Names, Numbers and Ring Default 4-2
2 Speed Dial 4-3
3 Call History 4-3
Redial List 4-3
Answered Calls 4-4
Missed Calls 4-4
4 Ring Tone 4-4
5 Preferences 4-4
5.1 Block Caller ID 4-5
5.2 Block Anonymous Call 4-5
5.3 Do Not Disturb 4-5
5.4 Secure Call 4-5
5.5 Dial Assistance 4-6
5.6 Preferred Audio Device 4-6
6 Call Forward 4-6
6.1 CFWD All Number 4-6
6.2 CFWD Busy Number 4-6
6.3 CFWD No Ans Number 4-7
6.4 CFWD No Ans Delay 4-7
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7 Time/Date 4-7
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8 Voice Mail 4-7
9 Network 4-8
9.1 DCHP 4-8
9.2 Current IP Address 4-8
9.3 Host Name 4-9
9.4 Domain 4-9
9.5 Current NetMask 4-9
9.6 Current Gateway 4-9
9.7 Enable Web Server 4-9
9.8 Non DHCP IP Address 4-10
9.9 Non DHCP Subnet Mask 4-10
9.10 Non DHCP Default Route 4-10
9.11 Non DHCP DNS 1 4-10
9.12 Non DHCP DNS 2 4-10
9.13 Non DHCP NTP Server 1 4-10
9.14 Non DHCP NTP Server 2 4-11
Contents
10 Product Info 4-11
10.1 Product Name 4-11
10.2 Serial Number 4-11
10.3 Software Version 4-11
10.4 Hardware Version 4-12
10.5 MAC Address 4-12
10.6 Client Cert 4-12
11 Status 4-12
Phone 4-12
Ext 1/2/3/4 4-13
Line 1, 2,3,4 4-13
12 Reboot 4-13
13 Restart 4-13
14 Factory Reset 4-13
15 Set Password 4-14
16 Set LCD Contrast 4-14
17 CallPark Status 4-14
18 Language (SPA922, 942, and 962) 4-14
CHAPTER
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5 SPA900 Series Phone Field Reference 4-1
Info Tab 4-2
System Information 4-2
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Contents
Product Information 4-2
Phone Status 4-3
Ext 1/2/3/4/5/6 Status 4-4
Line 1/2/3/4/5/6 Status 4-4
Downloaded Ring Tone 4-5
System Tab 4-6
System Configuration 4-6
Internet Connection Type 4-6
Static IP Settings 4-7
PPPoE Settings 4-7
Optional Network Configuration 4-7
VLAN Settings 4-8
SIP Tab 4-9
SIP Parameters 4-9
SIP Timer Values (sec) 4-11
Response Status Code Handling 4-12
RTP Parameters 4-13
SDP Payload Types 4-14
4-17
NAT Support Parameters 4-17
Linksys Key System Parameters 4-18
Regional Tab 4-19
Call Progress Tones 4-19
Distinctive Ring Patterns 4-21
Control Timer Values (sec) 4-22
Vertical Service Activation Codes 4-22
Outbound Call Codec Selection Codes 4-27
Miscellaneous 4-29
Phone Tab 4-33
General 4-33
Line Key 1/2/3/4/5/6 4-33
Miscellaneous Line Key Settings 4-34
Line Key LED Pattern 4-34
Supplementary Services 4-35
Ring Tone 4-36
Auto Input Gain (dB) 4-37
Background Picture (SPA 962) 4-37
Ext 1/2/3/4/5/6 Tab 4-38
General 4-38
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Linksys 900 Series IP Phone Administrator Guide
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Share Line Appearance 4-38
NAT Settings 4-39
Network Settings 4-39
SIP Settings 4-40
Call Feature Settings 4-42
Proxy and Registration 4-43
Subscriber Information 4-44
Audio Configuration 4-46
Dial Plan 4-47
User 4-49
Call Forward 4-49
Speed Dial 4-50
Supplementary Services 4-50
Audio Volume 4-51
Phone GUI Menu Color Settings (SPA962) 4-51
Contents
APPENDIX
APPENDIX
I
NDEX
A Acronyms
B Glossary
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Contents
Linksys 900 Series IP Phone Administrator Guide
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Document Version 3.0
Preface
This guide describes administration and use of the Linksys SPA900 Series IP phones. It contains the
following sections:
• Document Audience, page xi
• Linksys 900 Series IP Telephones, page xi
• How This Document is Organized, pag e 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 LVS products
• VARs and resellers who need LVS configuration references
• System administrators or anyone who performs LVS 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 900 Series IP Telephones
The following summarizes the ports and features provided by the Linksys 900 Series IP phones
described in this document.
• SPA901—One line, small, affordable, no display.
• SPA921—One-line business phone.
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• SPA922—One-line business phone with Power over Ethernet (PoE) support and an extra Ethernet
port for connecting another device to the LAN.
Linksys 900 Series IP Phone Administrator Guide
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How This Document is Organized
• SPA941—Default two lines, upgradeable to four lines.
• SPA942—Default is two lines, upgradeable to four lines. Power over Ethernet (PoE) support and an
extra Ethernet port for connecting another device to the LAN.
• SPA962—Six lines, hi-res color display. Power over Ethernet (PoE) support and an extra Ethernet
port for connecting another device to the LAN.
Note PoE units (SPA922, SPA942, and SPA962) do not come with an external power adapter. The
PA100 power supply must be ordered separately if you are not using a PoE switch.
How This Document is Organized
This document is divided into the following chapters and appendices.
Chapter Contents
Chapter 1, “Introducing
Linksys 900 Series IP Phones”
Chapter 2, “Getting Started” This chapter describes how to use the different administration and
Chapter 3, “Managing Linksys
900 Series IP Phones”
Chapter 5, “SPA900 Series
Phone 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.
This chapter introduces the Linksys 900 Series IP phones.
configuration tools provided for managing a Linksys 9 00 Series IP
phone.
This chapter describes how to configure and monitor a Linksys 9 00
Series IP phone.
This chapter lists the function and usage for each field or parameter
on the Linksys 900 Series IP phone 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.
Linksys 900 Series IP Phone Administrator Guide
xii
configuration pages of the 900 Series phone administration web server. The
index at the end of this docume nt contains an alphabe tical listing of each
parameter, hyperlinked to the appropriate table in
Series Phone Field Reference”
Chapter 5, “SPA900
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Preface
Related Documentation
The following documentation provides additional information about features and functionality of
Linksys 900 Series IP phones:
• AA Quick Guide
• IVR Quick Guide
• SPA Provisioning Guide
The following documentation describes how to use other Linksys Voice System products:
• SPA9000 Administrator Guide
• LVS CTI Integration Guide
• LVS Integration with ITSP Hosted Voicemail Guide
• SPA900 Series IP Phones Administrator Guide
• Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment
• SPA 2.0 Analog Telephone Adapter Administrator Guide
Related Documentation
Technical Support
If you are an end user of LVS products and need technical support, contact the reseller or Internet
telephony service provider (ITSP) that supplied the equipment.
Technical support contact information for authorized Linksys Voice System partners is as follows:
• LVS Phone Support (requires an authorized partner PIN)
888 333-0244 Hours: 4am-6pm PST, 7 days a week
• E-mail support
voipsupport@linksys.com
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Technical Support
Preface
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Linksys 900 Series IP Phone Administrator Guide
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Overview
CHAPTER
1
Introducing Linksys 900 Series IP Phones
This guide describes the administration and use of Linksys analog telephone adapters (ATAs).
This chapter introduces the functionality of the Linksys 900 Series IP phones and includes the
following sections:
• Overview, page 1-2
• Feature Descriptions, page 1-8
• Technology Background, page 1-12
• Where to Go From Here, page 1-16
Table 1-1 summarizes the ports and features provided by the Linksys 900 Series IP pho nes
described in this document.
Table 1-1 Linksys SPA900 Series IP Phones
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Ethernet
(LAN)
Product Name RJ-45
SPA901 One (1) One (1) Small, affordable, no display
SPA921 One (1) One (1) One-line business phone
SPA922 Two (2) One (1) Power over Ethernet (PoE) support
SPA941 One (1) Four (4) Default is 2-lines active, upgradeable
SPA942 Two (2) Four (4) Default is 2-lines active, upgradeable. Power
SPA962 Two (2) Six (6) Six lines, hi-res color display
Note PoE units (SPA922, SPA942, and SPA962) do not come with an external power adapter. The
PA100 power supply must be ordered separately if you are not using a PoE switch.
Voice Lines Additional Features/Notes
over Ethernet (PoE) support
Linksys 900 Series IP Phone Administrator Guide
1-1
SPA900 Series Features
Chapter 1 Introducing Linksys 900 Series IP Phones
Figure 1-1 illustrates how the IP phones are connected in a VoIP network, including the
SPA3102, which acts as a SIP-PSTN gateway. As shown, the RTP400 and WRTP54G provide
QoS-enabled IP routers in addition to two ports for connecting analog telephone devices.
Figure 1-1 Linksys SPA900 Series IP Phones in a VoIP Network
PSTN
Up to 4 FXO lines
Local voicemail
SPA400
SIP-PSTN
gateway
Switch
ISP
Internet
SPA901, 921, 922, 941, 942, 962
SPA900 Series Features
The following telephony features are provided by the different models of the SP A900 Series IP
phones:
• Shared Line Appearance **
–
SPA901: Two Call Appearances Accessed Via Flash Key or Hook-Flash
–
SPA921 and SPA922: Two call appearances
–
SPA941 and SPA942: Four call appearances
–
SPA962: Six call appearances
• Line Status Indicators
FXS1
Fax/Analog
Phones
SPA9000
ITSP
IP PBX
FXS2
1-2
• Call Hold
• Music on Hold **
• Call Waiting
• Outbound Caller ID Blocking
• Call Transfer - Attended and Blind
Linksys 900 Series IP Phone Administrator Guide
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Chapter 1 Introducing Linksys 900 Series IP Phones
• Call Conferencing
• Call Pick Up - Selective and Group **
• Call Park and UnPark **
• Call Swap
• Call Back on Busy
• Call Blocking - Anonymous and Selective
• Call Forwarding - Unconditional, No Answer, On Busy
• Hot Line and Warm Line Automatic Calling
• Call Logs (60 entries each): Made, Answered, and Missed Calls
• Do Not Disturb (callers hear line busy tone)
• URI (IP) Dialing Support (Vanity Numbers)
• Date and Time with Intelligent Daylight Savings Support
• Call Duration and Start Time Stored in Call Logs
• Ten-User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com
• Speed Dialing
• Automatic Redial
• Configurable Dial/Numbering Plan Support - per Line
SPA900 Series Features
• Intercom **
• Group Paging **
• DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy
• Syslog, Debug, Report Generation, and Event Logging
• Secure Call Encrypted Voice Communication Support
• Built-in Web Server for Administration and Configuration with Multiple Security Levels
• Automated Provisioning, Multiple Methods. Up to 256-Bit Encryption: (HTTP, HTTPS, TFTP)
• Optionally Require Admin Password to Reset Unit to Factory Defaults
• NAT Traversal
• Set Preferred CODEC, Per Call, All Calls
• Call Return - Redial Last Caller
• Configurable Dial/Numbering Plan Support
• Support Linksys Voice System Automatic Configuration
** Feature requires support by SIP server
SPA901 Features
The SPA901 provides the following features that are not needed with the SPA900 Series IP
phones that provide an LCD display:
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• Built-in Interactive Voice Response (IVR) system to check status and change configuration
Linksys 900 Series IP Phone Administrator Guide
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SPA900 Series Features
• Ringer and Handset Volume Controls
• Handset Input Gain Adjustment
SPA92x, SPA94x, and SPA962 Features
The SPA921, SPA922, SPA941, SPA942, and SPA962 prov ide an LCD display and addition al
features that are not provided with the SPA901, including the following:
• Line Status Indicators: Active Line, Name, and Number
• Menu-Driven User Interface
• Digits Dialed with Number Auto-Completion
• Caller ID Name and Number and Outbound Caller ID Blockin g
• On-Hook Dialing
• Redial from Call Logs
• Personal Directory with Auto-dial (100 entries)
• On Hook Default Audio Configuration (Speakerphone and Headset)
• Called Number with Directory Name Matching
Chapter 1 Introducing Linksys 900 Series IP Phones
• Call Number using Name - Directory Matching or via Caller ID
• Subsequent Incoming Calls with Calling Name and Number
• Name and Identity (Text) Displayed at Start Up
• Distinctive Ringing Based on Calling and Called Number
Ensuring Voice Quality
Voice quality perceived by the subscribers of the IP Telephony service should be
indistinguishable from that of 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
1-4
G.729 8 kbps Medium 4
G.723.1 6.3, 5.3 kbps High 3.8
Linksys 900 Series IP Phone Administrator Guide
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Chapter 1 Introducing Linksys 900 Series IP Phones
Note SPA900 Series IP phones support all the above voice coding algorithms.
The following factors contribute to 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 play s
a large role in determining the voice quality of the reconstructed speech signal at the other end.
SP A90 0 Series IP phones support the most popular aud io 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 complex ity of the codec gets higher at the same
bit rate.
• Silence suppression—SP A900 Series IP phones ap ply 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 silenc e suppresse d inte rvals 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 lower the perceived voice qual ity. SPA900 Series IP phones apply an e rror concealment
algorithm to alleviate the effect of packet loss.
SPA900 Series Features
• Network jitter—The IP network can induce varying delay of received packets. The RTP receiver in
SPA900 Series IP phones keeps a reserve of samples to absorb the network jitter, instead of pl aying
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 jitter buffer size is too
small, many late packets may be considered as lost and thus lowers the voice quality . SPA900 Series
IP phones 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. SPA900 Series IP phones have a near-end echo canceller with at least 8 ms
tail length to compensate for impedance match. SPA900 Series IP phones implement an echo
suppressor with comfort noise generator (CNG) so that any residual echo is not noticeable.
• 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 SPA900 Series 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. A reasonable delay figure should be about 50–100 ms. End-to-end delay larger than
300
ms is unacceptable to most callers. SPA900 Series IP phones support end-to-end delays well
within acceptable thresholds.
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Feature Descriptions
Feature Descriptions
SPA900 Series IP phones are full featured, fully programmable IP phones 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 SPA900 Series IP phones.
• SIP Proxy Redundancy, page 1-8
• Supported Codecs, page 1-8
• Other Features, page 1-9
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. SPA900 Series IP phones support the use of backup SIP proxy servers so
that service disruption should be nearly eliminated.
Chapter 1 Introducing Linksys 900 Series IP Phones
A simple way to support proxy redundancy is to configure a static list of SIP proxy servers in
the SP A900 Series IP phone configuration profile, where the list is arranged in order of priority .
The SPA900 Series IP phone attempts to contact the highest priority proxy server wh enever
possible.
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, for instance, it would not be feasible to configure one static list of proxy servers per
covered domain into every SPA900 Series IP phone. One solution to th is situation is through
the use of DNS SR V records. SPA900 Series IP phones can be instructed to contact a SIP proxy
server in a domain named in SIP messages. The SPA900 Series IP phone consults the DNS
server to get a list of hosts in the given domain that provides 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 SPA900 Series IP phone tries to
contact the list of hosts in the order of their stated priority.
If the SPA900 Series IP phone is currently using a lower priority proxy server, it periodically
probes the higher priority proxy to see whether it is back on line, and attempts to switch back
to the higher priority proxy whenever possible.
Supported Codecs
Negotiation of the optimal voice codec sometimes depends on the ability of SPA900 Series IP
phone to “match” a codec name with the far-end device/gateway codec name. SPA900 Series
IP phones allow the network administrator to individually name the various codecs that are
supported such that the correct codec successfully negotiates with the far-end equipment.
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Linksys 900 Series IP Phone Administrator Guide
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Chapter 1 Introducing Linksys 900 Series IP Phones
The administrator can select the low-bit-rate codec used for each line. G.711a and G.711u are
always enabled.
Table 1-3 describes the codecs supported by the Linksys SPA900 Series IP
phones.
Table 1-3 Codecs Supported by Linksys SPA900 Series IP Phones
Codec (Voice Compression
Algorithm)
G.711 (A-law and mµ-law) This very low complexity codec supports uncompressed 64
G.726 This low complexity codec supports compressed 16, 24, 32,
G.729A The ITU G.729 voice coding algorithm is used to compress
Feature Descriptions
Description
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.
and 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.
G.723.1 SPA900 Series IP phones support 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.
When no static payload value is assigned per RFC 1890, SPA900 Series IP phones can support
dynamic payloads for G.726.
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Feature Descriptions
Other Features
Chapter 1 Introducing Linksys 900 Series IP Phones
Table 1-4 summarizes the features provided by SPA900 Series IP Phones.
Table 1-4 Linksys ATA Features
Feature Description
Music On Hold On a connected call, SPA900 Series IP phones may place the remote
party on call. If the remote party indicates that they can still receive
audio while the call is holding, the MOH server sends streaming
audio.
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 R TP 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 *) SPA900 Series
IP phones 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
SPA900 Series IP phones have config urable 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
Jitter
Buffer—Dynamic
(Adaptive)
This feature allows the user to hear the call progress tones (such as
ringing) that are generated from the far-end network.
SPA900 Series IP phones 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.
SPA900 Series IP phones have a Network Jitter Level control setting
for each line of service. The jitter level decides how aggressively
SPA900 Series IP phones try 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.
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Table 1-4 Linksys ATA Features (continued)
Feature Description
Feature Descriptions
Voice Activity
Detection with
Silence Suppression
and Comfort Noise
Generation
Configurable Dial
Plan with Interdigit
Timers
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.
SPA900 Series IP phones have 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.
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Report Generation
and Event Logging
SPA900 Series IP phones report a variety of status and error reports
to assist service providers in diagnosing problems and evaluating 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
SPA900 Series IP phones support detailed logging of all activities for
further debugging. The debug information may be sent to a
configured Syslog server. SPA900 Series IP phones provide
configuration settings that determine the type of activity/events that
should be logged, as for instance, a debug level setting.
Dynamic Payload When no static payload value is assigned per RFC 1890, SPA900
Series IP phones can support dynamic payloads for G.726.
Call Statistics and
Reporting
The statistics collected by SPA900 Series IP phones during normal
operation statistics are available in the Info tab. Line status is reported
for each line (1 and 2). Each line maintains up to 2 calls: Call 1 and 2.
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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 1-12
• Using 900 Series Phones with a Firewall or Router, page 1-12
• Using 900 Series Phones with a Firewall or Router, page 1-12
Session Initiation Protocol
Linksys 900 Series IP phones are implemented using open standards, such as Session Initiation
Protocol (SIP), allowing interoperation with all ITSPs supporting SIP.
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
Chapter 1 Introducing Linksys 900 Series IP Phones
Figure 1-2 illustrates a
SIP UA
2
4
SIP Proxy
RTP
SIP Proxy
3
SIP Proxy
1
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).
Using 900 Series Phones with a Firewall or Router
When using a 900 Series phone behind a firewall or router, make sure that the following ports
are not blocked:
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Chapter 1 Introducing Linksys 900 Series IP Phones
• SIP ports—By default, UDP port 5060 and 5061
• RTP ports—16384 to 16482
If security is not a concern in your environment, you can consider disabling SPI, if this function
exists on your firewall.
Network Address Translation
This section describes issues that arise when using the LVS system on a network behind a
network address translation (NAT) device. It includes the following topics:
• NAT Overview, page 1-13
• NAT Types, page 1-14
• Simple Traversal of UDP Through NAT, page 1-14
• SIP-NAT Interoperation, page 1-15
NAT Overview
Technology Background
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.
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, th at 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.
The ITSP may support NAT mapping using a Session Border Controller (see Figure 1-3).
Figure 1-3 NAT Support with Session Border Controller Provided by ITSP
192.168.1.101
Private IP address
192.168.1.1
192.168.1.102
NAT Device
DHCP
server
SPA9000
SIP Proxy
192.168.1.100
External IP address
assigned by ISP
ISP
Internet
ITSP
Session Border
Controller
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Technology Background
NAT Types
Chapter 1 Introducing Linksys 900 Series IP Phones
This is the preferred option because it eliminates the need for managing NAT on the 900 Series
phone. If this is not available, you need to discuss with the ITSP how to use the NAT Support
Parameters provided by the 900 Series phone, such as <Outbound Proxy> and <STUN Server
Enable>.
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 ways that NAT is implemented can be divided into the following categories:
• Full cone NAT—Also known as one-to-one NAT. All requests from the same int ernal 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 externa l 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.
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 NAT s (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:
1-12
Step 1 Enable debugging on the 900 Series phone:
1. Make sure you do no t have firewall runn ing on you r PC that could bl ock the syslog por t (by default
this is 514).
Linksys 900 Series IP Phone Administrator Guide
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Chapter 1 Introducing Linksys 900 Series IP Phones
2. On the administration web server , System tab, set <Debug Server> to the IP address and port number
of your syslog server.
Note that this address and port number has to be reachable from the SPA900 Series IP phone.
3. Set <Debug level> to 3, but do not 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.
SIP-NAT Interoperation
In the case of SIP, the addresses where messages/data should be sent to a 900 Series phone
system are embedded in the SIP messages sent by the device. If the 900 Series phone 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.
Technology Background
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 900 Series phone 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 900 Series phone system must perform the
following tasks:
• 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
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 900 Series phone 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 perio d, the 900 Seri es phone system continu es to send
periodic keep-alive packets through the mapping to extend its validity as necessary.
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Linksys 900 Series IP Phone Administrator Guide
1-13
Where to Go From Here
Where to Go From Here
To do this ... Refer to
Use the different administration and
configuration tools provided for managing
Linksys 900 Series IP phone.
Configure and monitor a Linksys 900 Series
IP phone.
Refer to the function and usage for each field
or parameter on the Linksys 900 Series IP
phone administration web server pages.
Find the expansion of an acronym used in this
document.
Find the definition of a term used in this
document.
Chapter 1 Introducing Linksys 900 Series IP Phones
Chapter 2, “Getting Started”
Chapter 3, “Managing Linksys 900 Series IP
Phones”
Chapter 5, “SPA90 0 Series Phone Field
Reference”
Appendix A, “Acronyms”
Appendix B, “Glossary”
The following documentation provides additional documentation for Linksys SPA900 Series
IP phones:
• IVR Quick Guide
• SPA Provisioning Guide
The following documentation describes how to use other Linksys Voice System products:
• SPA9000 Administrator Guide
• LVS CTI Integration Guide
• LVS Integration with ITSP Hosted Voicemail Guide
• Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment
• SPA 2.0 Analog Telephone Adapter Administrator Guide
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Linksys 900 Series IP Phone Administrator Guide
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CHAPTER
2
Getting Started
This chapter describes the tools and utilities available for administering Linksys SPA900 Series
phones. It includes the following sections:
• Linksys 900 Series IP Phones, page 2-1
• Establishing Connectivity, page 2-9
• Using the Administration Web Server, page 2-11
• Web Interface URLs, page 2-13
• Provisioning, page 2-14
• Using the Interactive Voice Response Interface, page 2-16
Note If the SPA900 Series IP phone is supplied or sponsored by an Internet telephone service provider (ITSP),
certain network and service settings may be preconfigured. Depending on the configuration policy,
access by an end user to specific configuration settings may be restricted or blocked.
Linksys 900 Series IP Phones
The Linksys SPA900 Series provides fully -featured VoIP phones that integrate with the
Linksys SPA9000 to provide connectivity to other local station s, and through an ITSP to IP
phones over the Internet, In addition, the optional SPA400 integrates with the SPA9000 and
provides connectivity between SPA900 IP phones and the PSTN. This section summarizes the
ports and hardware features provided by each device. It includes the following topics:
• Caring for Your Hardware, page 2-2
• SPA901 , page 2-2
• SPA92x , SPA94x, and SPA962 Hardware Feature s, page 2-3
• SPA92x , SPA94x, and SPA962 Hardware Feature s, page 2-3
• SPA922 , page 2-5
• SPA941 , page 2-5
• SPA942 , page 2-7
• SPA962 , page 2-7
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Linksys 900 Series IP Phones
Caring for Your Hardware
The Linksys 900 Series IP phones are electronic devices that should not be exposed to
excessive heat, sun, cold or water. To clean th e equipment, use a slightly moistened paper or
cloth towel. Do not spray or pour cleaning solution directly onto the hardware unit.
SPA901
The SPA901 provides an entry-level IP phone th at can be wall mounted (see Figure 2-1). The
following are the hardware features provided by the SPA901:
• Voice Mail Message Waiting Indicator Light
• Redial Button
• Dedicated Flash Button
• Volume Control Button Cycles Through Volume Levels. Controls Ringer and Handset Volume.
• Standard 12-Button Dialing Pad
• High Quality Handset and Cradle
• Ethernet LAN – 10BaseT RJ-45
• 5-volt DC Universal (100-240 Volt) Switching Power Adaptor
Chapter 2 Getting Started
Figure 2-1 SPA901
The following tables describe the status indicators and controls on the front of the device and
the ports on the back panel of the device.
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