Polycom SoundPoint IP 600 SIP, SoundPoint IP 500 SIP Administrator's Manual

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Administrator Guide

SoundPoint® IP SIP

Version 1.2.0

4 March 2004

Page 2

Notices

1. Specifications subject to change without notice.

Polycom, Inc.

1565 Barber Lane, Milpitas CA 95035, USA

www.polycom.com

Part Number: 1725-11518-120 Rev A

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Administrator Guide - SoundPoint® IP Table of Contents

Table of Contents

1 Overview ......................................................... 1

2 Installation and Operation ................................. 3

2.1 Installation Models ..................................................3

2.2 Installation Process..................................................4

2.2.1 Basic Network Setup....................................................................................................................... 4

2.2.1.1 DHCP or Manual TCP/IP Setup..............................................................................................5

2.2.1.2 Local User Interface Setup Menus ..........................................................................................6

2.2.1.3 Reset to Factory Defaults......................................................................................................9

2.2.2 Application Configuration................................................................................................................ 9

2.2.2.1 Centralized Configuration......................................................................................................10

2.2.2.2 Local Telephone Configuration...............................................................................................15

2.2.2.3 SIP Authentication Parameters...............................................................................................16

3 Features .......................................................... 17

3.1 Basic Features.........................................................17

3.1.1 Call Log ......................................................................................................................................... 17

3.1.2 Call Timer ...................................................................................................................................... 17

3.1.3 Call Waiting ................................................................................................................................... 17

3.1.4 Called Party Identification............................................................................................................... 17

3.1.5 Calling Party Identification.............................................................................................................. 18

3.1.6 Configurable Feature Keys.............................................................................................................. 18

3.1.7 Connected Party Identification ........................................................................................................ 21

3.1.8 Context Sensitive Volume Control.................................................................................................... 21

3.1.9 Customizable Audio Sound Effects................................................................................................... 21

3.1.10 Message Waiting Indication .......................................................................................................... 22

3.1.11 Distinctive Incoming Call Treatment............................................................................................... 22

3.1.12 Distinctive Ringing........................................................................................................................ 23

3.1.13 Distinctive Call Waiting ................................................................................................................. 23

3.1.14 Do-Not-Disturb............................................................................................................................. 24

3.1.15 Handset, Headset, and Speakerphone........................................................................................... 24

3.1.16 Local Contact Directory................................................................................................................. 25

3.1.16.1 Local Contact Directory File Format.......................................................................................26

3.1.17 Local Digit Map............................................................................................................................ 27

3.1.18 Microphone Mute ......................................................................................................................... 28

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Administrator Guide - SoundPoint® IP Table of Contents

3.1.19 Multiple Call Appearances............................................................................................................. 28

3.1.20 Shared Call Appearances .............................................................................................................. 29

3.1.21 Customizable Fonts and Indicators................................................................................................. 30

3.1.22 Soft Key-Driven User Interface...................................................................................................... 31

3.1.23 Speed Dial ................................................................................................................................... 31

3.1.24 Time and Date Display.................................................................................................................. 32

3.2 Call Management Features........................................33

3.2.1 Automatic Off-hook Call Placement................................................................................................. 33

3.2.2 Call Hold........................................................................................................................................ 33

3.2.3 Call Transfer .................................................................................................................................. 34

3.2.4 Three-Way Conference, Local.......................................................................................................... 34

3.2.5 Call Diversion (Call Forward) .......................................................................................................... 34

3.3 Audio Processing Features ........................................35

3.3.1 Low-Delay Audio Packet Transmission............................................................................................. 35

3.3.2 Jitter Buffer and Packet Error Concealment ..................................................................................... 35

3.3.3 Local Conference Mixing................................................................................................................. 36

3.3.4 Voice Activity Detection (VAD)......................................................................................................... 36

3.3.5 DTMF Tone Generation ................................................................................................................... 37

3.3.6 DTMF Event RTP Payload ................................................................................................................ 37

3.3.7 Acoustic Echo Cancellation (AEC) ..................................................................................................... 37

3.3.8 Audio Codecs.................................................................................................................................. 38

3.3.9 Background Noise Suppression (BNS).............................................................................................. 39

3.3.10 Comfort Noise Fill......................................................................................................................... 39

3.3.11 Automatic Gain Control (AGC)....................................................................................................... 39

3.4 Presence and Instant Messaging Features ...................40

3.4.1 Presence........................................................................................................................................ 40

3.4.2 Instant Messaging .......................................................................................................................... 40

3.5 Localization Features ...............................................41

3.5.1 Multilingual User Interface ............................................................................................................. 41

3.5.2 Downloadable Fonts ....................................................................................................................... 43

3.5.3 Synthesized Call Progress Tones...................................................................................................... 43

3.6 Advanced Server Features ........................................43

3.6.1 Voicemail Integration ..................................................................................................................... 43

3.6.2 Multiple Registrations..................................................................................................................... 44

3.6.3 Server Redundancy ........................................................................................................................ 45

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Administrator Guide - SoundPoint® IP Table of Contents

3.6.3.1 DNS SIP Server Name Resolution...........................................................................................46

3.7 Accessory Internet Features.......................................46

3.7.1 Services Browser ............................................................................................................................ 46

3.8 Security Features.....................................................46

3.8.1 Local User and Administrator Privilege Levels.................................................................................. 46

3.8.2 Incoming Signaling Validation......................................................................................................... 47

4 Optimizing SoundPoint....................................... 49

4.1 Ethernet Switch .......................................................49

4.2 Application Network Setup .......................................49

4.2.1 RTP Ports....................................................................................................................................... 49

4.2.2 Working with Network Address Translation (NAT) ............................................................................50

4.3 Updating and Rebooting...........................................51

4.4 Event Logging .........................................................52

4.5 Audio Quality Issues and VLANs ................................53

4.5.1 IP TOS ........................................................................................................................................... 53

4.5.2 IEEE 802.1p/Q............................................................................................................................... 54

4.5.3 RTCP Support ................................................................................................................................. 54

4.6 Configuration Files...................................................55

4.6.1 Core Configuration - ipmid.cfg ........................................................................................................ 55

4.6.1.1 Localization <localization/> .................................................................................................55

4.6.1.2 User Preferences <user_preferences/> ..................................................................................59

4.6.1.3 Tones <tones/> .................................................................................................................60

4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> ..............................................................62

4.6.1.5 Sound Effects <sound_effects/>...........................................................................................63

4.6.1.6 Voice Settings <voice/> ......................................................................................................68

4.6.1.7 Quality of Service <QOS/> ..................................................................................................78

4.6.1.8 Basic TCP/IP <TCP_IP/>.....................................................................................................80

4.6.1.9 Web Server <HTTPD/>........................................................................................................82

4.6.1.10 Call Handling Configuration <call/>.....................................................................................83

4.6.1.11 Directory <directory/>.......................................................................................................84

4.6.1.12 Fonts <font/>..................................................................................................................85

4.6.1.13 Keys <keys/>..................................................................................................................88

4.6.1.14 Bitmaps <bitmaps/>.........................................................................................................91

4.6.1.15 Indicators <indicators/>.....................................................................................................92

4.6.1.16 Event Logging <logging/> .................................................................................................94

4.6.1.17 Security <security/> .........................................................................................................97

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4.6.1.18 Provisioning <provisioning/>..............................................................................................97

4.6.1.19 RAM Disk <RAMdisk/>......................................................................................................98

4.6.1.20 Request <request/>..........................................................................................................99

4.6.1.21 Feature <feature/>...........................................................................................................99

4.6.1.22 Resource <resource/>.......................................................................................................100

4.6.2 SIP Configuration - sip.cfg .............................................................................................................. 101

4.6.2.1 Protocol <volpProt/>...........................................................................................................101

4.6.3 Per-phone Configuration - phone1.cfg............................................................................................. 108

4.6.3.1 Registration <reg/> ............................................................................................................108

4.6.3.2 Calls <call/>......................................................................................................................110

4.6.3.3 Diversion <divert/>.............................................................................................................111

4.6.3.4 Dial Plan <dialplan/>..........................................................................................................113

4.6.3.5 Messaging <msg/> ............................................................................................................115

4.6.3.6 Network Address Translation <nat/> .....................................................................................116

4.6.3.7 User Preferences <user_preferences/> ..................................................................................117

5 Session Initiation Protocol (SIP)........................... 119

5.1 Basic Protocols ........................................................119

5.1.1 RFC and Internet Draft Support....................................................................................................... 119

5.1.2 Request Support............................................................................................................................. 119

5.1.3 Header Support.............................................................................................................................. 120

5.1.4 Response Support........................................................................................................................... 122

5.1.4.1 1xx Responses - Provisional ..................................................................................................122

5.1.4.2 2xx Responses - Success ......................................................................................................122

5.1.4.3 3xx Responses - Redirection..................................................................................................123

5.1.4.4 4xx Responses - Request Failure............................................................................................123

5.1.4.5 5xx Responses - Server Failure..............................................................................................124

5.1.4.6 6xx Responses - Global Failure ..............................................................................................124

5.1.5 Hold Implementation...................................................................................................................... 125

5.1.6 Reliability of Provisional Responses................................................................................................. 125

5.1.7 Transfer......................................................................................................................................... 125

5.1.8 Third Party Call Control................................................................................................................... 125

5.2 Protocol Extensions..................................................125

5.2.1 RFC and Internet Draft Support....................................................................................................... 126

5.2.2 Request Support............................................................................................................................. 126

5.2.3 Header Support.............................................................................................................................. 126

5.2.4 SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE).......................................... 126

5.2.5 Shared Call Appearance Signaling................................................................................................... 127

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Administrator Guide - SoundPoint® IP Overview

1 Overview

SoundPoint® IP is a feature-rich, enterprise-class voice communications terminal for Ethernet TCPIP networks. It is designed to facilitate high-quality audio and text message communications. This telephone is an endpoint in the overall network topology designed to interoperate with other compatible equipment including application servers, media servers, internetworking gateways, voice bridges, and other endpoints.

Remote

Boot Server

10/100

Ethernet

Switch

Ethernet

Internet

789101112

123456

789101112

123456

Router / Firewall

12x6x8x2x9x3x10x4x11x5x7x

Polycom

SoundPoint IP

500/600s

10/100

12x6x8x2x9x3x10x4x11x5x7x

Ethernet

Hub

IDC

789101112

Ethernet

123456

789101112

Ethernet

123456

Remote

Application

Server

12x6x8x2x9x3x10x4x11x5x7x

PSTN

Gateway

Modem Bank

12x6x8x2x9x3x10x4x11x5x7x

Ethernet

12x6x8x2x9x3x10x4x11x5x7x

Switches

Voice Bridge

IDC

Local

Application

Server

Local

Boot Server

PC PC

SoundPoint® IP connects physically to a standard office twisted-pair (IEEE 802.3) 10/ 100 megabytes per second Ethernet LAN and sends and receives all data using the

same packet-based technology. Since SoundPoint® IP is a data terminal, digitized audio being just another type of data from the telephone’s perspective, SoundPoint® IP is capable of vastly more than traditional business telephones. As SoundPoint® IP

runs the same protocols as your office personal computer, many innovative applica-

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tions can be developed without resorting to specialized technology. Regardless of SoundPoint® IP’s diverse application potential, it is fundamentally a good office tele-

phone, providing the productivity enhancing features needed today such as multiple call appearances, full-duplex speakerphone, hold, transfer, conference, forward, voice mail compatibility, and contact directory.

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Administrator Guide - SoundPoint® IP Installation and Operation

2 Installation and Operation

This section describes the basic steps that are needed to make your Soundpoint® IP telephone operational.

2.1 Installation Models

SoundPoint® IP supports diverse installation models scaling from stand-alone telephones to large, centrally provisioned systems with thousands of telephones. For any

size system, SoundPoint® IP telephones can be centrally provisioned from a boot server via a system of global and per-telephone configuration files. To augment the

central provisioning model, or as the sole method in smaller systems, SoundPoint telephones may be configured using user interfaces driven from the telephones themselves: both a local setup user interface and a web server-based user interface are available to make configuration changes.

master config file

application binary

config files

dictionary files

user interface resource files

Boot Server

0004f2002999.cfg sip.ld ipmid.cfg sip.cfg phone2999.cfg SoundPointIP-dictionary.xml

Local User Interface

0004f2002999-phone.cfg 0004f2002999-directory.cfg

SoundPoint IP

MAC 00:04:f2:00:29:99

config overrides directory

0004f2002999-boot.log 0004f2002999-app.log

Local

Web Server

event log files

A boot server allows global and per-phone configuration to be managed centrally via text XML-format configuration files that are downloaded by the telephones at boot

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time. The boot server also facilitates automated application upgrades, diagnostics, and a measure of fault tolerance.

The configuration served by the boot server can be augmented by changes made locally on the telephone itself or via the telephone’s built-in web server. If file uploads are permitted, the boot server allows these local changes to be backed up automatically.

Polycom recommends the boot server central provisioning model for installations involving more than a few telephones. The investment required is minimal in terms of time and equipment, and the benefits are significant.

SoundPoint® IP also supports dynamic host configuration protocol (DHCP). When set up, DHCP permits plug-and-play TCP/IP network setup.

2.2 Installation Process

Regardless of whether or not you will be installing a centrally provisioned system, there are two steps required to get your telephones up and running.

1. Basic TCP/IP Network Setup such as IP address and subnet mask. For more information, see 2.2.1 Basic Network Setup on page 4.

2. Application Configuration such as application specific parameters. For more information, see

2.2.2 Application Configuration on page 9.

2.2.1 Basic Network Setup

SoundPoint® IP boots up in two phases:

• Phase 1: bootROM - a generic program designed to load the application.

• Phase 2: application - the SIP telephone application.

Networking starts in Phase 1. The bootROM application uses the network to query the boot server for upgrades or configuration changes, which is an optional process that will happen automatically when properly deployed. The boot server can be on the local LAN or anywhere on the Internet. The bootROM then loads the configured application. The application will restart networking using most of the parameters established by the bootROM (a DHCP query will be performed by the application).

Basic network settings can be changed during Phase 1 using the bootROM’s setup menu. A similar, but more sophisticated menu system is present in the application for changing the same network parameters. For more information, see 2.2.1.2 Local User Interface Setup Menus on page 6.

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2.2.1.1 DHCP or Manual TCP/IP Setup

Basic network settings can be derived from DHCP or entered manually using the telephone’s LCD-based user interface. Polycom recommends using DHCP where possible to eliminate repetitive manual data entry.

The following table shows the manually entered networking parameters that may be overridden by parameters obtained from a DHCP server:

Parameter DHCP Option DHCP

12 3

IP address

subnet mask

IP gateway

boot server address

SNTP server address

SNTP GMT offset

DNS server IP address

alternate DNS server IP

See 2.2.1.2.2

DHCP Menu

on page 7

42 then 4

•- •

•• •

•- •

address

DNS domain

•- •

Configuration File (Phase 2: application only)

 priority when more than one source exists 

Local FLASH

CDP

VLAN ID

a. CDP - Cisco Discovery Protocol b. The VLAN ID may be overridden by a value obtained from a connected Ethernet switch if

the switch supports the Cisco Discovery Protocol (CDP).

-•

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2.2.1.2 Local User Interface Setup Menus

Access to Network Configuration Menu:

Phase 1: bootROM The network configuration menu is accessible during the auto-boot

countdown of the bootROM phase of operation. Press the key to launch the main menu.

Phase 2: application The network configuration menu is accessible from the main menu.

Navigate to Menu>Settings>Network Configuration. This menu is locked by default. Enter the administrator password to unlock. (Factory default password: 456)

SoundPoint® IP network configuration parameters may be edited by means of a main menu and two sub-menus: DHCP Menu and Server Menu.

Use the soft keys, the arrow keys, the Sel/3, and the Del/X keys to make changes.

Parameters that are dimmed are read-only due to the value of other parameters. For example, if the DHCP Client parameter is enabled, the Phone IP Addr and Subnet Mask parameters are dimmed since these are guaranteed to be supplied by the DHCP server (mandatory DHCP parameters) and the statically assigned IP address and subnet mask will never be used in this configuration.

Setup soft

2.2.1.2.1 Main Menu

Configuration parameters that may be edited on the main setup menu are described in the table below:

Name Possible Values

DHCP Client Enabled, Disabled If enabled, DHCP will be used to obtain the

Phone IP Address dotted-decimal IP

Subnet Mask dotted-decimal subnet

IP Gateway dotted-decimal IP

address

mask

address

Description

parameters discussed in 2.2.1.1 DHCP or Manual TCP/IP Setup on page 5.

Telephone’s IP address.

Note: Disabled when DHCP client is enabled.

Telephone’s subnet mask.

Note: Disabled when DHCP client is enabled.

Telephone’s default router.

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Administrator Guide - SoundPoint® IP Installation and Operation

Name Possible Values

SNTP Address dotted-decimal IP

address

domain name string

GMT Offset -12 through +13 Offset of the local time zone from Greenwich

DNS Server dotted-decimal IP

address

DNS Alternate Server dotted-decimal IP

address

DNS Domain domain name string Telephone’s DNS domain.

CDP Enabled, Disabled If enabled, the telephone will attempt to deter-

VLAN ID Null, 0 through 4094 Telephone’s 802.1Q VLAN identifier.

a. A parameter value of “???” indicates that the parameter has not yet been set and saved in the

telephone’s configuration. Any such parameter should have its value set before continuing.

Description

SNTP server from which the telephone will obtain the current time.

Mean Time in half hour increments.

Primary server to which the telephone directs Domain Name System queries.

Secondary server to which the telephone directs Domain Name System queries.

mine its VLAN ID via the CDP.

The DHCP and Server sub-menus may be accessed from the main setup menu.

2.2.1.2.2 DHCP Menu

The DHCP menu is accessible only when the DHCP client is enabled. DHCP configuration parameters are described in the following table:

Name

Timeout 1 through 600 Number of seconds the telephone waits for secondary

Possible Values

Description

DHCP Offer messages before selecting an offer.

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Administrator Guide - SoundPoint® IP Installation and Operation

Possible

Name

Values

Description

Boot Server Option 66

Custom

Static

Custom+Opt.66

Boot Server Option 128 through 254 When the boot server parameter is set to Custom, this

Boot Server Option Type

IP Address, String

Option 66: The telephone will look for option number 66 (string type) in the response received from the DHCP server. The DHCP server must be configured to send the boot server address in option 66.

Custom: The telephone will look for the option number specified by the “Boot Server Option” parameter (below), and the type specified by the “Boot Server Option Type” parameter (below) in the response received from the DHCP server.

Static: The telephone will use the boot server configured via the Server Menu. For more information, see

2.2.1.2.3 Server Menu on page 8.

Custom+Opt.66: The telephone will first use the custom option if present or use Option 66 if the custom option is not present.

parameter specifies the DHCP option number in which the telephone will look for its boot server.

When the Boot Server parameter is set to Custom, this parameter specifies the type of the DHCP option in which the telephone will look for its boot server.

2.2.1.2.3 Server Menu

Name Possible Values Description

Server Type

Server Address dotted-decimal IP address

FTP User

FTP, Trivial FTP

domain name string

any string When the Server Type parameter is set to FTP,

When set to FTP, the telephone will use the File Transfer Protocol (FTP) to obtain configuration and telephone application files from the boot server.

When set to Trivial FTP, the telephone will use the Trivial File Transfer Protocol (TFTP) to obtain configuration and telephone application files from the boot server.

The boot server to use if the DHCP client is disabled, or the DHCP server does not send a boot server option, or the Boot Server parameter is set to Static.

this is the user name used when the telephone logs into the FTP server.

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Administrator Guide - SoundPoint® IP Installation and Operation

Name Possible Values Description

FTP Password

a. Using TFTP will make management of the telephone more difficult. For more information,

see 2.2.2.1.1 FTP vs. TFTP on page 10.

b. The FTP server user name and FTP password should be changed from the default values.

Note that FTP is an insecure protocol and the user chosen should have very few privileges on the server.

any string When the Server Type parameter is set to FTP,

this is the password used when the telephone logs in to the FTP server.

2.2.1.3 Reset to Factory Defaults

The basic network configuration referred to in the preceding sections can be reset to factory defaults. To perform this function, simultaneously press and hold the 4, 6, 8 and * dial pad keys until the password prompt appears. Enter the administrator password to initiate the reset. This will reset the administrator password as well.

2.2.2 Application Configuration

While it is possible to make telephone calls with SoundPoint® IP using its default configuration, most installations will require some basic configuration changes to get things running optimally. These changes can be made using the central boot server model, if a boot server has been set up, or some, but not all changes can be made using the telephone’s internal configuration web server.

Advantages of using a boot server:

1. The centralized repository for application images and configuration files permits application updates and coordinated configuration parameters.

2. Some parameters can only be modified using boot server configuration files.

3. The multilingual feature requires boot server-resident dictionary files.

4. The customized sound effect wave files require a boot server.

5. When file uploads are permitted, the boot server is the repository for:

• boot process and application event log files - very effective when diagnosing system problems

• local configuration changes via the <Ethernet address>-phone.cfg boot server configuration overrides file - the telephone treats the boot server copy as the original when booting

• per-telephone contact directory named <Ethernet address>-directory.cfg

6. The boot server copy of the application images and configuration files can be used to “repair” a damaged telephone configuration in the same way that system repair disks work for PCs.

The following sections discuss the available configuration options.

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2.2.2.1 Centralized Configuration

The SoundPoint® IP application consists of an executable image file (sip.ld) and one or more XML-format configuration files. In the centrally provisioned model, these files are stored on a boot server and cached in the telephone. If the boot server is available at boot time, the telephone will automatically synchronize its configuration cache with the boot server: bootROM image, application executable, and configuration files are all upgraded this way. The telephone requires that a SNTP server be properly configured for this feature to work optimally.

2.2.2.1.1 FTP vs. TFTP

If using a trivial FTP (TFTP) protocol boot server, synchronization with the boot server will be based on file names, not file timestamps, as is the case with FTP. Executable upgrades and changes made to configuration files will not be recognized by the telephone unless their names are changed.

Example:

file.ld  file01.ld (an upgrade to SIP 1.0.1))

ipmid.cfg  ipmid01.cfg

The master configuration file, which references the other files and is always downloaded and parsed by the telephone, will need to change to reflect these name changes.

Polycom does not recommend TFTP boot servers for actively managed systems. File name management is the responsibility of the System Administrator.

2.2.2.1.2 Configuration Files

SoundPoint® IP configuration files consist of master configuration files and application configuration files.

2.2.2.1.2.1 Master Configuration File

Central provisioning requires that an XML-format master configuration file be located in the home directory on the boot server.

Per-telephone Master Configuration File

If per-telephone customization is required (for all applications that require per-telephone customization), the file should be named <Ethernet address>.cfg, where Ether- net address is the Ethernet MAC address of the telephone in question. For A-F hexadecimal digits, use lower case only, for example, 0004f200106c.cfg. The Ethernet address can be viewed using the About soft key during the auto-boot countdown of

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Administrator Guide - SoundPoint® IP Installation and Operation

the bootROM or via the Menu>System Status>General menu in the application. It is also printed on a label on the back of the telephone.

Default Master Configuration File

For systems in which the configuration is identical for all telephones (no per-telephone <Ethernet address>.cfg files), the default master configuration file may be used to set the configuration for all telephones. The file named 000000000000.cfg (<12 zeros>.cfg) is the default master configuration file and it is recommended that one be present on the boot server. If a telephone does not find its own <Ethernet address>.cfg file, it will use this one, and establish a baseline configuration. This file is part of the standard Polycom distribution of configuration files. It should be used as the template for the <Ethernet address>.cfg files.

The default SIP master configuration file, 000000000000.cfg, is shown below:

Example:

< ?xm l version= "1 .0" stand alone ="ye s" ?>

Default Master SIP Configuration File -->

<!-<!--

Edit and rename this file to <Ethernet-address>.cfg for each

phone.

<!-- < APPLICATION APP_FILE_PATH="sip.ld"

-->

$Revision: 1.24 $ $Date: Mar 26 2003 11:59:02 $ -->

CONFIG_FILES="phone1.cfg, sip.cfg, ipmid.cfg" MISC_FILES="" LOG_FILE_DIRECTORY= ""/>

Master configuration files contain four XML attributes:

APP_FILE_PATH The path name of the application executable. Has a maximum length

of 127 characters.

CONFIG_FILES A comma-separated list of configuration files. Has a maximum length

of 511 characters, including commas and white space.

MISC_FILES

LOG_FILE_DIRECTORY An alternative directory to use for log files if required. This is left

a. MISC_FILES is not used for the SIP application, but is used for the MGCP application.

A comma-separated list of other required files.

blank by default.

Note

The order of the configuration files listed in CONFIG_FILES is significant.

• The files are processed in the order listed (left to right).

• The same parameters may be included in more than one file.

• The parameter found first in the list of files will be the one that is effective. This provides a convenient means of overriding the behavior of one or more telephones without altering the baseline configuration files for an entire system.

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Administrator Guide - SoundPoint® IP Installation and Operation

2.2.2.1.2.2 Application Configuration Files

Typically, the files are arranged in the following manner although parameters may be moved around within the files and the file names themselves can be changed as needed.

Per-phone settings  phoneXXXX.cfg

Application settings  sip.cfg

Core settings  ipmid.cfg

Category Description Example

Core Contain parameters that affect the basic operation of the telephone

such as voice codecs, gains, and tones. All telephones in an installation usually share this category of files.

Applicationspecific

User Contain parameters unique to a particular telephone user. Typical

Contain parameters that dictate performance of a particular telephone application.

Typical parameters include the IP address of an application server. All telephones in an installation usually share application-specific files. This file would normally be modified from Polycom templates.

parameters include:

• display name

• unique addresses

Each telephone in an installation usually has its own customized version of user files derived from Polycom templates.

ipmid.cfg

sip.cfg, mgcp.cfg

phone1.cfg

These application configuration files dictate the behavior of the telephone once it is running the executable specified in the master configuration file.

Important

Configuration files should only be modified by a knowledgeable System Administrator. Applying incorrect parameters may render the telephone unusable.

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Administrator Guide - SoundPoint® IP Installation and Operation

2.2.2.1.3 Deploying a SoundPoint® IP boot server

The following table describes the steps required for successful deployment of a SoundPoint® IP boot server. These instructions assume an FTP protocol boot server.

For more information on TFTP boot servers, see 2.2.2.1.1 FTP vs. TFTP on page 10.

Step: Instructions:

1. Set-up FTP Server:

Install FTP server application or locate suitable existing server. Use RFC-compliant FTP servers. The fol-

Note: Typically all SoundPoint

IP telephones are configured with the same FTP account, but the FTP account provides a means of conveniently partitioning the configuration. Give each account an unique home directory on the server and change the configuration on an account-by-

lowing FTP servers have been tested with

Soundpoint

IP and are known to work acceptably:

• Linux: ProFTPd 1.2.2 through 1.2.9 rc2p, ftpdbsd-0.3.3 (Linux port), we-ftpd 2.6.0

• Windows 2000 Server: IIS 5.0, WFTPD 2.03

• Create account and home directory.

account basis.

2. Copy all files: Copy all files from the distribution.zip file to the

SoundPoint

IP home directory. Maintain the same

folder hierarchy.

3. Create per-telephone configuration

files

Obtain a list of telephone Ethernet addresses (barcoded label on underside of telephone).

Create per-telephone phoneXXXX.cfg and <Ethernet address>.cfg files by using the 00000000000.cfg and phone1.cfg files from the distribution as templates.

Edit contents of phoneXXXX.cfg as appropriate. For example, edit the registration parameters.

Edit the CONFIG_FILES attribute of the <Ethernet

address>.cfg files so that it references the appropriate phoneXXXX.cfg file. (Replace the reference to

phone1.cfg with phoneXXXX.cfg.)

4. Edit <application>.cfg: See 4.6.2 SIP Configuration - sip.cfg on page 101, particularly for SIP server address.

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Step: Instructions:

5. Edit ipmid.cfg: See 4.6.1 Core Configuration - ipmid.cfg on page 55.

Most of the default settings are typically adequate, however, if overriding SNTP settings are not available via DHCP, the SNTP GMT offset and (possibly) the SNTP server address will need to be edited for the correct local conditions. Changing the default daylight savings parameters will likely be necessary outside of North American locations.

(Optional) Disable the local web (HTTP) server or alter its signalling port if local security policy dictates.

Change the default location settings:

• user interface language

• time and date format

6. Decide on FTP server security pol-

icy:

7. Reboot telephones after configuring

their boot server via DHCP or statically:

Polycom recommends allowing file uploads to the FTP server where the security environment permits. This allows event log files to be uploaded and changes made by the telephone user to the configuration (via the web server and local user interface) and changes made to the directory to be backed up. For organizational purposes, configuring a separate log file directory is recommended, but not required (see LOG_FILE_DIRECTORY in 2.2.2.1.2.1 Master Configuration File on page 10). File permissions should give the minimum access required, and the account used should have no other rights on the server. The telephone's FTP account needs to be able to add files to which it can write in the log file directory and the root directory. It must also be able to list files in all directories mentioned in the [mac].cfg file. All other files that the telephone needs to read, such as the application executable and the standard configuration files, should be made read-only via file server file permissions.

See 2.2.1 Basic Network Setup on page 4.

To reboot telephones manually, press and hold the following keys simultaneously until a confirmation tone is heard or for about three seconds:

IP 400/500: Volume-, Volume+, Hold, and Messages

IP 600: Volume-, Volume +, Mute, and Messages

Monitor the boot server event log and the uploaded event log files (if permitted):

Ensure that the configuration process completed correctly.

Start making calls!

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a. The FTP account name and password must match those configured in the telephones them-

selves. Defaults are: name: PlcmSpIp, password: PlcmSpIp

b. This step may be omitted if per-telephone configuration is not needed.

2.2.2.2 Local Telephone Configuration

As the only method of modifying telephone configuration or as a distributed method of augmenting a centralized provisioning model, a local telephone-based configuration web server is available, unless disabled via ipmid.cfg. For more information, see

4.6.1.9 Web Server <HTTPD/> on page 82. The telephone’s local user interface also permits certain application settings to be modified, such as ring type or regional settings such as time/date format and language.

Local Web Server Access Point your web browser to http://<phoneIPAddress>/.

Configuration pages are accessible from the menu along the top banner.

The web server will issue an authentication challenge to all pages except for the home page.

Credentials are (case sensitive):

• User Name: Polycom

• Password: The administrator password is used for this.

Local Settings Menu Access Some items in the Settings menu are locked to prevent accidental

changes. To unlock these menus, enter the user or administrator passwords.

The administrator password can be used anywhere that the user password is used.

Factory default passwords are:

• User password: 123

• Administrator password: 456

Passwords:

Network Configuration Requires administrator password.

Authentication Requires user password.

Reset User Settings Requires user password.

Changes made via the web server or local user interface are stored internally as overrides. These overrides take precedence over settings contained in the configuration obtained from the boot server that existed previously within the telephone.

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If the boot server permits uploads, these override setting will be saved in a file called <Ethernet address>-phone.cfg on the boot server.

Important

Local configuration changes will continue to override the boot server-derived configuration until deleted via the Reset User Settings menu selection.

2.2.2.3 SIP Authentication Parameters

There are several pairs of username/password fields in the Authentication menu of the application. These correspond to the SIP digest authentication settings for each registration. Authentication parameters will be used in response to challenges unless you include non-Null parameters in the per-telephone (phoneXXXX.cfg) configuration files. For more information, see 4.6.3.1 Registration <reg/> on page 108. Reg User x should contain the authentication username for the first registration (corresponds to reg.x.auth.userId) and Reg Password x should contain the corresponding password (corresponds to reg.x.auth.password).

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3 Features

This section describes SoundPoint® IP’s many features and corresponding administration points. References are made frequently to 4.6 Configuration Files on page 55.

3.1 Basic Features

3.1.1 Call Log

The telephone maintains a call log. The log:

• contains call information such as remote party identification, time and date, and call duration,

• allows for convenient redialing of previous outgoing calls and for returning incoming calls,

• saves contact information from call log entries in the contact directory.

The call log is stored in volatile memory and is maintained automatically by the telephone. The call log can be cleared manually by the user and will be erased on reboot.

3.1.2 Call Timer

A call timer is provided on the display. A separate call timer is maintained for each distinct call in progress.

3.1.3 Call Waiting

When an incoming call arrives while the user is active on another call, the incoming call is presented to the user visually on the LCD display. A configurable sound effect such as the familiar call-waiting beep will be mixed with the active call audio as well.

3.1.4 Called Party Identification

SoundPoint® IP displays and logs the identity of the remote party specified for outgoing calls. This is the party that the user intends to connect with.

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3.1.5 Calling Party Identification

The telephone displays the caller identity, derived from the network signalling, when an incoming call is presented. For calls from parties for which a directory entry exists, the local name assigned to the directory entry may optionally be substituted.

Central (boot server)

Local

Configuration File: ipmid.cfg

Configuration File: phone1.cfg

Web Server (if enabled)

Local Telephone User Interface

Specify whether or not to use directory name substitution.

• For more information, see 4.6.1.2 User Preferences <user_preferences/> on page 59.

Specify whether or not to use directory name substitution.

• For more information, see 4.6.3.7 User Preferences <user_preferences/> on page 117.

Specify whether or not to use directory name substitution. Navigate to: http://<phoneIPAddress>/coreConf.htm#us

Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

None.

3.1.6 Configurable Feature Keys

All SoundPoint® IP key functions can be changed from the factory defaults, although this is typically not necessary. The scrolling timeout for specific keys can be configured.

Central (boot server)

Local

Configuration File: ipmid.cfg

Web Server (if enabled)

Local Telephone User Interface

Set the key scrolling timeout, key functions, and sub-pointers for each key (usually not necessary).

• For more information, see 4.6.1.13 Keys <keys/> on page 88.

None.

The following diagrams and table show the default SIP key layouts for SoundPoint® IP 500 and IP 600 models.

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SoundPoint IP 500 SIP Key Layout

Directo ries

Conference

Services

Call Lists

Conference

Transfer

Redial

19 20 21

GHI

PQRS

ABC21DEF

ABC

JKL

TUV87WXYZ

TUV

OPER

2627

222324

DEF

MNOJKLGHI

MNO

WXYZ

SoundPoint IP 600 SIP Key Layout

Key ID

Sel

Del

Messages

Do Not Di sturb

Hold

Directo ries

Services

Conference

Transfer

Redial

Hold

ABC21DEF

ABC

19 20 21

JKL

GHI

TUV87WXYZ

PQRS

TUV

PQRS

WXYZ

OPER

DEF

MNOJKLGHI

MNO

25262728

222324

131415

Key ID

Sel

Del

Messages

Do Not Di sturb

Hold

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Key ID IP 500 Function IP 600 Function

1 ArrowUp ArrowUp

2 ArrowLeft ArrowLeft

3 Select ArrowDown

4 ArrowRight ArrowRight

5 ArrowDown Select

6 Delete Delete

7 Menu Menu

8 Messages Messages

9 DoNotDisturb DoNotDisturb

10 Hold MicMute

11 VolUp VolUp

12 VolDown VolDown

13 DialpadPound DialpadPound

14 Dialpad0 Dialpad0

15 DialpadStar DialpadStar

16 Dialpad9 Dialpad9

17 Dialpad8 Dialpad8

18 Dialpad7 Dialpad7

19 Dialpad4 Dialpad4

20 Dialpad5 Dialpad5

21 Dialpad6 Dialpad6

22 Dialpad3 Dialpad3

23 Dialpad2 Dialpad2

24 Dialpad1 Dialpad1

25 SoftKey4 SoftKey4

26 SoftKey3 SoftKey3

27 SoftKey2 SoftKey2

28 SoftKey1 SoftKey1

29 Conference Services

30 CallHistory Directories

31 Services Line6

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Key ID IP 500 Function IP 600 Function

32 Directories Conference

33 Line3 Line2

34 Line2 Line1

35 Line1 Line3

36 Redial Redial

37 Transfer Transfer

38 Headset Headset

39 MicMute Handsfree

40 Handsfree Hold

41 n/a Line4

42 n/a Line5

3.1.7 Connected Party Identification

Where possible, SoundPoint® IP displays and logs the identity of the remote party to which the user has connected. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion.

3.1.8 Context Sensitive Volume Control

The volume of user interface sound effects, such as the ringer, and the receive volume of call audio is adjustable. While transmit levels are fixed according to the TIA/EIA810-A standard, receive volume is adjustable. If using the default configuration parameters, the receive handset/headset volume resets to nominal after each call to comply with regulatory requirements. See 4.6.1.6.2 Volume Persistence <volume/> on page 70.

3.1.9 Customizable Audio Sound Effects

SoundPoint® IP audio sound effects used for incoming call alerting and other indications are customizable. Sound effects can be composed of patterns of synthesized tones or sample audio files. The default sample audio files may be replaced with alternates in .wav file format. Supported .wav formats include:

• mono G.711 (13-bit dynamic range, 8-khz sample rate),

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• mono L16/1600 (16-bit dynamic range, 16-kHz sample rate).

Note

The alternate sampled audio sound effect files must be present on the boot server or the Internet for downloading at boot time.

Configuration File: ipmid.cfg

Specify patterns used for sound effects and the individual tones or sampled audio files used within them.

Central (boot server)

Web Server (if enabled)

Local

Local Phone User Interface

For more information, see:

• 4.6.1.1.3 Call Progress Tones <callProgTones> on page 58

• 4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> on page 62

• 4.6.1.5 Sound Effects <sound_effects/> on page 63

Specify sampled audio wave files to replace the built-in defaults. Navigate to:

http://<phoneIPAddress>/coreConf.htm#sa

Changes are saved to local flash and backed up to <Ethernet address>phone-.cfg on the boot server and will permanently override global settings unless deleted via the Reset User Set tings menu selection.

None.

3.1.10 Message Waiting Indication

SoundPoint® IP will flash a message-waiting indicator LED when instant messages are waiting, and it can be configured to do so when voice messages are waiting.

3.1.11 Distinctive Incoming Call Treatment

SoundPoint® IP can automatically apply distinctive treatment to calls containing specific attributes. The distinctive treatment that can be applied includes customizable alerting sound effects and automatic call diversion or rejection. Call attributes that can trigger distinctive treatment include the calling party name or SIP contact (number or URL format).

Administration: Distinctive Incoming Call Treatment

For more information, see 3.1.16 Local Contact Directory on page 25.

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3.1.12 Distinctive Ringing

There are three aspects to Distinctive Ringing:

1. The user can select the ring type for each line. There are many different ring patterns to choose from.

2. The ring type for specific callers can be assigned in the contact directory. For more information, see page 22. This feature has higher priority than Item 1.

3. The SIP Alert-Info field can be used to map calls to specific ring types. This feature has higher priority than Items 1 and 2.

3.1.11 Distinctive Incoming Call Treatment on

Central (boot server)

Local

Configuration file: sip.cfg

XML File: <Ethernet address>-directory.xml

Web Server (if enabled)

Local Phone User Interface

Specify the mapping of Alert-Info strings to ring types.

• For more information, see 4.6.2.1.3.2 Alert Infor-

This file can be created manually using an XML editor.

• For more information, see 3.1.16.1 Local Contact

None.

The user can edit the ring types selected for each line under the Settings menu. The user can also edit the directory contents.

Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. These changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

3.1.13 Distinctive Call Waiting

mation <alertInfo/> on page 104.

Directory File Format on page 26.

The SIP Alert-Info field can be used to map calls to distinct call waiting types, currently limited to two styles.

Central (boot server)

Local

Configuration file: sip.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the mapping of Alert-Info strings to call waiting types.

• For more information, see 4.6.2.1.3.2 Alert Information <alertInfo/> on page 104.

None.

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3.1.14 Do-Not-Disturb

A do-not-disturb feature is available to temporarily stop all incoming call alerting. Calls can optionally be treated as though the telephone is busy while Do-Not-Disturb (DND) is enabled. Incoming calls received while DND is enabled are logged as missed.

Central (boot server)

Local

Configuration file: ipmid.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Specify whether or not DND results in incoming calls being given busy treatment.

• For more information, see 4.6.1.10 Call Handling Configuration <call/> on page 83.

Specify whether DND is treated as a per-registration feature or a global feature on the telephone.

• For more information, see 4.6.3.2.1 Do Not Disturb <donotdisturb/> on page 110.

None.

User DND changes are recorded in flash memory and backed up in the local overrides file of <Ethernet address>-phone.cfg on the boot server if available.

3.1.15 Handset, Headset, and Speakerphone

SoundPoint® IP comes standard with a handset and a full-duplex speakerphone. A dedicated connector is provided for a headset (not supplied). The telephone provides dedicated keys for convenient selection of either the speakerphone or headset.

Configuration file: ipmid.cfg

Central (boot server)

Local

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Enable or disable persistent headset mode.

• For more information, see 4.6.1.2 User Preferences <user_preferences/> on page 59.

Enable or disable persistent headset mode.

• For more information, see 4.6.3.7 User Preferences <user_preferences/> on page 117.

None.

Enable or disable persistent headset mode via the Settings menu. Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu.

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3.1.16 Local Contact Directory

SoundPoint® IP maintains a local contact directory. The directory can be downloaded from the boot server and edited locally. Contact information from previous calls may be easily added to the directory for convenient future access. The directory is the central database for several other features including speed-dial, distinctive incoming call treatment, presence, and instant messaging.

Central (boot server)

Configuration file: ipmid.cfg

XML file: 000000000000-directory.xml

XML file: <Ethernet address>-directory.xml

Set whether the directory uses volatile storage on the telephone (required on the IP 500 platform for directories greater than 25 entries).

• For more information, see 4.6.1.11 Directory <directory/> on page 84.

A sample file named 000000000000-directory~.xml (Note extra “~” in the file name) is included with the application file distribution. This file can be used as a template for the per-telephone <Ethernet address>directory.xml directories (edit contents then rename to <Ethernet address>-directory.xml). It also can be used to seed new telephones with an initial directory (edit contents than remove “~” from file name). Telephones without a local directory, such as new units from the factory, will download the 00000000000-directory.xml directory and base their initial directory on it. These files should be edited with an XML editor.

• For information on file format, see 3.1.16.1 Local Contact Directory File Format on page 26.

This file can be created manually using an XML editor.

• For information on file format, see 3.1.16.1 Local Contact Directory File Format on page 26.

Local

Web Server (if enabled)

Local Phone User Interface

None.

The user can edit the directory contents at will. Changes will be stored in the telephone’s flash file system and backed up to the boot server copy of <Ethernet address>-directory.xml if this is configured. When the telephone boots, the boot server copy of the directory, if present, will overwrite the local copy.

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3.1.16.1 Local Contact Directory File Format

An example local contact directory is shown. Look to the table for an explanation of each element.

Local Contact Directory File example:

<?xml version="1.0" encoding="UTF-8" standalone="yes" ?> <directory>

<item_list>

<item>

</item>

• • •

<item>

<ln>Smith</ln> <fn>Bill</fn> <ct>1003</ct> <sd>3</sd> <rt>3</rt> <dc /> <ad>0</ad> <ar>0</ar> <bw>0</bw> <bb>0</bb>

</item>

</item_list>

</directory>

Element Permitted Values Interpretation

fn UTF-8 encoded string of up to

40 bytes

ln UTF-8 encoded string of up to

40 bytes

first name

last name

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Element Permitted Values Interpretation

ct UTF-8 encoded string contain-

ing digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL

sd Null, 1 to 40 speed-dial index

rt Null, 1 to 21 ring type

dc UTF-8 encoded string contain-

ing digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL

ad 0,1 auto divert

contact

Cannot be Null or duplicated; is used by the telephone to address a remote party in the same way that a string of digits or a SIP URL are dialed manually by the user. This element is also used to associate incoming callers with a particular directory entry.

Associates a particular entry with a speed dial bin for one-touch dialing or dialing from the speed dial menu.

When incoming calls can be associated with a directory entry by matching the address fields, this field is used to specify ring type to be used.

divert contact

The forward-to address for the autodivert feature.

ar 0,1

bw 0,1 buddywatching

bb 0,1 buddyblock

a. In some cases, this will be less than 40 characters due to UTF-8’s variable length encoding. b. If auto-divert is also enabled, it has precedence over auto-reject.

3.1.17 Local Digit Map

SoundPoint® IP has a local digit map feature to automate the setup phase of numberonly calls. When properly configured, this feature eliminates the need for using the

If 1, automatically diverts callers that match the directory entry to the address specified in divertcontact.

auto-reject

If 1, automatically rejects callers that match the directory entry.

If 1, add this contact to the list of watched telephones.

If 1, block this contact from watching this telephone.

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Send soft key when making outgoing calls. Instead, as soon as a digit pattern matching

the digit map is found, the call setup process will complete automatically. The Sound-

Point

IP digit map feature is similar to the digit map feature of the Media Gateway

Control Protocol (MGCP) and the configuration syntax is the same as that specified in

2.1.5 of RFC 3435. The telephone behavior when the user dials digits that do not match the digit map is configurable. It is also possible to strip a trailing # from the digits sent.

Central (boot server)

Local

Configuration file: sip.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Specify impossible match behavior, trailing # behavior, digit map matching strings, and time out value.

• For more information, see 4.6.2.1.4 Dial Plan <dialplan/> on page 106.

Specify per-registration impossible match behavior, trailing # behavior, digit map matching strings, and time out values that override those in sip.cfg.

• For more information, see 4.6.3.4 Dial Plan <dialplan/> on page 113.

Specify digit map matching strings and time out value. Navigate to: http://<phoneIPAddress>/appConf.htm#ls

None.

3.1.18 Microphone Mute

A microphone mute feature is provided. When activated, visual feedback is provided. This is a local function and cannot be overridden by the network.

3.1.19 Multiple Call Appearances

SoundPoint® IP supports multiple concurrent calls. The hold feature can be used to pause activity on one call and switch to another call. When active on one call, an additional incoming call is presented using the familiar “call waiting” style. Soft keys with call disposition options are presented to the user. The current user interface is limited to two concurrent calls per registration (line). This is an artificial limit and will be expanded in the future.

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3.1.20 Shared Call Appearances

SoundPoint® IP allows calls and lines on multiple telephones to be logically related to each other. A call that is active on one telephone will be presented visually to telephones which share that appearance. Mutual exclusion features emulate traditional PBX or key system privacy for shared calls. Incoming calls can be presented to multiple telephones simultaneously. This feature is dependent on support from a SIP server which binds the appearances together logically and looks after the necessary state notifications and performs an access control function. For more information, see 5.2.5 Shared Call Appearance Signaling on page 127.

Central (boot server)

Configuration file: ipmid.cfg

Configuration file: sip.cfg

Configuration file: phone1.cfg

Specify whether diversion should be disabled on shared lines.

• For more information, see 4.6.1.10 Call Handling Configuration <call/> on page 83.

Specify line-seize subscription period.

• For more information, see 4.6.2.1.2 Server <server/> on page 102.

Specify standard or non-standard behavior for processing line-seize subscription for mutual exclusion feature.

• For more information, see 4.6.2.1.3.4 Special Events <specialEvent/> on page 105.

Specify per-registration line type (private or shared) and line-seize subscription period if using per-registration servers. A shared line will subscribe to a server providing call state information.

• For more information, see 4.6.3.1 Registration <reg/> on page 108.

Specify per-registration whether diversion should be disabled on shared lines.

• For more information, see 4.6.3.3 Diversion <divert/> on page 111.

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Local

Web Server (if enabled)

Local Phone User Interface

Specify line-seize subscription period. Navigate to:

http://<phoneIPAddress>/appConf.htm#se

Specify standard or non-standard behavior for processing line-seize subscription for mutual exclusion feature. Navigate to:

http://<phoneIPAddress>/appConf.htm#ls

Specify per-registration line type (private or shared) and line-seize subscription period if using per-registration servers, and whether diversion should be disabled on shared lines. Navigate to:

http://<phoneIPAddress>/reg.htm

Changes are saved to local flash and backed up to <Ether- net address>-phone.cfg on the boot server. They will permanently override global settings unless deleted via the Reset User Settings menu selection.

None.

3.1.21 Customizable Fonts and Indicators

The SoundPoint® IP user interface can be customized by changing the fonts and graphic icons used on the display and the LED indicator patterns. Pre-existing fonts embedded in the software can be overwritten or new fonts can be downloaded. The bitmaps and bitmap animations used for graphic icons on the display can be changed and repositioned. LED flashing sequences and colors can be changed.

Configuration File: ipmid.cfg

Central (boot server)

Specify fonts to overwrite existing ones or specify new fonts.

• For more information, see 4.6.1.12 Fonts <font/> on page 85.

Specify which bitmaps to use.

• For more information, see 4.6.1.14 Bitmaps <bitmaps/> on page 91.

Specify how to create animations and LED indicator patterns.

• For more information, see 4.6.1.15 Indicators <indicators/ > on page 92.

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Web Server (if enabled)

Local

Local Phone User Interface

None.

3.1.22 Soft Key-Driven User Interface

SoundPoint® IP user interface software makes extensive use of intuitive, context-sensitive soft key menus.

3.1.23 Speed Dial

Entries in the local directory can be linked to the speed dial system. The speed dial system allows calls to be placed quickly from dedicated keys as well as from a speed dial menu.

XML file: <Ethernet address>-directory.xml

Central (boot server)

The <sd>x</sd> element in the <Ethernet address>-directory.xml file links a directory

entry to a speed dial resource within the telephone. Speed dial entries are mapped automatically to unused line keys and are available for selection within the speed dial menu. (Press the up-arrow key from the idle display to jump to SpeedDial).

Local

• For more information, see 3.1.16.1 Local Contact Directory File Format on page 26.

Web Server (if enabled) None.

Local Phone User Interface The user can edit the directory contents. The

Speed Dial Index field is used to link directory entries to speed dial operations.

Changes will be stored in the telephone’s flash file system and backed up to the boot server copy of <Ethernet address>-directory.xml if this is configured. When the telephone boots, the boot server copy of the directory, if present, will overwrite the local copy.

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3.1.24 Time and Date Display

SoundPoint® IP maintains a local clock and calendar. Time and date are displayed in certain operating modes such as when the telephone is idle. The clock and calendar must be synchronized to a remote SNTP timeserver.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Set the time and date display formats.

• For more information, see 4.6.1.1.2 Date and Time <datetime/> on page 58.

Set the basic SNTP settings.

• For more information, see 4.6.1.8.2 Time Synchronization <SNTP/> on page 80.

Set daylight savings parameters.

• For more information, see 4.6.1.8.2 Time Synchronization <SNTP/> on page 80.

Set the basic SNTP and daylight savings settings.

Navigate to: http://<phoneIPAddress>/coreConf.htm#ti

Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. They will permanently override global settings unless deleted via the Reset User Settings menu selection.

The basic SNTP settings can be made in the Network Configuration menu.

• For more information, see 2.2.1.1 DHCP or Manual TCP/IP Setup on page 5.

The user can edit the time and date format under the Settings menu.

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3.2 Call Management Features

3.2.1 Automatic Off-hook Call Placement

SoundPoint® IP supports an optional automatic off-hook call placement feature for each registration.

Central (boot server)

Local

3.2.2 Call Hold

Call hold is a fundamental SoundPoint® IP feature. The purpose of hold is to pause activity on one call so that the user may use the telephone for another task, such as to make or receive another call. Network signalling is employed to request that the remote party stop sending media and to inform them that they are being held. A configurable local hold reminder feature can be used to remind the user that they have placed calls on hold.

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Specify which registrations have the feature and what contact to call when going off hook.

• For more information, see 4.6.3.2.2 Automatic Offhook Call Placement <autoOffHook/> on page 110.

None.

Central (boot server)

Local

Configuration file: sip.cfg

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or a=inactive) outgoing hold signalling is used.

• For more information, see 4.6.2.1.3 SIP <SIP/> on page 103.

Specify local hold reminder options.

• For more information, see 4.6.1.10.1 Hold, Local Reminder <localReminder/> on page 84.

None.

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3.2.3 Call Transfer

Call transfer enables the user (User A or transferring user) to transform an existing call with User B (primary call) into a new call between User B and a third user C (transferred-to user) selected by User A. The telephone offers both blind transfers (call is transferred to User C without first consulting privately with User C) and transfers with consultation (call is transferred to User C after first consulting privately with User C). While in the connected state of a consultation call, User A may elect to initiate a threeway conference among the three parties rather than transfer the primary call to

User C1.

3.2.4 Three-Way Conference, Local

SoundPoint® IP can conference together the local user with the remote parties of two independent calls. The telephone’s local audio processing resources are used for the audio bridging and there is no dependency on network signaling.

3.2.5 Call Diversion (Call Forward)

SoundPoint® IP provides a flexible call diversion feature to divert (forward) calls to another destination. Call diversion can be applied automatically to all calls, only when the telephone is busy, or after an extended period of alerting. The user can elect to manually divert calls while they are in the alerting state to a predefined or manually specified destination. The call diversion feature works in conjunction with the distinctive incoming call treatment feature. The user’s ability to originate calls is unaffected by all call diversion options. Each registration (line) has its own diversion properties.

Configuration file: phone1.cfg

Central (boot server)

Set all call diversion settings including a global forward-to contact and individual settings for call forward all, call forward busy, call forward no-answer, and call forward do-notdisturb.

• For more information, see 4.6.3.3 Diversion <divert/> on page 111.

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Local

Web Server (if enabled)

Local Phone User Interface

Set all call diversion settings.

Navigate to: http://<phoneIPAddress>/reg.htm

The user can set the call forward all setting from idle (enabled/disable and specify the forward-to contact) as well as divert callers while the call is alerting.

3.3 Audio Processing Features

SoundPoint® IP uses proprietary state-of-the-art digital signal processing (DSP) technology to provide the best audio experience in a desktop telephone.

3.3.1 Low-Delay Audio Packet Transmission

SoundPoint® IP is designed to minimize latency for audio packet transmission.

3.3.2 Jitter Buffer and Packet Error Concealment

SoundPoint® IP employs a high-performance jitter buffer and packet error concealment system designed to mitigate packet inter-arrival jitter and out-of-order or lost (lost or excessively delayed by the network) packets. The jitter buffer is adaptive and configurable for different network environments. When packets are lost, a concealment algorithm minimizes the resulting negative audio consequences.

Central (boot server)

Configuration file: ipmid.cfg

Set the jitter buffer tuning parameters including minimum and maximum size and shrink aggression.

• For more information, see 4.6.1.6.1.2 Codec Profiles <profiles/> on page 69.

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Web Server (if enabled)

Local

Local Phone User Interface

Set the jitter buffer tuning parameters including minimum and maximum size and shrink aggression.

Navigate to: http://<phoneIPAddress>/coreConf.htm#au

None.

3.3.3 Local Conference Mixing

SoundPoint® IP’s audio processing subsystem contains a flexible three-party conferencing capability. This feature can be used to set up local three-party conferences where no external protocol signaling is involved.

3.3.4 Voice Activity Detection (VAD)

The purpose of VAD is to conserve network bandwidth by detecting periods of relative “silence” in the transmit data path and replacing that silence efficiently with special packets that indicate silence is occurring. For those compression algorithms without

an inherent VAD function, such as G.711, SoundPoint® IP is compatible with the comprehensive codec-independent comfort noise transmission algorithm specified in RFC

3389. This algorithm is derived from G.711 Appendix II, which defines a comfort noise (CN) payload format (or bit-stream) for G.711 use in packet-based, multimedia

communication systems. SoundPoint® IP generates CN packets (also known as Silence Insertion Descriptor (SID) frames) and also decodes CN packets, efficiently regenerating a facsimile of the background noise at the remote end.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Enable or disable VAD and set the detection threshold.

• For more information, see 4.6.1.6.10 Voice Activity Detection <VAD/> on page 77.

None.

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3.3.5 DTMF Tone Generation

SoundPoint® IP generates DTMF tones in response to user dialing on the dial pad. SoundPoint® IP transmits the DTMF tones in the RTP streams of connected calls. The

telephone can encode the DTMF tones using the active voice codec or using RFC 2833 compatible encoding. The coding format decision is based on the capabilities of the remote endpoint.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Set the DTMF tone levels, autodialing on and off times, and other parameters.

• For more information, see 4.6.1.3.1 Dual Tone MultiFrequency <DTMF/> on page 60.

None.

3.3.6 DTMF Event RTP Payload

SoundPoint® IP is compatible with RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals. RFC 2833 describes a standard RTP-compatible

technique for conveying DTMF dialing and other telephony events over an RTP media stream. SoundPoint® IP generates RFC 2833 (DTMF only) events but does not regen-

erate, nor otherwise use, DTMF events received from the remote end of the call.

Central (boot server)

Configuration file: ipmid.cfg

Enable or disable RFC 2833 support in SDP offers and specify the payload value to use in SDP offers.

• For more information, see 4.6.1.3.1 Dual Tone MultiFrequency <DTMF/> on page 60.

Web Server (if enabled)

Local

Local Phone User Interface

None.

3.3.7 Acoustic Echo Cancellation (AEC)

SoundPoint® IP employs advanced acoustic echo cancellation for hands-free operation. Both linear and non-linear techniques are employed to aggressively reduce echo yet provide for natural full-duplex communication patterns.

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3.3.8 Audio Codecs

The following table summarizes SoundPoint® IP’s audio codec support:

Effective audio bandwidth

Algorithm MIME Type Ref. Bit Rate

Sample Rate

Frame Size

G.711µ-law PMCU RFC

1890

G.711a-law PCMA RFC

1890

G.729AB G729 RFC

1890

SID CN RFC

3389

RFC 2833 telephone-

event

Configuration file: ipmid.cfg

Central (boot server)

Web Server (if enabled)

RFC 2833

64 Kbps 8 Ksps 10ms - 80ms 3.5KHz

8 Kbps 8 Ksps 10ms - 80ms 3.5KHz

N/A N/A N/A N/A

Specify codec priority, preferred payload sizes, and jitter buffer tuning parameters.

For more information, see

• 4.6.1.6.1.1 Codec Preferences <preferences/> on page 69, and

• 4.6.1.6.1.2 Codec Profiles <profiles/> on page 69.

Specify codec priority, preferred payload sizes, and jitter buffer tuning parameters.

Navigate to: http://<phoneIPAddress>/coreConf.htm#au

Changes are saved to local flash and backed up to <Ethernet

Local

Local Phone User Interface

address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

None.

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3.3.9 Background Noise Suppression (BNS)

This feature, designed primarily for handsfree operation, reduces background noise to enhance communication in noisy environments.

Configuration file:

Central (boot server)

Local

ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

3.3.10 Comfort Noise Fill

Comfort noise fill is designed to help provide a consistent noise level to the remote user of a handsfree call. Fluctuations in perceived background noise levels are an undesirable side effect of the non-linear component of most AEC systems. This feature uses noise synthesis techniques to smooth out the noise level in the direction toward the remote user, providing a more natural call experience.

Specify whether BNS is enabled for each of handset, headset, and handsfree and the degree of suppression during periods of speech and during silence periods.

• For more information, see 4.6.1.6.6 Background Noise Suppression <NS/> on page 75.

None.

3.3.11 Automatic Gain Control (AGC)

This feature, applicable to handsfree operation, is used to boost the transmit gain of the

local talker in certain circumstances.2 This increases the effective user-telephone radius and helps with the intelligibility of soft-talkers.

Configuration file:

Central (boot server)

Local

2. AGC support will be available in a subsequent release.

ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify whether AGC is enabled for each of handset, headset, and handsfree. AGC is recommended only for handsfree operation.

• For more information, see 4.6.1.6.7 Automatic Gain Control <AGC/> on page 76.

None.

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3.4 Presence and Instant Messaging Features

SoundPoint® IP contains both presence and instant messaging features. These presence and instant messaging features are compatible with Microsoft® Windows® and

MSN saging features are integrated with the contact directory features, using its contact database.

3.4.1 Presence

SoundPoint® IP’s presence feature allows the telephone to monitor the status of other users/devices and allows other users to monitor it. The status of monitored users is displayed visually and is updated in real time. The user can block others from moni-

toring their telephone and is notified when a change in monitored status occurs3. Telephone status changes are broadcast automatically to monitoring telephones when the user engages in calls or invokes do-not-disturb. The user can also manually specify a state to convey, overriding, and perhaps masking, the automatic behavior.

Messenger versions 4.6 and 4.7. SoundPoint® IP’s presence and instant mes-

XML file: <Ethernet

Central (boot server)

Local

address>-directory.xml

Web Server (if enabled)

Local Phone User Interface

3.4.2 Instant Messaging

The <bw>0</bw> (buddy watching) and <bb>0</bb> (buddy blocking) elements in the <Ethernet address>directory.xml file dictate the presence aspects of directory entries.

• For more information, see 3.1.16.1 Local Contact Directory File Format on page 26.

None.

The user can edit the directory contents. The Watch Buddy and Block Buddy fields control the buddy behavior of contacts.

SoundPoint® IP supports sending and receiving instant text messages. The user is alerted to incoming messages visually and audibly. The user can choose to view the

3. Notification when a change in monitored status occurs will be available in a subsequent release.

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messages immediately or when it is convenient. For sending messages, the user can choose to either select a message from a pre-set list of short messages, or an alphanumeric text entry mode allows the typing of custom messages using the dial pad. Message sending can be initiated by replying to an incoming message or by initiating a new dialog. The destination for new dialog messages can be entered manually or selected from the contact directory, the preferred method.

3.5 Localization Features

3.5.1 Multilingual User Interface

SoundPoint® IP has a multilingual user interface. The System Administrator or the user can choose the language. Support for major western European languages is included and additional languages can be easily added. Support for Asian languages (Chinese, Japanese, and Korean) is also included but will render only on the Sound-

Point® IP 600’s higher resolution display.

Basic character support available on all SoundPoint® IP platforms include the following Unicode character ranges:

Name Range

C0 Controls and Basic Latin U+0000 - U+007F

C1 Controls and Latin-1 Supplement U+0080 - U+00FF

Cyrillic (partial) U+0400 - U+045F

Extended character support available on SoundPoint

IP 600 platform include the fol-

lowing Unicode character ranges4.

Name Range

CJK Symbols and Punctuation U+3000 - U+303F

Hiragana U+3040 - U+309F

Katakana U+30A0 - U+30FF

Bopomofo U+3100 - U+312F

4. Within a Unicode range, some characters may not be supported due to their infrequent usage.

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Name Range

Hangul Compatibility Jamo U+3130 - U+318F

Bopomofo Extended U+31A0 - U+31BF

Enclosed CJK Letters and Months U+3200 - U+327F

CJK Compatibility U+3300 - U+33FF

CJK Unified Ideographs U+4E00 - U+9FFF

Hangul Syllables U+AC00 - U+D7A3

CJK Compatibility Ideographs U+F900 - U+FAFF

CJK Half-width forms U+FF00 - U+FFFF

Note

The multilingual feature relies on dictionary files resident on the boot server. The dictionary files are downloaded from the boot server whenever the language is changed or at boot time when a language other than the internal US English language has been configured. If the dictionary files are inaccessible, the language will revert to the internal language.

Note

Currently, the multilingual feature is only available in the SIP application. At this time, the

SoundPoint

IP bootROM application is English only.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the boot-up language and the selection of language choices to be made available to the user.

For more information, see:

• 4.6.1.1.1 Multilingual <multilingual/> on page 56, and

• 4.6.1.1.1.1 Adding New Languages on page 57.

None.

The user can select their preferred language under the Settings menu. Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

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3.5.2 Downloadable Fonts

New fonts can be loaded onto the telephone. For more information, see 4.6.1.12 Fonts <font/> on page 85.

3.5.3 Synthesized Call Progress Tones

In order to emulate the familiar and efficient audible call progress feedback generated by the PSTN and traditional PBX equipment, SoundPoint® IP synthesizes call

progress tones during the life cycle of a call. These call progress tones are easily configurable for compatibility with worldwide telephony standards or local preferences.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the basic tone frequencies, levels, and basic repetitive cadences.

• For more information, see 4.6.1.3.2 Chord Sets <chord_sets/> on page 61.

Specify downloaded sampled audio files for advanced call progress tones.

• For more information, see 4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> on page 62.

Specify patterns.

For more information, see:

• 4.6.1.5.1 Patterns <patterns/> on page 64, and

• 4.6.1.5.1.1 Call Progress Patterns on page 65.

None.

3.6 Advanced Server Features

3.6.1 Voicemail Integration

SoundPoint® IP is compatible with voicemail servers. The subscribe contact and callback mode can be configured per user/registration on the telephone. The telephone can be configured with a SIP URL to be called automatically by the telephone when the user elects to retrieve messages. Voicemail access can be configured to be via a

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single key press if only one registration has voicemail set up. A message-waiting signal from a voicemail server will trigger the message-waiting indicator to flash.

Central (boot server)

Local

Configuration file: ipmid.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

For one-touch voicemail access, enable the “one-touch voicemail” user preference.

• For more information, see 4.6.3.7 User Preferences <user_preferences/> on page 117.

For one-touch voicemail access, choose to bypass instant messages to remove the step of selecting between instant messages and voicemail after pressing the Messages button (instant messages are still accessible from the Main Menu).

On a per-registration basis, specify a subscribe contact for solicited NOTIFY applications, a callback mode (self callback or another contact), and the contact to call when the user accesses voicemail.

• For more information, see 4.6.3.5 Messaging <msg/> on page 115.

For one-touch voicemail access, enable the “one-touch voicemail” user preference and choose to bypass instant messages to remove the step of selecting between instant messages and voicemail after pressing the Messages button (instant messages are still accessible from the Main Menu).

Navigate to: http://<phoneIPAddress>/coreConf.htm

On a per-registration basis, specify a subscribe contact for solicited NOTIFY applications, a callback mode (self callback or another contact) to call when the user accesses voicemail.

Navigate to: http://<phoneIPAddress>/reg.htm

Local Phone User Interface

None.

3.6.2 Multiple Registrations

SoundPoint® IP supports multiple registrations per telephone. Each registration is mapped to the familiar concept of a telephone line. The user can select which line to

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use for outgoing calls or which registration to use when initiating new instant message dialogs.

Central (boot server)

Local

Configuration file: sip.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

Specify the local SIP signaling port and an array of SIP servers to register to. For each server specify the registration period and the signaling failure behavior.

• For more information, see 4.6.2.1.1 Local <local/> on page 101 and 4.6.2.1.2 Server <server/> on page 102.

For up to six registrations, specify a display name, a SIP address, an optional display label, an authentication user ID and password, and an optional array of registration servers. The authentication user ID and password are optional and for security reasons can be omitted from the configuration files. The local flash parameters will be used instead. The optional array of servers and their associated parameters will override the servers specified in sip.cfg if non-Null.

• For more information, see 4.6.3.1 Registration <reg/> on page 108.

Specify the local SIP signaling port and an array of SIP servers to register to.

Navigate to: http://<phoneIPAddress>/appConf.htm

For up to six registrations, specify a display name, a SIP address, an optional display label, an authentication user ID and password, and an optional array of registration servers. The authentication user ID and password are optional and for security reasons can be omitted from the configuration files. The local flash parameters will be used instead. The optional array of servers will override the servers specified in sip.cfg in non-Null. This will also override the servers on the appConf.htm web page.

Local Phone User Interface

3.6.3 Server Redundancy

SoundPoint® IP can be configured with multiple SIP servers, one primary and one or more backup. SoundPoint

server fails. Backup server configuration can be static or use advanced DNS methods.

IP will switch to a backup server when the current primary

Navigate to: http://<phoneIPAddress>/reg.htm

None

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3.6.3.1 DNS SIP Server Name Resolution

If a DNS name is given for a proxy/registrar address, the IP address(es) associated with that name will be discovered as specified in RFC 3263 - Locating SIP Servers. If a port is given, the only lookup will be an A record. If no port is given, NAPTR and SRV records will be tried, before falling back on A records if NAPTR and SRV records return no results. If no port is given, and none is found through DNS, 5060 will be used.

See http://www.ietf.org/rfc/rfc3263.txt for an example.

3.7 Accessory Internet Features

3.7.1 Services Browser

The Services Browser is not available with this release. In the future, the Services key will be used to launch the services browser.

3.8 Security Features

3.8.1 Local User and Administrator Privilege Levels

Several local settings menus are protected with two privilege levels, user and administrator, each with its own password. The telephone will prompt for either the user or administrator password before granting access to the various menu options. When the user password is requested, the administrator password will also work. The web server is protected by the administrator password.

Central (boot server)

Configuration file: ipmid.cfg

Specify the minimum lengths for the user and administrator passwords.

• For more information, see 4.6.1.17.1 Password Lengths <pwd/><length> on page 97.

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Local

Web Server (if enabled)

Local Phone User Interface

None.

The user and administrator passwords can be changed under the Settings menu. Passwords can consist of ASCII characters 32-127 (0x20-0x7F) only.

Changes are saved to local flash but are not backed up to <Ethernet address>-phone.cfg on the boot server for security reasons.

3.8.2 Incoming Signaling Validation

Three optional levels of security are provided for validating incoming network signaling:

• source IP address validation

• digest authentication

• both

Central (boot server)

Configuration File: sip.cfg

Specify the type of validation to perform on a request-byrequest basis, appropriate to specific event types in some cases.

• For more information, see 4.6.2.1.3.3 Request Validation <requestValidation/> on page 104.

Local

Web Server (if enabled)

Local Phone User Interface

None.

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4 Optimizing SoundPoint

4.1 Ethernet Switch

SoundPoint® IP contains two Ethernet ports, labeled LAN and PC, and an embedded Ethernet switch that runs at full line-rate. The Ethernet switch allows a personal computer and other Ethernet devices to connect to the office LAN by daisy chaining through the telephone, eliminating the need for a stand-alone hub. The

SoundPoint phone. SoundPoint® IP can be powered via a local AC power adapter or can be line-

powered (power supplied via the signaling or idle pairs of the LAN Ethernet cable). Line powering typically requires that the telephone plug directly into a dedicated LAN

jack. Devices that do not require LAN power can then plug into the SoundPoint® IP PC Ethernet port.

IP switch gives higher transmit priority to packets originating in the tele-

SoundPoint® IP 500 Switch - Port Priorities

To help ensure good voice quality, the Ethernet switch embedded in the SoundPoint® IP should be configured to give voice traffic emanating from the tele-

phone higher transmit priority than those from a device connected to the PC port. If not using a VLAN (VLAN blank in the setup menu), this will automatically be the case. If using a VLAN, ensure that the 802.1p priorities for both default and RTP packet types are set to 2 or greater. Otherwise, these packets will compete equally with those from the PC port. For more information, see 4.6.1.7 Quality of Service <QOS/> on page 78.

4.2 Application Network Setup

4.2.1 RTP Ports

SoundPoint® IP is compatible with RFC 1889 - RTP: A Transport Protocol for RealTime Applications. Consistent with RFC 1889, SoundPoint® IP treats all RTP streams

as bi-directional from a control perspective and expects that both RTP endpoints will negotiate the respective destination IP addresses and ports. This allows RTCP to operate correctly even with RTP media flowing in only a single direction, or not at all. It also allows greater security: packets from unauthorized sources can be rejected.

SoundPoint address. Packets arriving from a non-negotiated IP address can be discarded.

IP can filter incoming RTP packets arriving on a particular port by IP

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SoundPoint® IP can also enforce symmetric port operation for RTP packets: packets arriving with the source port set to other than the negotiated remote sink port can be rejected.

SoundPoint® IP can also jam the destination transport port to a specified value regardless of the negotiated port. This can be useful for punching through firewalls. When this is enabled, all RTP traffic will be sent to the specified port and will be expected to arrive on that port as well. Incoming packets are sorted by the source IP address and port, allowing multiple RTP streams to be multiplexed.

The RTP port range used by SoundPoint® IP can be specified. Since the telephone supports conferencing and multiple RTP streams, several ports can be used concur-

rently. Consistent with RFC 1889, SoundPoint® IP uses the next higher odd port to send and receive RTCP.

Configuration file: ipmid.cfg

Central (boot server)

Web Server (if enabled)

Local

Local Phone User Interface

Specify whether to filter incoming RTP packets by IP address, whether to require symmetric port usage, whether to jam the destination port and specify the local RTP port range start.

• For more information, see 4.6.1.8.3.1 RTP <RTP/> on page 82.

Specify whether to filter incoming RTP packets by IP address, whether to require symmetric port usage, whether to jam the destination port and specify the local RTP port range start.

Navigate to: http://<phoneIPAddress>/coreConf.htm#rt

None.

4.2.2 Working with Network Address Translation (NAT)

SoundPoint® IP can work with certain types of network address translation (NAT). SoundPoint® IP signaling and RTP traffic use symmetric ports (the source port in

transmitted packets is the same as the associated listening port used to receive packets) and the external IP address and ports used by the NAT on the telephone’s behalf can be configured on a per-telephone basis.

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Central (boot server)

Local

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the external NAT IP address and the ports to be used for signaling and RTP traffic.

• For more information, see 4.6.3.6 Network Address Translation <nat/> on page 116.

Specify the external NAT IP address and the ports to be used for signaling and the RTP traffic.

Navigate to: http://<phoneIPAddress>/coreConf.htm#na

None.

4.3 Updating and Rebooting

The bootROM, application executable, and configuration files can be updated automatically via the centralized provisioning (boot server) model.

To automatically update:

1. Back-up old application and configuration files. The old configuration can be easily restored by reverting to the back-up files.

2. Customize new configuration files or apply new or changed parameters to the old configuration files. Differences between old and new versions of configuration files are explained in the Release the software.Changes to site-wide configuration files such as ipmid.cfg can be done manually, but a scripting tool is useful to change per-phone configuration files.

3. Save the new configuration files and images (such as sip.ld) on the boot

server.

4. Reboot the telephones (see below).

For more information, see 2.2.2 Application Configuration on page 9.

5. For TFTP, refer to 2.2.2.1.1 FTP vs. TFTP on page 10

Notes which accompany

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To reboot telephones manually, press and hold the following keys simultaneously until a confirmation tone is heard or for about three seconds:

Manual Reboot: Key Presses

IP 500: Volume-, Volume+, Hold, Messages

IP 600 Volume-, Volume+, Mute, Messages

Centralized Reboot:

The telephones can be rebooted remotely via the SIP signaling protocol. Refer to

4.6.2.1.3.4 Special Events <specialEvent/> on page 105.

4.4 Event Logging

The telephones maintain both boot and application event log files. These files can be helpful when diagnosing problems. The event log files are stored in SoundPoint® IP’s

flash file system and are periodically uploaded to the provisioning boot server if permitted by security policy. The files are stored in the telephone’s home directory or a user-configurable directory on the boot server. Both overwrite and append modes are supported for the application event log.

The event log files are:

• <Ethernet address>-boot.log

• <Ethernet address>-app.log

The boot log file is uploaded to the boot server after every reboot.

The application log file is uploaded periodically or when the local copy reaches a predetermined size.

As an additional diagnostic tool, both log files can be uploaded on demand to the boot server by pressing and holding the four arrow keys until a confirmation tone is heard or for about three seconds. Log files uploaded in this manner are named:

• <Ethernet address>-now-boot.log

• <Ethernet address>-now-app.log

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Central (boot server)

Local

Configuration file: ipmid.cfg

Configuration file: sip.cfg

Configuration file: <Ethernet address>.cfg

Web Server (if enabled)

Local Phone User Interface

Specify a multitude of event logging settings.

• For more information, see 4.6.1.16 Event Logging <logging/> on page 94.

Specify “sip” event logging settings.

• For more information, see 4.6.2.1.5 SIP Logging <logging/> <level/> <change/> on page 108.

Specify different directory to use for log files if desired.

• For more information, see 2.2.2.1.2.1 Master Configuration File on page 10.

Specify a multitude of event logging settings.

Navigate to: http://<phoneIPAddress>/coreConf.htm#lo

Specify “sip” event logging settings.

Navigate to: http://<phoneIPAddress>/appConf.htm#lo

None.

4.5 Audio Quality Issues and VLANs

SoundPoint® IP contains both network layer and Ethernet layer support for prioritizing voice and signaling traffic over the network. Quality of Service (QoS) parameters include IP type-of-service (TOS) bits, and Ethernet IEEE 802.1p user priority. These

can be set on a per-protocol basis. SoundPoint® IP also supports RTCP per RFC 1889.

4.5.1 IP TOS

The “type of service” field in an IP packet header consists of four TOS bits and a 3-bit precedence field. Each TOS bit can be set to either 0 or 1. The precedence field can be set to a value from 0 through 7. The type of service can be configured specifically for RTP packets and call control packets, such as SIP signaling packets.

Central (boot server)

Configuration file: ipmid.cfg

Specify protocol-specific IP TOS settings.

• For more information, see 4.6.1.7.2 IP TOS <IP/> on page 78.

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Web Server (if enabled)

Local

Local Phone User Interface

4.5.2 IEEE 802.1p/Q

The telephone will tag all Ethernet packets it transmits with an 802.1Q VLAN header when it has a valid VLAN ID set in its network configuration, or is instructed to tag packets via Cisco Discovery Protocol (CDP) running on a connected Ethernet switch. The 802.1p/Q user_priority field can be set to a value from 0 to 7. The user_priority can be configured specifically for RTP packets and call control packets, such as SIP signaling packets, with default settings configurable for all other packets.

Central (boot server)

Configuration file: ipmid.cfg

Web Server (if enabled)

Specify IP TOS settings.

Navigate to: http://<phoneIPAddress>/coreConf.htm#qo

None.

Specify default and protocol-specific 802.1p/Q settings.

• For more information, see 4.6.1.7.1 Ethernet IEEE

802.1p/Q <Ethernet/> on page 78.

Specify 802.1p/Q settings.

Navigate to http://<phoneIPAddress>/coreConf.htm#qo

Local Phone User Interface

Local

4.5.3 RTCP Support

SoundPoint® IP supports RTCP per RFC 1889. For each RTP stream, which, by convention, uses even ports only, SoundPoint

and receive RTCP reports.

Specify whether CDP is to be used or manually set the VLAN ID.

Phase 1: bootRom - Navigate to: SETUP menu during autoboot countdown.

Phase 2: Application - Navigate to: Menu>Settings>Network Configuration

• For more information, see 2.2.1.2.1 Main Menu on page 6.

IP uses the next higher odd port to send

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4.6 Configuration Files

This section is a reference for all parameters that are configurable when using the centralized provisioning installation model. It is divided into three sections:

• Core Configuration - ipmid.cfg

• Application Configuration - sip.cfg for the SIP application, mgcp.cfg for the MGCP application.

• Per-phone Configuration - phone1.cfg

Notes

In the following tables, Null should be interpreted as the empty string, that is, attributeName=“” when the file is viewed in a text editor. To enter special characters in a configuration file, enter the appropriate sequence using a text editor. See the table below.

Special Character Required Character Sequence in Text Editor

” "

’ '

4.6.1 Core Configuration - ipmid.cfg

4.6.1.1 Localization <localization/>

SoundPoint® IP has a multilingual user interface. It supports both North American and international time and date formats. The call progress tones can also be customized. For more information, see 4.6.1.1.3 Call Progress Tones <callProgTones> on page 58, 4.6.1.3.2 Chord Sets <chord_sets/> on page 61, and 4.6.1.5.1.1 Call Progress Patterns on page 65.

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4.6.1.1.1 Multilingual <multilingual/>

SoundPoint® IP’s multilingual feature is based on string dictionary files downloaded from the boot server. These files are encoded in standalone XML format. Several western European and Asian languages are included with the distribution.

Permitted

Attribute

Values

Interpretation

lcl.ml.lang Null

An exact match for one of the folder names under the SoundPointIPLocalization folder on the boot server.

lcl.ml.lang.menu.x String in the format

language_region

lcl.ml.lang.cpt.x positive integer The call progress tone index to be

If Null, the default internal language (US English) will be used, otherwise, the language to be used may be specified in the format language-region.

Multiple lcl.ml.lang.menu.x attributes are supported - as many languages as are desired. However, the lcl.ml.lang.menu.x attributes must be sequential (lcl.ml.lang.menu.1, lcl.ml.lang.menu.2, lcl.ml.lang.menu.3, ..., lcl.ml.lang.menu.N) with no gaps and the strings must match exactly a folder name under the SoundPointIPLocalization folder on the boot server for SoundPoint to be able to locate the dictionary file.

associated to this language. See

4.6.1.1.3 Call Progress Tones <callProgTones> on page 58.

lcl.ml.lang.clock.x.24HourClock 0,1 If attribute present, overrides

lcl.datetime.time.24HourClock;

If 1, display time in 24-hour clock mode rather than a.m./p.m.

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Permitted

Attribute

Values

Interpretation

lcl.ml.lang.clock.x.format string which

includes ‘D’, ‘d’ and ‘M’ and two optional comma’s

lcl.ml.lang.clock.x.longFormat 0, 1 If attribute present, overrides

lcl.ml.lang.clock.x.dateTop 0, 1 If attribute present, overrides

lcl.ml.lang.y.list “All” or a comma-

separated list

If attribute present, overrides lcl.datetime.date.format; D = day of week d = day M = month Up to two comma’s may be included. e.g. D,dM = Thursday, 3 July Md,D = July 3, Thursday The field may contain 0, 1 or 2 comma’s which can occur only between characters and only one at a time i.e. “D,,dM” is illegal.

lcl.datetime.date.longFormat;

If 1, display the day and month in long format (Friday/November), otherwise use abbreviations (Fri/Nov).

lcl.datetime.date.dateTop;

If 1, display date above time, otherwise display time above date.

A list of the languages supported on hardware platform ‘y’ where ‘y’ can be IP_500 or IP_600.

4.6.1.1.1.1 Adding New Languages

Follow these steps to add new languages to those included with the distribution:

1. Create a new dictionary file based on an existing one.

2. Change the strings making sure to encode the XML file in UTF-8 but also

ensuring the UTF-8 characters chosen are within the Unicode character ranges indicated in

3. Place the file in an appropriately named folder according to the format

language_region parallel to the other dictionary files under the SoundPoint IPLocalization folder on the boot server.

4. Add a lcl.ml.lang.clock.menu.x attribute to the configuration file.

5. Add lcl.ml.lang.clock.x.24HourClock, lcl.ml.lang.clock.x.format,

lcl.ml.lang.clock.x.longFormat and lcl.ml.lang.clock.x.dateTop attributes and set them according to the regional preferences.

6. (Optional) Set lcl.ml.lang to be the new language_region string.

3.5.1 Multilingual User Interface on page 41.

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4.6.1.1.2 Date and Time <datetime/>

Permitted

Attribute

lcl.datetime.time.24HourClock 0,1 If 1, display time in 24-hour clock mode rather

Values

Interpretation

than a.m./p.m.

lcl.datetime.date.format string which

includes ‘D’, ‘d’ and ‘M’ and two optional comma’s

lcl.datetime.date.longFormat 0,1 If 1, display the day and month in long format

lcl.datetime.date.dateTop 0, 1 If 1, display date above time else display time

Controls format of date string. D = day of week d = day M = month Up to two comma’s may be included. e.g. D,dM = Thursday, 3 July Md,D = July 3, Thursday The field may contain 0, 1 or 2 comma’s which can occur only between characters and only one at a time i.e. “D,,dM” is illegal.

(Friday/November), otherwise, use abbreviations (Fri/Nov).

above date.

4.6.1.1.3 Call Progress Tones <callProgTones>

Call progress tone overrides can be used to customize the tones for a particular country or region. The overrides set offered by default spans all default languages on the telephone. Tone overrides are based on the ITU-T Recommendation E.180 Supplement 2 entitled Telephone Network and ISDN - Operation, numbering, routing and mobile service - Various tones used in national networks.

Permitted

Attribute

lcl.cpt positive

lcl.cpt.menu.x string String to specify the country or region e.g. Italy.

Values

integer OR blank

Interpretation

The index of the default tone overrides to be selected by the phone. If blank, default call progress tones are used.

Multiple lcl.cpt.menu.x strings are supported, the strings are displayed in the Call Progress Tones menu. The lcl.cpt.menu.x attributes must be sequential (lcl.cpt.menu.1, lcl.cpt.menu.2, lcl.cpt.menu.3, ..., lcl.cpt.menu.N) with no gaps.

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In the following table, x is the index of the region as specified by the x index of the lcl.cpt.menu.x attribute above, y is the chord set number and cat is one of cp or misc. For more information, see 4.6.1.5.1.1 Call Progress Patterns on page 65.

Permitted

Attribute

lcl.cpt.chord.cat.x.y.freq.z 0-1600 Frequency for this component in Hertz; up to four

lcl.cpt.chord.cat.x.y.level.z -57 to 3 Level of this component in dBm0.

Values Interpretation

chord-set components can be specified (z=1, 2, 3,

4).

lcl.cpt.chord.cat.x.y.onDur positive

integer

lcl.cpt.chord.cat.x.y.offDur positive

integer

lcl.cpt.chord.cat.x.y.repeat positive

integer

On duration in milliseconds, 0=infinite.

Off duration in milliseconds, 0=infinite.

Specifies how many times the ON/OFF cadence is repeated, 0=infinite.

4.6.1.2 User Preferences <user_preferences/>

Permitted

Attribute

up.headsetMode 0,1 0 If set to 1, the headset will be selected as

up.useDirectoryNames 0,1 0 If set to 1, the name fields of directory

Values

Default Interpretation

the preferred transducer after its first use until the headset key is pressed again; otherwise, handsfree will be selected preferentially over the headset.

entries which match incoming calls will be used for caller identification display and in the call lists instead of the name provided via network signaling.

up.welcomeSoundEnabled 0, 1 1 If set to 1, play welcome sound effect

after a reboot.

up.welcomeSoundOnWarmBootEnabled

up.oneTouchVoiceMail 0, 1 0 If set to 1, the voicemail summary dis-

0, 1 0 If set to 1, play welcome sound effect on

warm as well as cold boots, otherwise only a cold boot will trigger the welcome sound effect.

play is bypassed and voicemail is dialed directly (if configured).

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4.6.1.3 Tones <tones/>

This section describes configuration items for the tone resources available in the telephone.

4.6.1.3.1 Dual Tone Multi-Frequency <DTMF/>

Permitted

Attribute

tone.dtmf.level -33 to -3 -15 Level of the high frequency compo-

Values

Default Interpretation

nent of the DTMF digit measured in dBm0; the low frequency tone will be two dB lower.

tone.dtmf.onTime positive

integer

tone.dtmf.offTime positive

integer

tone.dtmf.chassis.masking 0, 1 0 If set to 1, DTMF tones will be sub-

50 When a sequence of DTMF tones is

played out automatically, this is the length of time in milliseconds the tones will be generated for; this is also the minimum time the tone will be played for when dialing manually (even if key press is shorter).

50 When a sequence of DTMF tones is

played out automatically, this is the length of time in milliseconds the telephone will pause between digits; this is also the minimum interdigit time when dialing manually.

stituted with a non-DTMF pacifier tone when dialing in hands-free mode. This prevents DTMF digits being broadcast to other surrounding telephony devices or being inadvertently transmitted in-band due to local acoustic echo.

Note: tone.dtmf.chassis.masking should only be enabled when tone.dtmf.viaRtp is disabled.

tone.dtmf.stim.pac.offHookOnly 0, 1 0 Not currently used.

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Permitted

Attribute

tone.dtmf.viaRtp 0, 1 1 If set to 1, encode DTMF in the

tone.dtmf.rfc2833Control 0, 1 1 If set to 1, the telephone will indi-

Values

Default Interpretation

active RTP stream, otherwise, DTMF may be encoded within the signaling protocol only when the protocol offers the option.

Note: tone.dtmf.chassis.masking should be enabled when tone.dtmf.viaRtp is disabled.

cate a preference for encoding DTMF via RFC 2833 format in its Session Description Protocol (SDP) offers by showing support for the telephone-event payload type; this does not affect SDP answers, these will always honor the DTMF format present in the offer since the telephone has native support for RFC

2833.

tone.dtmf.rfc2833Payload 96-127 101 The telephone-event payload encod-

4.6.1.3.2 Chord Sets <chord_sets/>

Chord sets are the building blocks of sound effects that use synthesized rather than sampled audio (most call progress and ringer sound effects). A chord-set is a multifrequency note with an optional on/off cadence. A chord-set can contain up to four frequency components generated simultaneously, each with its own level.

There are three blocks of chord sets:

• callProg: used for call progress sound effect patterns

• ringer

• misc (miscellaneous)

All three blocks use the same chord set specification format.

ing in the dynamic range to be used in SDP offers.

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In the following table, x is the chord-set number and cat is one of callProg, ringer, or misc.

Permitted

Attribute

tone.chord.cat.x.freq.y 0-1600 Frequency for this component in Hertz; up to four

tone.chord.cat.x.level.y -57 to 3 Level of this component in dBm0.

Values Interpretation

chord-set components can be specified (y=1, 2, 3,

4).

tone.chord.cat.x.onDur positive

integer

tone.chord.cat.x.offDur positive

integer

tone.chord.cat.x.repeat positive

integer

On duration in milliseconds, 0=infinite.

Off duration in milliseconds, 0=infinite.

Specifies how many times the ON/OFF cadence is repeated, 0=infinite.

4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/>

SoundPoint® IP supports the following sampled audio WAVE file (.wav) formats:

• mono 8 kHz G.711 µ-Law

• G.711 A-Law

• L16/1600 (16-bit, 16 kHz sampling rate, mono)

The telephone uses built-in wave files for some sound effects. The built-in wave files can be replaced with files downloaded from the boot server or from the Internet, however, these are stored in volatile memory so the files will need to remain accessible should the telephone need to be rebooted. Files will be truncated to a maximum size of 300 kilobytes.

In the following table, x is the sampled audio file number.

Attribute Permitted Values Interpretation

saf.x Null OR valid path

name OR an RFC 1738-compliant URL to a HTTP, FTP, or TFTP wave file resource.

Note: Refer to the above wave file format restrictions.

If Null, the telephone will use a built-in file;

If set to a path name, SoundPoint download this file at boot time from the boot server;

If set to a URL, SoundPoint load this file at boot time from the Internet.

Note: A TFTP URL is expected to be in the format: tftp://<host>/[pathname]<filename>, for example: tftp:// somehost.example.com/sounds/example.wav

IP will attempt to

IP will attempt to down-

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The following table defines the default usage of the sampled audio files with the telephone:

Sampled Audio File Default use within telephone (pattern reference)

1 Welcome SE (se.pat.misc.7)

2 Ringer 13 (se.pat.ringer.13)

3 Ringer 14 (se.pat.ringer.14)

4 Ringer 15 (se.pat.ringer.15)

5 Ringer 16 (se.pat.ringer.16)

6 Ringer 17 (se.pat.ringer.17)

7 Ringer 18 (se.pat.ringer.18)

8 Ringer 19 (se.pat.ringer.19)

9 Ringer 20 (se.pat.ringer.20)

10 Ringer 21 (se.pat.ringer.21)

11 Ringer 22 (se.pat.ringer.22)

12-24 Not used.

4.6.1.5 Sound Effects <sound_effects/>

SoundPoint® IP uses both synthesized (based on the chord-sets described earlier) and sampled audio sound effects. Sound effects are defined by patterns: rudimentary sequences of chord-sets, silence periods, and wave files.

Permitted

Attribute

se.stutterOnVoiceMail 0, 1 1 If set to 1, stuttered dial tone is used in place

se.appLocalEnabled 0, 1 1 If set to 1, local user interface sound effects

Values

Default Interpretation

of normal dial tone to indicate that one or more messages (voice-mail) are waiting at the message center.

such as confirmation/error tones, will be enabled.

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4.6.1.5.1 Patterns <patterns/>

Patterns use a simple script language that allows different chord sets or wave files to be strung together with periods of silence. The script language uses the following instructions:

Instruction Meaning Example

sampled (n) Play sampled audio

file n

chord (n, d) Play chord set n (d is

optional and allows the chord set ON duration to be overridden to d milliseconds)

silence (d) Play silence for d

milliseconds (Rx audio is not muted)

branch (n) Advance n instruc-

tions and execute that instruction (n must be negative and must not branch beyond the first instruction)

se.pat.callProg.x.inst.y.type =”sampled” (sampled audio file instruction type)

se.pat.callProg.x.inst.y.value =”3” (specifies sampled audio file 3)

se.pat.callProg.x.inst.y.type = “chord” (chord set instruction type)

se.pat.callProg.x.inst.y.value = “3” (specifies call progress chord set 3)

se.pat.callProg.x.inst.y.param = “2000” (override ON duration of chord set to 2000 milliseconds)

se.pat.callProg.x.inst.y.type = “silence” (silence instruction type)

se.pat.callProg.x.inst.y.value = “300” (specifies silence is to last 300 milliseconds)

se.pat.callProg.x.inst.y.type = “branch” (branch instruction type)

se.pat.callProg.x.inst.y.value = “-5” (step back 5 instructions and execute that instruction)

a. Currently, patterns that use the sampled instruction are limited to the following format:

sampled followed by optional silence and optional branch back to the beginning.

In the following table, x is the pattern number, y is the instruction number. Both x and y need to be sequential. There are three categories of sound effect patterns: callProg (call progress patterns), ringer and misc (miscellaneous).

Attribute Permitted Values Interpretation

se.pat.callProg.x.name UTF-8 encoded

string

Used for identification purposes in the user interface (currently used for ringer patterns only); for patterns that use a sampled audio file which has been overridden by a downloaded replacement, the se.pat.ringer.x.name parameter will be overridden in the user interface by the file names of the wave file.

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Attribute Permitted Values Interpretation

se.pat.callProg.x.inst.y.type sampled OR chord

se.pat.callProg.x.inst.y.value integer Instruction type:

se.pat.callProg.x.inst.y.param positive integer If instruction type is chord, this optional parameter

4.6.1.5.1.1 Call Progress Patterns

The following table maps call progress patterns to their usage within the telephone.

Call progress pattern number

Use within SoundPoint® IP

OR silence OR branch

As above.

Interpretation:

sampled

chord

silence

branch

specifies the on duration to be used, overriding the on duration specified in the chord-set definition.

sampled audio file number

chord set number

silence duration in ms

number of instructions to advance

1 dial tone

2 busy tone

3 ring back tone

4 reorder tone

5 stuttered dial tone

6 call waiting tone

7 alternate call waiting tone (distinctive)

8 confirmation tone

9 howler tone (off-hook warning)

10 record warning

11 message waiting tone

12 alerting

13 intercom announcement tone

14 barge-in tone

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4.6.1.5.1.2 Ringer Patterns

The following table maps ringer pattern numbers to their default descriptions.

Ringer pattern number

Default description

2 Low Trill

3 Low Double Trill

4 Medium Trill

5 Medium Double Trill

6 High Trill

7 High Double Trill

8 Highest Trill

9 Highest Double Trill

10 Beeble

11 Triplet

12 Ringback-style

14 Sampled audio file 3

15 Sampled audio file 4

Silent Ring

Sampled audio file 2

16 Sampled audio file 5

17 Sampled audio file 6

18 Sampled audio file 7

19 Sampled audio file 8

20 Sampled audio file 9

21 Sampled audio file 10

22 Sampled audio file 11

a. Silent Ring will only provide a visual indication of an incoming call, but no audio

indication.

b. Sampled audio files 1-21 all use the same built-in file unless that file has been

replaced with a downloaded file. For more information, see 4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> on page 62.

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4.6.1.5.1.3 Miscellaneous Patterns

The following table maps miscellaneous patterns to their usage within the telephone.

Miscellaneous pattern number

1 new message waiting indication

2 new instant message

3 not used

4 local hold notification

5 positive confirmation

6 negative confirmation

7 welcome (boot up)

Use within SoundPoint® IP

4.6.1.5.2 Ring type <ringType/>

Ring type is used to define a simple class of ring to be applied based on some credentials that are usually carried within the network protocol. The ring class includes attributes such as call-waiting and ringer index, if appropriate. The ring class can use one of four types of ring that are defined as follows:

ring Play a specified ring pattern or call waiting indication

visual Provide only a visual indication (no audio indication) of incoming call (no

ringer needs to be specified).

answer

ring-answer

a. Note that auto-answer on incoming call is currently only applied if there is no other

call in progress on the phone at the time.

b. See note a.

In the following table, x is the ring class number. The x index needs to be sequential.

Provide auto-answer on incoming call

Provide auto answer on incoming call after a ring period

Attribute Permitted Values Interpretation

se.rt.enabled 0,1 Set to 1 to enable the ring type feature within

the telephone, 0 otherwise.

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Attribute Permitted Values Interpretation

se.rt.modification.enabled 0,1 Set to 1 to allow user modification via local

user interface of the pre-defined ring type

enabled for modification

se.rt.x.name UTF-8 encoded string Used for identification purposes in the user

interface

se.rt.x.type ring OR visual OR

answer OR ringanswer

se.rt.x.ringer integer - only relevant

if the type is set to ‘ring’ or ‘ring-answer’

se.rt.x.callWait integer - only relevant

if the type is set to ‘ring’ or ‘ring-answer’

se.rt.x.timeout positive integer - only

relevant if the type is set to ‘ring-answer’. Default value is 2000.

se.rt.x.mod 0,1 Set to 1 if the user interface should allow for

a. Modification via user interface will be implemented in a future release. b. Modification via user interface will be implemented in a future release.

As defined in table above.

The ringer index to be used for this class of ring. The ringer index should match one of

4.6.1.5.1.2 Ringer Patterns on page 66.

The call waiting index to be used for this class of ring. The call waiting index should match one defined in Progress Patterns on page 65.

The duration of the ring in milliseconds before the call is auto answered. If this field is omitted or is left blank, a value of 2000 is used.

modification by the user of the ringer index used for this ring class.

4.6.1.5.1.1 Call

4.6.1.6 Voice Settings <voice/>

4.6.1.6.1 Voice Coding Algorithms <codecs/>

SoundPoint® IP supports the following voice codecs:

MIME

Algorithm

G.711µ-law PMCU G711mu 64 Kbps 8 Ksps 10ms - 80ms 3.5KHz

G.711a-law PCMA G711A 64 Kbps 8 Ksps 10ms - 80ms 3.5KHz

G.729AB G729 G729AB 8 Kbps 8 Ksps 10ms - 80ms 3.5KHz

Type

Label Bit Rate

Sample Rate

Frame Size

Effective Audio Bandwidth

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4.6.1.6.1.1 Codec Preferences <preferences/>

Permitted

Attribute

voice.codecPref.G711Mu Null, 1-3 1 Specifies the codec preference.

Values

Default Interpretation

voice.codecPref.G711A Null, 1-3 2

voice.codecPref.G729AB Null, 1-3 3

4.6.1.6.1.2 Codec Profiles <profiles/>

The following profile attributes can be adjusted for each of the three supported codecs. In the table, x=G711Mu, G711A, or G729AB.

Attribute

voice.audioProfile.x.payloadSize 10, 20, 30,

Permitted Values

...80

1=highest

3=lowest

Null=do not use

Give each codec a unique priority, this will dictate the order used in SDP negotiations.

Note

: If less than two codecs are speci-

fied, SoundPoint cally use G711Mu and G711A with default parameters.

IP will automati-

Interpretation

Preferred Tx payload size in milliseconds to be provided in SDP offers and used in the absence of ptime negotiations. This is also the range of supported Rx payload sizes.

voice.audioProfile.x.jitterBufferMin 40, 50, 60, ...

(multiple of

10)

The smallest jitter buffer depth (in milliseconds) that must be achieved before play out begins for the first time. Once this depth has been achieved initially, the depth may fall below this point and play out will still continue. This parameter should be set to the smallest possible value which is at least two packet payloads, and larger than the expected short term average jitter.

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Permitted

Attribute

Values

Interpretation

voice.audioProfile.x.jitterBufferShrink 10, 20, 30, ...

(multiple of

10)

voice.audiProfile.x.jitterBufferMax > jitterBuf-

ferMin, <=500, multiple of 10

4.6.1.6.2 Volume Persistence <volume/>

The user’s selection of the receive volume during a call can be remembered between calls. This can be configured per termination (handset, headset and handsfree/chassis). In some countries regulations exist which dictate that receive volume should be reset to nominal at the start of each call on handset and headset.

The absolute minimum duration time (in milliseconds) of RTP packet Rx with no packet loss between jitter buffer size shrinks. Use smaller values (1000 ms) to minimize the delay on known good networks. Use larger values to minimize packet loss on networks with large jitter (3000 ms).

The largest jitter buffer depth to be supported (in milliseconds). Jitter above this size will always cause lost packets. This parameter should be set to the smallest possible value that will support the expected network jitter.

Attribute

voice.volume.persist.handset 0, 1 0 If set to 1, the receive volume will be

voice.volume.persist.headset 0, 1 0

voice.volume.persist.handsfree 0, 1 1

4.6.1.6.3 Gains <gains/>

The default gain settings have been carefully adjusted to comply with the TIA-810-A digital telephony standard. Changing these is not advised.

Attribute

voice.gain.rx.analog.handset -30, -27, -24,

Permitted Values

..., 0

Default Interpretation

remembered between calls. If set to 0, the receive volume will be reset to nominal at the start of each call.

TIA-810-A Default

0 Rx analog gain in dB when

Interpretation

using handset.

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Permitted

Attribute

voice.gain.rx.analog.headset 0 Rx analog gain in dB when

voice.gain.rx.analog.chassis 0 Rx analog gain in dB when

voice.gain.rx.analog.ringer 0 Rx analog gain in dB for

voice.gain.rx.digital.handset -30, -27, -24,

voice.gain.rx.digital.headset -30, -27, -24,

voice.gain.rx.digital.chassis -15, -12 -9,

Values

..., 0

..., 15

TIA-810-A Default

-15 Rx digital gain in dB. Sets

-21 Rx digital gain in dB. Sets

0 Rx digital gain in dB. Sets

Interpretation

using headset.

using speakerphone.

sound effects via speakerphone.

handset volume control initial position at beginning of call.

headset volume control initial position at beginning of call.

speakerphone volume control initial position after a reboot.

voice.gain.rx.digital.ringer -30, -27, -24,

..., 0

voice.gain.rx.analog.handset.sidetone

voice.gain.rx.analog.headset.sidetone -24

voice.gain.tx.analog.handset -36, -33, -30,

voice.gain.tx.analog.headset 3 Tx analog gain in dB when

voice.gain.tx.analog.chassis -3 Tx analog gain in dB when

voice.gain.tx.digital.handset 60, -57, -54,

voice.gain.tx.digital.headset 0 Tx digital gain in dB when

voice.gain.tx.digital.chassis 3 Tx digital gain in dB when

-12, -14, 16,..., -24, 26

..., 12

..., 30

-21 Rx digital gain in dB. Sets sound effects volume control initial position after a reboot.

-24 Analog sidetone gain in dB, -26=no sidetone.

3 Tx analog gain in dB when

using handset.

using headset.

using speakerphone.

0 Tx digital gain in dB when

using handset.

using headset.

using speakerphone.

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Permitted

Attribute

voice.gain.tx.analog.preamp.handset 0, 14, 23, -1 23 Preamplifier gain in dB.

voice.gain.tx.analog.preamp.headset 23

voice.gain.tx.analog.preamp.chasis 0, 20, 32, 42, -142

Values

TIA-810-A Default

4.6.1.6.4 Acoustic Echo Cancellation <AEC/>

These settings control the performance of the speakerphone acoustic echo canceller. Changing these settings is not advised.

Permitted

Attribute

voice.aec.hs.enable 0, 1 0 If set to 1, enable AEC for hand-

voice.aec.hs.lowFreqCutOff 125 to 7600, <

Values

highFreqCutOff

Default Interpretation

125 PCM channel low frequency cut-

Interpretation

-1=mute

set.

off, in Hz, for handset. Reasonable upper limit is 300Hz (narrowband telephony). The frequency cut-offs define the frequency range over which Tx audio DSP (AEC, AES, etc.,) is going to be performed. All Tx outside this frequency range will Tx to zero. These limits have no effect when there is no DSP being performed (such as in the low delay handset mode).

voice.aec.hs.highFreqCutOff 125 to 7600, >

lowFreqCutOff

3800 PCM channel high frequency

cutoff, in Hz, for handset. Reasonable lower limit is 3500Hz (narrowband telephony). The frequency cut-offs define the frequency range over which Tx audio DSP is going to be performed. All Tx outside this frequency range will Tx to zero. These limits have no effect when there is no DSP being performed (such as in the low delay handset mode).

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Permitted

Attribute

voice.aec.hs.erlTab_0_300 -30, -29, ..., 0 -24 Open loop ERL table 0 to 300

voice.aec.hs.erlTab_300_600 -30, -29, ..., 0 -24 Open loop ERL table 300 to 600

voice.aec.hs.erlTab_600_1500 -30, -29, ..., 0 -24 Open loop ERL table 600 to

voice.aec.hs.erlTab_1500_3500 -30, -29, ..., 0 -24 Open Loop ERL table 1500 to

voice.aec.hs.erlTab_3500_7000 -30, -29, ..., 0 -24 Open loop ERL table 3500 to

Values

Default Interpretation

Hz, in dB, for handset. (ERL is the ratio of the digital ref signal going to the handset speaker DAC and the digital signal detected by the handset microphone ADC).

Hz, in dB, for handset.

1500 Hz, in dB, for handset.

3500 Hz, in dB, for handset.

7000 Hz, in dB, for handset.

voice.aec.hd.enable 0, 1 0 If set to 1, enable AEC for head-

set.

voice.aec.hd.lowFreqCutOff 125 to 7600, <

highFreqCutOff

voice.aec.hd.highFreqCutOff 125 to 7600, <

lowFreqCutOff

voice.aec.hd.erlTab_0_300 -30, -29, ..., 0 -24 See voice.aec.hs.erlTab_0_300

voice.aec.hd.erlTab_300_600 -30, -29, ..., 0 -24 See voice.aec.hs.erlTab_300_600

voice.aec.hd.erlTab_600_1500 -30, -29, ..., 0 -24 See

voice.aec.hd.erlTab_1500_3500 -30, -29, ..., 0 -24 See

voice.aec.hd.erlTab_3500_7000 -30, -29, ..., 0 -24 See

125 See voice.aec.hs.lowFreqCutOff

3800 See voice.aec.hs.highFreqCutOff

voice.aec.hs.erlTab_600_1500

voice.aec.hs.erlTab_1500_3500

voice.aec.hs.erlTab_3500_7000

voice.aec.hf.enable 0, 1 1 If set to 1, enable AEC for hands-

free.

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Permitted

Attribute

Values

Default Interpretation

voice.aec.hf.lowFreqCutOff 125 to 7600, <

highFreqCutOff

voice.aec.hf.highFreqCutOff 125 to 7600, >

lowFreqCutOff

voice.aec.hf.erlTab_0_300 -30, -29, ..., 0 -6 See voice.aec.hs.erlTab_0_300

voice.aec.hf.erlTab_300_600 -30, -29, ..., 0 0 See voice.aec.hs.erlTab_300_600

voice.aec.hf.erlTab_600_1500 -30, -29, ..., 0 0 See

voice.aec.hf.erlTab_1500_3500 -30, -29, ..., 0 -6 See

voice.aec.hf.erlTab_3500_7000 -30, -29, ..., 0 -6 See

125 See voice.aec.hs.lowFreqCutOff

3800 See voice.aec.hs.highFreqCutOff

voice.aec.hs.erlTab_600_1500

voice.aec.hs.erlTab_1500_3500

voice.aec.hs.erlTab_3500_7000

4.6.1.6.5 Acoustic Echo Suppression <AES/>

These settings control the performance of the speakerphone acoustic echo suppressor. Changing these settings is not advised.

Permitted

Attribute

voice.aes.hs.enable 0, 1 0 If set to 1, enable AES for handset.

voice.aes.hs.duplexBalance 0, 1, 2, ..., 15 7 Index into table of pre-defined full/

voice.aes.hd.enable 0, 1 0 If set to 1, enable AES for headset.

voice.aes.hd.duplexBalance 0, 1, 2, ..., 15 0 Index into table of pre-defined full/

voice.aes.hf.enable 0, 1 1 If set to 1, enable AES for handsfree.

voice.aes.hf.duplexBalance.0 0, 1, 2, ..., 15 10 Index into table of pre-defined full/

Values

Default Interpretation

half-duplex tuning parameters for varying duplex balance for handset.

half-duplex tuning parameters for varying duplex balance for headset.

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain values of -9 dB and lower.

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Permitted

Attribute

voice.aes.hf.duplexBalance.1 0, 1, 2, ..., 15 9 Index into table of pre-defined full/

voice.aes.hf.duplexBalance.2 0, 1, 2, ..., 15 8 Index into table of pre-defined full/

voice.aes.hf.duplexBalance.3 0, 1, 2, ..., 15 7 Index into table of pre-defined full/

voice.aes.hf.duplexBalance.4 0, 1, 2, ..., 15 7 Index into table of pre-defined full/

Values

Default Interpretation

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain of -6 dB.

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain of -3 dB.

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain of 0 dB.

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain of 3 dB.

voice.aes.hf.duplexBalance.5 0, 1, 2, ..., 15 4 Index into table of pre-defined full/

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain of 6 dB

voice.aes.hf.duplexBalance.6 0, 1, 2, ..., 15 3 Index into table of pre-defined full/

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain of 9 dB

voice.aes.hf.duplexBalance.7 0, 1, 2, ..., 15 2 Index into table of pre-defined full/

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain of 12 dB

voice.aes.hf.duplexBalance.8 0, 1, 2, ..., 15 1 Index into table of pre-defined full/

half-duplex tuning parameters for duplex balance at hands-free Rx digital gain of 15 dB

4.6.1.6.6 Background Noise Suppression <NS/>

These settings control the performance of the transmit background noise suppression feature.

Permitted

Attribute

voice.ns.hs.enable 0, 1 0 If set to 1, enable NS for handset.

Values

Default Interpretation

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Permitted

Attribute

voice.ns.hs.signalAttn -9, -8, ..., 0 -6 Noise suppression in dB with active sig-

voice.ns.hs.silenceAttn -9, -8, ..., 0 -9 Noise suppression in dB with no active

voice.ns.hd.enable 0, 1 0 If set to 1, enable NS for headset.

voice.ns.hd.signalAttn -9, -8, ..., 0 0 Noise suppression in dB with active sig-

voice.ns.hd.silenceAttn -9, -8, ..., 0 0 Noise suppression in dB with active sig-

voice.ns.hf.enable 0, 1 1 If set to 1, enable NS for hands-free.

voice.ns.hf.signalAttn -9, -8, ..., 0 -3 Noise suppression in dB with active sig-

voice.ns.hf.silenceAttn -9, -8, ..., 0 -6 Noise suppression in dB with no active

Values

Default Interpretation

nal for handset.

signal for handset.

nal for headset.

nal for handsfree.

signal for handsfree.

4.6.1.6.7 Automatic Gain Control <AGC/>

These settings control the performance of the transmit automatic gain control feature.

Permitted

Attribute

voice.agc.hs.enable 0, 1 0 If set to 1, enable AGC for handset.

voice.agc.hd.enable 0, 1 0 If set to 1, enable AGC for headset.

voice.agc.hf.enable 0, 1 0 If set to 1, enable AGC for handsfree.

Values

Default Interpretation

4.6.1.6.8 Receive Equalization <RXEQ/>

These settings control the performance of the handsfree receive equalizaton feature.

Permitted

Attribute

voice.rxEq.hf.IP_500.enable 0, 1 1 If set to 1, enable Rx equalization for

Values

Default Interpretation

handsfree (IP 500).

voice.rxEq.hf.IP_600.enable 0, 1 1 If set to 1, enable Rx equalizaton for

handsfree (IP 600).

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Permitted

Attribute

voice.rxEq.hf.type 0, 1 0 Filter type used for RxEq, 0=FIR,

Values

Default Interpretation

1=IIR.

voice.rxEq.hf.size max 63 for

FIR

max 12 for IIR

voice.rxEq.hf.coef.0 to .62 Don’t change

these values.

63 Filter size.

- Filter coefficients.

4.6.1.6.9 Transmit Equalization <TXEQ/>

These settings control the performance of the handsfree transmit equalization feature.

Permitted

Attribute

voice.txEq.hf.IP_500.enable 0, 1 1 If set to 1, enable Tx equalization for

voice.txEq.hf.IP_600.enable 0, 1 1 If set to 1, enable Tx equalizaton for

voice.txEq.hf.coef.1 to 20 Don’t change

Values

these values.

Default Interpretation

- Filter coefficients.

handsfree (IP_500).

handsfree (IP_600).

4.6.1.6.10 Voice Activity Detection <VAD/>

These settings control the performance of the voice activity detection (silence suppression) feature.

Permitted

Attribute

voice.vadEnable 0, 1 0 If set to 1, enable VAD.

voice.vadThresh integer from

Values

0 to 30

Default Interpretation

15 The threshold for determining what is active

voice and what is background noise in dB. This does not apply to G.729AB codec operation which has its own built-in VAD function.

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4.6.1.7 Quality of Service <QOS/>

These settings control the Quality of Service (QOS) options on SoundPoint® IP.

4.6.1.7.1 Ethernet IEEE 802.1p/Q <Ethernet/>

These settings control the 802.1p/Q user_priority field.

4.6.1.7.1.1 RTP <RTP/>

These parameters apply to RTP packets.

Permitted

Attribute

qos.ethernet.rtp.user_priority 0-7 5 User-priority used for RTP packets.

Values

Default Interpretation

4.6.1.7.1.2 Call Control <CallControl/>

These parameters apply to call control packets, such as the network protocol signaling.

Attribute

qos.ethernet.callControl.user_priority 0-7 5 User-priority used for call con-

4.6.1.7.1.3 Other <Other/>

These default parameter values are used for all packets which are not set explicitly.

Permitted

Attribute

qos.ethernet.other.user_priority 0-7 2 User-priority used for packets that

Values

Permitted Values

Default Interpretation

trol packets.

Default Interpretation

do not have a per-protocol setting.

4.6.1.7.2 IP TOS <IP/>

These settings control the “type of service” field in outgoing packets.

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4.6.1.7.2.1 RTP <RTP/>

These parameters apply to RTP packets.

Permitted

Attribute

qos.ip.rtp.min_delay 0, 1 1 If set to 1, set min-delay bit in

qos.ip.rtp.max_throughput 0, 1 1 If set to 1, set max-throughput

qos.ip.rtp.max_reliability 0, 1 0 If set to 1, set max-reliability

qos.ip.rtp.min_cost 0, 1 0 If set to 1, set min-cost bit in

Values

Default Interpretation

the IP TOS field of the IP header, or else don’t set it.

bit in the IP TOS field of the IP header, or else don’t set it.

the IP TOS field of the IP header, or else don’t set it.

qos.ip.rtp.precedence 0-7 5 If set to 1, set precedence bits

4.6.1.7.2.2 Call Control <CallControl/>

These parameters apply to call control packets, such as the network protocol signaling.

Attribute

qos.ip.callControl.min_delay 0, 1 1 If set to 1, set min-delay bit in

qos.ip.callControl.max_throughput 0, 1 0 If set to 1, set max-throughput

qos.ip.callControl.max_reliability 0, 1 0 If set to 1, set max-reliability bit

Permitted Values

in the IP TOS field of the IP header, or else don’t set them.

Default Interpretation

the IP TOS field of the IP header, or else don’t set it.

bit in the IP TOS field of the IP header, or else don’t set it.

in the IP TOS field of the IP header, or else don’t set it.

qos.ip.callControl.min_cost 0, 1 0 If set to 1, set min-cost bit in the

IP TOS field of the IP header, or else don’t set it.

qos.ip.callControl.precedence 0-7 5 If set to 1, set precedence bits in

the IP TOS field of the IP header, or else don’t set them.

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4.6.1.8 Basic TCP/IP <TCP_IP/>

4.6.1.8.1 Network Monitoring <netMon/>

Do not alter these values.

Attribute Permitted Values Default

tcpIpApp.netMon.enabled 0, 1 1

tcpIpApp.netMon.period 1 to 86400 30

4.6.1.8.2 Time Synchronization <SNTP/>

The following table describes the parameters used to set up time synchronization and daylight savings time. The defaults shown will enable daylight savings time for North America.

Daylight savings defaults:

• don’t use fixed day, use first or last day of week in the month,

• start DST on the first Sunday in April at 2 am,

• stop DST on the last Sunday in October at 2 am.

Permitted

Attribute

tcpIpApp.sntp.resyncPeriod positive

tcpIpApp.sntp.address

tcpIpApp.sntp.gmtOffset positive or

tcpIpApp.sntp.daylightSavings.enable 0, 1 1 If set to 1, apply day-

Values

integer

valid host name or IP address

negative integer

Default Interpretation

86400 (24 hours)

clock Address of the SNTP

-28800 (Pacific time)

Time in seconds between SNTP resyncs.

server.

Offset in seconds of the local time zone from GMT.

Note: 3600 seconds per hour

light savings rules to displayed time.

tcpIpApp.sntp.daylightSavings.fixedDayEnable

0, 1 0 If set to 1, “April 1st” is

used, otherwise “the first Sunday in April” is used.

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Permitted

Attribute

tcpIpApp.sntp.daylightSavings.start.month 1-12 4 (April) Month to start DST.

tcpIpApp.sntp.daylightSavings.start.date 1-31 1 Day of the month to

tcpIpApp.sntp.daylightSavings.start.time 0-23 2 Time of day to start

Values

Default Interpretation

1=Jan, 2=Feb, ..., 12=Dec

start DST.

DST, in 24 hour clock. 2=2 am, 14=2 pm

tcpIpApp.sntp.daylightSavings.start.dayOfWeek

tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth

tcpIpApp.sntp.daylightSavings.stop.month 1-12 10 Month to stop DST.

tcpIpApp.sntp.daylightSavings.stop.date 1-31 1 Day of the month to

tcpIpApp.sntp.daylightSavings.stop.time 0-23 2 Time of day to stop

tcpIpApp.sntp.daylightSavings.stop.dayOfWeek

1-7 1 Day of week to apply

DST. 1=Sun, 2=Mon, ..., 7=Sat

0, 10 0 If set to 1 and fixedDay-

Enable=0, start DST on the last day of the week (specified by dayOfWeek) in the month, rather than the first in the month.

1=Jan, 2=Feb, ..., 12=Dec

start DST.

DST, in 24 hour clock. 2= 2 am, 14=2 pm

1-7 1 Day of week to stop

DST. 1=Sun, 2=Mon, ..., 7=Sat

tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth

a. Both tcpIpApp.sntp.address and tcpIpApp.sntp.gmtOffset can be provided via DHCP. If so,

the DHCP parameters will override the parameters in ipmid.cfg.

0, 1 1 If set to 1 and fixedDay-

Enable=0, stop DST on the last day of the week (specified by dayOfWeek) in the month, rather than the first in the month.

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4.6.1.8.3 port <port/>

4.6.1.8.3.1 RTP <RTP/>

Permitted

Attribute

tcpIpApp.port.rtp.filterByIp 0, 1 1 If set to 1, reject RTP pack-

tcpIpApp.port.rtp.filterByPort 0, 1 0 If set to 1, reject RTP pack-

Values

Default Interpretation

ets arriving from (sent from) a non-negotiated (via SDP) IP address.

ets arriving from (sent from) a non-negotiated (via SDP) port.

tcpIpApp.port.rtp.forceSend Null, 1024-

65534

tcpIpApp.port.rtp.mediaPortRangeStart Null, even

integer from 1024-65534

Null When non-Null, send all

RTP packets to, and expect all RTP packets to arrive on, the specified port. Note: both tcpIpApp.port.rtp.filterByIp and tcpIpApp.port.rtp.filterByPort must be enabled for this to work.

Null If set to Null, the value 2222

will be used for the first allocated RTP port, otherwise, the specified port will be used. Subsequent ports will be allocated from a pool starting with the specified port plus two up to a value of (start-port + 46), after which the port number will wrap back to the starting value.

4.6.1.9 Web Server <HTTPD/>

SoundPoint® IP contains a local web server for user and administrator features. This can be disabled for applications where it is not needed or where it poses a security

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threat. The web server supports both basic and digest authentication. The authentication user name and password are not configurable for this release.

Permitted

Attribute

httpd.enabled 0, 1 1 If set to 1, the HTTP server will be enabled.

Values

Default Interpretation

4.6.1.9.1 Configuration <cfg/>

Permitted

Attribute

httpd.cfg.enabled 0, 1 1 If set to 1, the HTTP server configuration

httpd.cfg.port 1-65535 80 Port is 80 for HTTP servers. Care should be

Values

Default Interpretation

interface will be enabled.

taken when choosing an alternate port.

4.6.1.10 Call Handling Configuration <call/>

Permitted

Attribute

call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with

call.enableOnNotRegistered 0, 1 1 If set to 1, calls will be allowed when the

call.offeringTimeOut positive

call.ringBackTimeOut positive

call.shared.disableDivert 0, 1 1 If set to 1, disable diversion feature for

Values

integer

Default Interpretation

the reason “busy” if do-not-disturb is enabled.

telephone is not successfully registered, otherwise, calls will not be permitted without a valid registration.

60 Time in seconds to allow an incoming call

to ring before dropping the call, 0=infi-

nite

60 Time in seconds to allow an outgoing call

to remain in the ringback state before dropping the call, 0=infinite.

shared lines.

call.shared.seizeFailReorder 0, 1 1 If set to 1, play re-order tone locally on

shared line seize failure.

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Permitted

Attribute

call.shared.oneTouchResume 0, 1 0 If set to 1, when a shared line has a call on

a. The call diversion, no answer feature will take precedence over this feature if enabled. For

more information, see 4.6.3.3.3 No Answer <noanswer/> on page 112.

Values

Default Interpretation

hold the remote user can press that line and resume the call. If more than one call is on hold on the line then the first one will be selected and resumed automatically. If set to 0, pressing the shared line will bring up a list of the calls on that line and the user can select which call the next action should be applied to.

4.6.1.10.1 Hold, Local Reminder <localReminder/>

Permitted

Attribute

call.hold.localReminder.enabled 0, 1 0 If set to 1, periodically notify the

call.hold.localReminder.period non-negative

call.hold.localReminder.startDelay non-negative

Values

integer

Default Interpretation

60 Time in seconds between subse-

90 Time in seconds to wait before

4.6.1.11 Directory <directory/>

The directory is stored in either flash memory or RAM on SoundPoint® IP. The direc-

tory size is limited based on the amount of flash memory in the telephone6.

6. IP 500 has 2 megabytes of flash memory, IP 600 has 4 megabytes of flash memory.

local user that calls have been on hold for an extended period of time.

quent reminders.

the initial reminder.

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When the volatile storage option is enabled, ensure that a properly configured boot server that allows uploads is available to store a back-up copy of the directory or its contents will be lost when the telephone reboots or loses power.

Permitted

Attribute

dir.local.volatile.2meg 0, 1 0 Attribute applies to platforms

dir.local.nonVolatile.maxSize.2meg 1 to 20 20 Attribute applies to platforms

Values

Default Interpretation

with 2 Mbytes of flash memory.

If set to 1, use volatile storage for telephone-resident copy of the directory to allow for larger size.

with 2 Mbytes of flash memory. Maximum size in Kbytes of non-volatile storage that the directory will be permitted to consume.

dir.local.volatile.4meg 0, 1 0 Applies to platforms with 4

dir.local.nonVolatile.maxSize.4meg 1 to 50 50 Applies to platforms with 4

dir.local.volatile.maxSize 1 to 100 100 Maximum size in Kbytes of

4.6.1.12 Fonts <font/>

These settings control the SoundPoint® IP’s ability to dynamically load an external font file during boot up. Loaded fonts can either overwrite pre-existing fonts embed-

ded within the software (not recommended) or can extend the SoundPoint® IP’s font support for Unicode ranges not already embedded. The font file must be a Microsoft

Mbytes of flash memory.

If set to 1, use volatile storage for telephone-resident copy of the directory to allow for larger size.

Mbytes of flash memory. Maximum size in Kbytes of non-volatile storage that the directory will be permitted to consume.

volatile storage that the directory will be permitted to consume.

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.fnt or .fon7 file format. The font file name must follow a specific pattern as described:

• Font file name: <fontName>_<fontHeightInPixels>_<fontRange>.<fontExtension>

• <fontName> is a free string of characters that typically carries the meaning of the font. Examples are “fontFixedSize” for a fixed-size font, or “fontPropor

tionalSize” for a proportional size font.

• <fontHeightInPixels> describes the font height in number of screen pixels.

• <fontRange> describes the Unicode range covered by this font. Since .fnt or .fon are 256 characters based blocks, the <fontRange> is Uxx00_UxxFF (.fnt file) or Uxx00_UyyFF (.fon file). For more information, see

3.5.1 Multilingual

User Interface on page 41.

• <fontExtension> describes the file type. Either .fnt for single 256 characters font or .fon for multiple .fnt files.

If it is necessary to overwrite an existing font, use these <fontName>_<fontHeightInPixels>:

SoundPoint® IP 500

“fontProp_10” This is the font used widely in the current implementation.

“fontPropSoftkey_10” This is the soft key specific font.

SoundPoint® IP 600

“fontProp_12” This is used in some specific menus like CPU load (a smaller font).

“fontProp_19” This is the font used widely in the current implementation including

for soft keys.

“fontProp_26” This is the font used to display time (but not date).

“fontProp_x” This is a small font used for the CPU/Load/Net utilization graphs,

this is the same as the “fontProp_10” for the SoundPoint

IP 500.

If the <fontName>_<fontHeightInPixels> does not match any of the names above, then the downloaded font will be applied against all fonts defined in the telephone, which means that you may lose the benefit of fonts being calibrated differently depending on their usage. For example, the font used to display the time on the

Sound Point

IP 600 is a large font, larger than the one used to display the date, and if

you overwrite this default font with a unique font, you lose this size aspect.

Example of usage:

• to overwrite the font used for SoundPoint® IP 500 soft keys for ASCII, the name should be “fontPropSoftkey_10_U0000_U00FF.fnt”

7. .fon file format is a collection of .fnt fonts mangled together within a single file.

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• to add support for a new font that will be used everywhere and that is not currently supported. For example, for the Eastern/Central European Czech language, this is Unicode range 100-17F, the name could be “fontCzechIP500_10_U0100_U01FF.fnt” and “fontCzechIP600_19_U0100_U01FF.fnt”

When defining a single .fon file, there is a need for a “font delimiter”, currently “Copyright Polycom Canada Ltd” is used as an embedded delimiter, but this can be configured using “font.delimiter”. The font delimiter is important to retrieve the different mangled .fnt blocks. This font delimiter must be placed in the “copyright” attribute of the .fnt header. .fon files are useful if you want to include support for a large number of font ranges at once, otherwise, if simply adding or changing a few fonts currently in use, multiple .fnt files are recommended since they are easier to work with individually.

Attribute Permitted Values Default Interpretation

font.delimiter string up to 256 ASCII

characters

4.6.1.12.1 IP_500 font <IP_500/>

Attribute Permitted Values Default Interpretation

font.IP_500.x.name fontName_height_Uxx00

_UyyFF.fon OR fontName_height_Uxx00 _UxxFF.fnt

4.6.1.12.2 IP_600 font <IP_600/>

Attribute Permitted Values Default Interpretation

Null Delimiter required to retrieve differ-

ent mangled .fnt blocks.

Null Defines the font file that will be

loaded from boot server during boot up.

Note: When several font.IP_500.x.names are defined, the index x must follow consecutive increasing order.

font.IP_600.x.name fontName_height_Uxx

00_UyyFF.fon OR fontName_height_Uxx 00_UxxFF.fnt

Null Defines the font file that will be

loaded from boot server during boot up.

Note: When several font.IP_600.x.names are defined, the index x must follow consecutive increasing order.

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4.6.1.13 Keys <keys/>

These settings control the scrolling behavior of keys and can be used to change key functions.

Permitted

Attribute

Values

Default Interpretation

key.scrolling.timeout positive

integer

1 The time-out after which a key that is enabled

for scrolling will go into scrolling mode until the key is released. Keys enabled for scrolling are menu navigation keys (left, right, up, down arrows), volume keys, and some context-specific soft keys. The value is an integer multiple of 500 milliseconds (1=500ms).

SoundPoint® IP key functions can be changed from the factory defaults, although this is typically not necessary. For each key whose function you wish to change, add an XML attribute in the format described in the following table to the <keys .../> element of the configuration file. These will override the built-in assignments.

In the following table, x is the key number. IP 500: x=1-40; IP 600: x=1-42

Permitted

Attribute

key.IP_500.x.function.prim Functions

key.IP_600.x.function.prim Functions

Values Interpretation

On IP 500, this attribute sets the function for key

listed below.

On IP 600, this attribute sets the function for key x.

key.IP_500.x.subPoint.prim positive

integer

key.IP_600.x.subPoint.prim positive

integer

The following table lists the functions that are available based on the application:

Function Usable in SIP application Usable in MGCP application

ArrowDown Yes Yes

ArrowLeft Yes Yes

ArrowRight Yes Yes

On IP 500, this attribute sets the sub-identifier for key functions with a secondary array identifier such as SpeedDial.

On IP 600, this attribute sets the sub-identifier for key functions with a secondary array identifier such as SpeedDial.

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Function Usable in SIP application Usable in MGCP application

ArrowUp Yes Yes

Browse No Yes

BuddyStatus Yes No

CallAppearance1 No No

CallAppearance2 No No

CallAppearance3 No No

CallAppearance4 No No

CallAppearance5 No No

CallAppearance6 No No

CallHistory Yes Yes

CallPark No No

CallPickup No No

CallWaiting No No

Callers No Yes

Cancel No No

ColdBoot No No

Conference Yes No

Delete Yes Yes

Dial No No

Dialpad0 Yes Yes

Dialpad1 Yes Yes

Dialpad2 Yes Yes

Dialpad3 Yes Yes

Dialpad4 Yes Yes

Dialpad5 Yes Yes

Dialpad6 Yes Yes

Dialpad7 Yes Yes

Dialpad8 Yes Yes

Dialpad9 Yes Yes

DialpadStar Yes Yes

DialpadPound Yes Yes

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Function Usable in SIP application Usable in MGCP application

Directories Yes Yes

Divert No No

DoNotDisturb Yes Yes

Email No No

Favorites No No

Flash No No

Handsfree Yes Yes

Headset Yes Yes

Feature No Yes

Help No No

Hold Yes No

Line1 Yes Yes

Line2 Yes Yes

Line3 Yes Yes

Line4 Yes Yes

Line5 Yes Yes

Line6 Yes Yes

LocalDirectory No No

Macro No No

Messages Yes No

Menu Yes Yes

MicMute Yes Yes

MyStatus Yes No

Null Yes Yes

Offline Yes No

Redial Yes Yes

Release No No

RemoteDirectory No No

Select Yes Yes

SelfTest No No

Services No No

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Function Usable in SIP application Usable in MGCP application

Setup Yes Yes

Shift No No

SoftKey1 Yes Yes

SoftKey2 Yes Yes

SoftKey3 Yes Yes

SoftKey4 Yes Yes

SpeedDial Yes Yes

SpeedDialMenu Yes Yes

Suspend No No

Transfer Yes No

VoiceMail No No

VolDown Yes Yes

VolUp Yes Yes

WarmBoot No No

4.6.1.14 Bitmaps <bitmaps/>

Bitmaps used by SoundPoint® IP are defined in this section.

4.6.1.14.1 Platform <IP_500/> and <IP_600/>

In the following table, x=IP_500 or IP_600, y is the bitmap number.

Attribute Permitted Values Interpretation

bitmap.x.y.name The bitmap name

representing a bitmap to be used.

This is a bitmap name to be used for assignment when doing animation. If the bitmap should be downloaded from the boot server, the bitmap name must be:

1. Different from any name already in usage in ipmid.cfg.

2. Must match the name of the corresponding <fileName>.bmp expected to be retrieved from the boot server.

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4.6.1.15 Indicators <indicators/>

Indicators (graphic icons, animations, and LED patterns) used by SoundPoint® IP are defined in this section.

Permitted

Attribute

ind.idleDisplay.enabled 0, 1 0 If set to 1, the idle display may support cus-

a. This will be implemented in a future release.

Values

4.6.1.15.1 Animations <Animations/><IP_500/> and <IP_600/>

This section defines bitmap animations composed of bitmap/duration couples. In the following table, x=IP_500 or IP_600, y is the animation number, z is the step in the animation.

Default Interpretation

tom icon presentation if configured prop-

erly in the animation section of ipmid.cfg.

Attribute Permitted Values Interpretation

ind.anim.x.y.frame.z.bitmap A bitmap name

defined previously.

ind.anim.x.y.frame.z.duration positive integer Duration in milliseconds for this step.

4.6.1.15.2 Patterns <Patterns/>

This section defines patterns for the LED indicators. In the following table, x is the pattern number, y is the step in the pattern.

Permitted

Attribute

ind.pattern.x.step.y.state On or Off Turn LED on or off for this step.

ind.pattern.x.step.y.duration positive integer Duration in milliseconds for this step. 0=infinite

ind.pattern.x.step.y.colour Red or Green

Values

(default is Red if not specified)

Bitmap to use.

0=infinite.

Interpretation

For bi-color LEDs, specify color.

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4.6.1.15.3 Classes <Classes/>

This section defines the available classes for the LED and graphical icon indicator types. In the following table, x is the class number, y is the identifier of the state number for that class.

Permitted

Attribute

ind.class.x.state.y.index positive integer For LED type indicators, index refers to pattern

Values

Interpretation

index, i.e. index x in the <Patterns/> tag above. For GraphicIcon type indicators, index refers to animation index, i.e. index y in the <Animations/> tag above.

4.6.1.15.4 Assignments <Assignments/>

This section assigns a type, a class, and, in the case of the GraphicIcon type, a physical location and size in pixels on the LCD display or in the case of the LED type, a physical LED number.

4.6.1.15.4.1 LEDs <led/>

In the following table, x is the LED number.

Permitted

Attribute

ind.led.x.index This is for internal usage only and should not be changed

ind.led.x.class positive integer Assigns the class (defined above) for this indicator.

ind.led.x.physNum This maps logical index to a specific physical LED.

Values

Interpretation

(this is the logical index).

4.6.1.15.4.2 Graphic Icons <gi/> <IP_500/> and <IP_600/>

In the following table, x=IP_500 or IP_600, y is the graphic icon number.

Permitted

Attribute

ind.gi.x.y.index This is for internal usage only and should not be changed

Values

Interpretation

(this is the logical index).

ind.gi.x.y.class positive integer Assigns the class (defined above) for this indicator.

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Permitted

Attribute

Values

Interpretation

ind.gi.x.y.physX IP 500: 0-159

IP 600: 0-319

ind.gi.x.y.physY IP 500: 0-79

IP 600: 0-159

ind.gi.x.y.physW IP 500: 1-160

IP 600: 1-320

ind.gi.x.y.physH IP 500: 1-80

IP 600: 1-160

For GraphicIcon type indicators, this is the x-axis location of the upper left corner of the indictor measured in pixels from left to right.

For GraphicIcon type indicators, this is the y-axis location of the upper left corner of the indicator measured in pixels from top to bottom.

For GraphicIcon type indicators, this is the width of the indicator measured in pixels.

For GraphicIcon type indicators, this is the height of the indicator measured in pixels.

4.6.1.16 Event Logging <logging/>

Warning!

Logging parameter changes can impair system operation. Do not change any logging parameters without prior consultation with Polycom.

The event logging system supports the following classes of events:

Level Interpretation

0 Debug only

1 High detail event class

2 Moderate detail event class

3 Low detail event class

4 Minor error - graceful recovery

5 Major error - will eventually incapacitate the system

6 Fatal error

Each event in the log contains the following fields separated by the | character:

• time or time/date stamp

• 1-5 character component identifier (such as “so”)

• event class

• cumulative log events missed due to excessive CPU load

• free form text - the event description

Polycom SoundPoint IP 600 SIP, SoundPoint IP 500 SIP Administrator's Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

1 Overview

2 Installation and Operation

2.1 Installation Models

2.2 Installation Process

2.2.1 Basic Network Setup

2.2.1.1 DHCP or Manual TCP/IP Setup

2.2.1.2 Local User Interface Setup Menus

2.2.1.2.1 Main Menu

2.2.1.2.2 DHCP Menu

2.2.1.2.3 Server Menu

2.2.1.3 Reset to Factory Defaults

2.2.2 Application Configuration

2.2.2.1 Centralized Configuration

2.2.2.1.1 FTP vs. TFTP

2.2.2.1.2 Configuration Files

2.2.2.1.3 Deploying a SoundPoint® IP boot server

2.2.2.2 Local Telephone Configuration

2.2.2.3 SIP Authentication Parameters

3 Features

3.1 Basic Features

3.1.1 Call Log

3.1.2 Call Timer

3.1.3 Call Waiting

3.1.4 Called Party Identification

3.1.5 Calling Party Identification

3.1.6 Configurable Feature Keys

3.1.7 Connected Party Identification

3.1.8 Context Sensitive Volume Control

3.1.9 Customizable Audio Sound Effects

3.1.10 Message Waiting Indication

3.1.11 Distinctive Incoming Call Treatment

3.1.12 Distinctive Ringing

3.1.13 Distinctive Call Waiting

3.1.14 Do-Not-Disturb

3.1.15 Handset, Headset, and Speakerphone

3.1.16 Local Contact Directory

3.1.16.1 Local Contact Directory File Format

3.1.17 Local Digit Map

3.1.18 Microphone Mute

3.1.19 Multiple Call Appearances

3.1.20 Shared Call Appearances

3.1.21 Customizable Fonts and Indicators

3.1.22 Soft Key-Driven User Interface

3.1.23 Speed Dial

3.1.24 Time and Date Display

3.2 Call Management Features

3.2.1 Automatic Off-hook Call Placement

3.2.2 Call Hold

3.2.3 Call Transfer

3.2.4 Three-Way Conference, Local

3.2.5 Call Diversion (Call Forward)

3.3 Audio Processing Features

3.3.1 Low-Delay Audio Packet Transmission

3.3.2 Jitter Buffer and Packet Error Concealment

3.3.3 Local Conference Mixing

3.3.4 Voice Activity Detection (VAD)

3.3.5 DTMF Tone Generation

3.3.6 DTMF Event RTP Payload

3.3.7 Acoustic Echo Cancellation (AEC)

3.3.8 Audio Codecs

3.3.9 Background Noise Suppression (BNS)

3.3.10 Comfort Noise Fill

3.3.11 Automatic Gain Control (AGC)

3.4 Presence and Instant Messaging Features

3.4.1 Presence

3.4.2 Instant Messaging

3.5 Localization Features

3.5.1 Multilingual User Interface

3.5.2 Downloadable Fonts

3.5.3 Synthesized Call Progress Tones

3.6 Advanced Server Features

3.6.1 Voicemail Integration

3.6.2 Multiple Registrations

3.6.3 Server Redundancy

3.6.3.1 DNS SIP Server Name Resolution

3.7 Accessory Internet Features