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MiVoice Office 250ENGINEERING GUIDELINES

RELEASE 6.2

NOTICE

The information contained in this document is believed to be accurate in all respects but is not warranted by Mitel Networks™ Corporation (MITEL®). The information is subject to change without notice and should not be construed in any way as a commitment by Mitel or any of its affiliates or subsidiaries. Mitel and its affiliates and subsidiaries assume no responsibility for any errors or omissions in this document. Revisions of this document or new editions of it may be issued to incorporate such changes.

No part of this document can be reproduced or transmitted in any form or by any means - electronic or mechanical - for any purpose without written permission from Mitel Networks Corporation.

For additional information and/or technical assistance in North America, certified technicians may contact:

Mitel Networks CorporationTechnical Support Department (USA)

1146 North Alma School RoadMesa, AZ 85201

1-888-777-EASY (3279)

Trademarks

The trademarks, service marks, logos and graphics (collectively "Trademarks") appearing on Mitel's Internet sites or in its publications are registered and unregistered trademarks of Mitel Networks Corporation (MNC) or its subsidiaries (collectively "Mitel") or others. Use of the Trademarks is prohibited without the express consent from Mitel. Please contact our legal department at [email protected] for additional information. For a list of the worldwide Mitel Networks Corporation registered trademarks, please refer to the website: http://www.mitel.com/trademarks.

MiVoice Office 250 Engineering GuidelinesRelease 6.2June 2016

®,™ Trademark of Mitel Networks Corporation© Copyright 2016, Mitel Networks Corporation

All rights reserved

Table of Contents

iii

Chapter 1: New Features

What’s New in this Release? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2

MiVoice Office 250 Release 6.2 New Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2

system park . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2

Import Audio Files for Audiotex Recording Enhancement . . . . . . . . . . . . . . . . . . . . . . . . . . . 2

MiVoice 5624 Wireless Phone Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

Discontinuation of support for CS platform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

Other Enhancements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

More Features in Release 6.2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

MOH (Music on Hold) for parked destinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

Group Pickup Enhancement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

Other Enhancements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

MiVoice Office 250 v6.1 SP1 New Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

Engineering Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

multi-Node video support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

SIP Phone Group – Password Alarm . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

MiVoice Office 250 Release 6.1 New Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

Product Re-branding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

New PS1 Hardware Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

MiVoice Office 250 Attendant Console Refresh . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

Digital Telephone Refresh . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

Virtualized PS1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

Video Support for SIP to SIP Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

BT CLIP CLID Enhancements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

SIP ACD Agent Login . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7


DTMF Decoding Payload . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

Propagation of Original Caller ID to P-Asserted Identity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

Do not use Out-bound Username in REGISTER . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8

Support for new DDM16 and DEM16 circuit board revisions . . . . . . . . . . . . . . . . . . . . . . . . . 8

MCD Interoperability Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8

Microsoft Lync Integration Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8

Support for New Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8

UPS Monitoring Support Changes on 6.0 SP2 and later software . . . . . . . . . . . . . . . . . . . . . 8


Engineering Guidleines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8

Mid Call Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

MBG SIP Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

Chapter 2:

MiVoice Office 250 Engineering Guidelines

iv

Engineering Guidelines

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

When to Add a Processor Expansion Card . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

Oversubscription . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .14

DSP Resource Costs and Limits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .15

System Calculator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .16

IP Phone Percentage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .16

High Priority vs. Shared . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .17

High Priority Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .17

Shared Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .17

Phones to Devices Percentage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .18

IP Phones to Shared Resources . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .18

When to add a Processing Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

High Call Traffic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .20

Hunt Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .21

All-Ring Hunt Group Members . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .21

All-Pages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .21

Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .21

Trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .21

IP Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .21

Hub Node in a Networked System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .21

Voice Mail Ports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .22

Enhanced Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .22

Processing Server Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .23

Using the System Calculator to Determine the need for a Processing Server . . . . . . . . . . . . .23

Call Load (calls per hour) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .24

OAI Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

Heavy Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .26

Light Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .26

CT Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .26

OAI Applications Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .26

UVM Coexisting with NuPoint UM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

Determining the Correct Compact Flash-Type Card Size. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

Basic vs. Advanced Ad Hoc Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31

Table of Contents

v

Video Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Supported devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Peripheral Devices for MiVoice Conference/Video Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

High Definition Multimedia Interface Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Ethernet Cameras . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Codecs compatibility matrix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

Bandwidth usage considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

Video phone bandwidth limitation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

Network configuration and bandwidth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

Bandwidth requirements for MiVoice Conference/Video Phone . . . . . . . . . . . . . . . . . . . . . . . . 35

General Information on IP Networking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Video Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Database Programming . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

SIP clients configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

Soft client camera settings to control bandwidth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

MiVoice Conference/Video Phone settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

MiVoice Border Gateway configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Teleworker Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Bandwidth requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Recommendations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

Mid-Call Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

Impacts to System Performance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

System Limitations and Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

System Calculator. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42

System Calculator Inputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42

System Calculator Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44

System Calculator Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44

System Recommendation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44

Phones to Devices Percentage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

IP Phones to Shared Percentage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46

Digital HX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46

IP Phone HX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47

IP Trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48

IP Phones/SIP Trunks HX w/ PEC-1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49

Large number of IP Phones, Trunks and UVM ports HX w/ PEC-1 . . . . . . . . . . . . . . . . . . . 50


vi

Digital HX w/ PS-1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .51

Digital and IP Phones HX w/ PS-1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .52

Fully-Loaded Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .53

Exceeds All System Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .54

Chapter 1

NEW FEATURES


2

WHAT’S NEW IN THIS RELEASE?

This section briefly describes the new features, hardware, and enhancements available on the MiVoice Office 250.

MIVOICE OFFICE 250 RELEASE 6.2 NEW FEATURES

SYSTEM PARK

System Park is a feature that allows calls to be parked by a phone, then either retrieved again by that phone, or retrieved by another phone.

• The Park/Pickup key with the same destination may be configured on several phones.

• then any of these phones can park or retrieve calls from the same destination.

• Multiple calls may be parked on the same destination.

• If in a call, the call will be added to the queue of parked calls.

• If not in a call, the first parked call on this destination will be retrieved.

• Parked calls will recall the person who parked the call after a timeout

• Phantom destinations use the new Park Recall Timer (default 180s, range 30-600s)

• Hunt Groups use the existing Hunt Group Recall Timer (default 180s, range 1-65'535s)

• If a call is parked on a Park/Pickup key, the key will flash.

• except if the Hunt Group has the ‘Group Call Pick-up’ flag disabled.

• Additional Notes:

• Phones without a Park/Pickup key can use the features 'transfer' and 'reverse transfer' to park and retrieve calls from destinations like "Park Location 1". Therefore it is rec-ommended to give simple numbers to such Phantom or Hunt Group destinations.

• Assigning a Park/Pickup key to a Hunt Group destination is an enhancement to the existing Group Pick-up feature, allowing Group Pick-up using a single key press, whilst also allowing for a visual indication.

IMPORT AUDIO FILES FOR AUDIOTEX RECORDING ENHANCEMENT

Audiotex Recordings are custom recordings used by the MiVoice Office 250 Unified Voice Messaging (UVM) applications like the Auto Attendant (AA) and Call Routing Announcement (CRA) features.

Until now, Audiotex recordings could only be recorded using a phone with access to the Voice Mail System Administrator’s Mailbox. Each recording is associated with a recording number and assigned to the application(s) in Database Programming.

From release 6.2, you can now import audio files and use them for Audiotex applications, including Auto Attendant (AA) and Call Routing Announcement (CRA) features. This will improve the quality and fidelity of the recordings and allow pre-recorded, professional recordings for auto attendant greetings and announcements.

New Features

3

Recording requirements are as follows:

• At this time, only WAV files are supported.

• Recordings must be less than 5 minutes in length.

MIVOICE 5624 WIRELESS PHONE SUPPORT

The MiVoice 5624 Wireless Phone seamlessly blends the wireless needs of the user with a wide range of critical applications like voice messaging and integrates with fixed legacy applications within the organization.

The MiVoice 5624 Wireless Phone is Mitel’s first VoWiFi (Voice over Wi-Fi) handset operating on 802.11n network; it supports the deployment of voice over a Wireless LAN (WLAN) without degrading the performance of an existing .11n network capacity. A MiVoice 5624 Wireless Phone-based solution delivers trouble-free WLAN vendor interoperability and scalability, as well as the capacity to integrate and communicate with a comprehensive range of external sources.

Notes:

• MiVoice 5624 Wireless Phone is already available outside North America.

• In North America the regulatory approval and introduction is in progress with MiVoice Busi-ness and will then also be available for MiVoice Office 250

• MiVoice 5624 Wireless Phone is Mitel’s rebranding of the earlier certified Ascom i62 and is technically the same. (Mitel SIP Center of Excellence specifications, 14-4940-00310).

DISCONTINUATION OF SUPPORT FOR CS PLATFORM

With Release 6.2, CS platform is not supported.

OTHER ENHANCEMENTS

• Security improvements for SSL connections (Webserver, SSH) used on the system.

• Open SSH 7.1

• OpenSSL 1.0.2a

• Web services are only accessible via TLS 1.2 encryption

MORE FEATURES IN RELEASE 6.2

MOH (Music on Hold) for parked destinations

To change from Ringback to MOH for Parked Calls, two configurations are needed:

1. Change the ‘Audio for Calls Ringing at this Device’, from Ringback to Music, at the Park Phantom(s)

2. Change the ‘Audio for Transfer to Ring’ from ‘Ringback’, to ‘Use Next Device’s Audio Source’, at the Trunk Group.”


4

Group Pickup Enhancement

Assigning a Park/Pickup key to a Hunt Group destination is an enhancement to the existing Group Pick-up feature, allowing Group Pick-up using a single key press, whilst also allowing for a visual indication.

Other Enhancements

MiVoice Office 250 Release 6.2 supports Exchange 2016.

MIVOICE OFFICE 250 V6.1 SP1 NEW FEATURES

ENGINEERING GUIDELINES

The Engineering Guidelines (this document) is now a stand-alone document. It was previously contained as an appendix in the MiVoice Office 250 Installation Manual.

MULTI-NODE VIDEO SUPPORT

MiVoice Office 250 Release 6.1 SP1 supports video calls between SIP clients connected to different nodes. A SIP device connected to Node 1 can now make a video call to a SIP device connected to Node 2.This is achieved by using the Peer-to-Peer (P2P) connection between two SIP endpoints for the video calls for the endpoints connected to MiVoice Office 250.

The MiVoice Office 250 Release 6.1 SP1 supports the following features.

• Video calls between SIP clients connected to different nodes.

• Backward compatibility. A P2P video call from 6.1 SP1 node to 6.1/6.0/5.1 node is modified to an audio call.

• SIP video devices as remote extensions via MBG.

• Video calls across the network like any other desktop device.

SIP PHONE GROUP – PASSWORD ALARM

MiVoice Office 250 Release 6.1 SP1 provides a new System Flag. This field is located in System\Flags.

When enabled, Alarm 149 will be generated each time a SIP Phone registers with the MiVoice Office 250, where the password used for the registration is either the same as the SIP Phone extension number or it is left empty. The alarm is raised against the SIP phone group to which the SIP Phone belongs. The SIP Phone will still able to register and operate, however the alarm will notify the Administrator that security needs to be reviewed immediately for this SIP Phone Group. This alarm will be cleared automatically by the system as soon as the SIP Phone registers again using new credentials. Note that the Administrator first has to make the password secure and cannot simply clear this alarm.

New Features

5

This feature is enabled by default. Mitel recommends that you do not disable this feature. The Alarms Notification Feature within System Administration and Diagnostics is used to send an email to an administrator when a System Alarm is generated.

MIVOICE OFFICE 250 RELEASE 6.1 NEW FEATURES

PRODUCT RE-BRANDING

Mitel has introduced new product naming as follows:

The re-branded interfaces for Database Programming, System Administration and Diagnostics, Administrator Web Portal, and the User Web Portal are updated with the new Mitel logo and color scheme. Additionally, the re-branding includes, but is not limited to, the following other MiVoice Office 250 interfaces:

• MiVoice Office 250 Installation Wizards

• MiVoice Office 250 Private Networking

• MiVoice Office 250 PS1 Server

NEW PS1 HARDWARE SERVER

The existing PS1 software supported hardware platform is limited to two server types the Dell R210 and Dell R220. Both server types are capable of running the 32-bit Debian 4.0 based distribution with the MiVoice Office 250 components.

As of May 2014, Dell retired the R210 Server and replaced it with the PowerEdge R220 Server.

MiVoice Office 250 PS1 software runs on the Dell R220 Server.

PREVIOUS NAME CURRENT NAME

Mitel 5000/5000 CP/Mitel 5000 Communications Platform

MiVoice Office 250

Mitel Database Programming MiVoice Office 250 Database Programming.

Mitel System Administration and Diagnostics MiVoice Office 250 System Administration and Diagnostics

Mitel 5000 Administrator Web Portal MiVoice Office 250 Administrator Web Portal

Mitel 5000 User Web Portal MiVoice Office 250 User Web Portal

Mitel Attendant Console MiVoice Office 250 Attendant Console

Mitel Applications Suite (MAS) MiCollab

Mitel Unified Communicator Advanced (UCA) MiCollab Client

Mitel Border Gateway MiVoice Border Gateway

NuPoint Unified Messaging MiCollab Unified Messaging

NOTEFor products related to releases prior to 6.1, naming will reference their previous, non re-branded names.


6

MIVOICE OFFICE 250 ATTENDANT CONSOLE REFRESH

Due to its historical heritage, earlier versions of the Attendant Console supported older operating systems such as Windows 98 and old drivers for HASP keys. These drivers do not support 64-bit versions of the Microsoft Windows.

As of 2014 new drivers for the Sentinel HASP are available from SafeNet Inc., and are compatible with a variety of 64-bit Windows versions, including Windows 7, Windows 8, and Windows 10.

The MiVoice Office 250 Attendant Console Release 3.5 can read data from valid HASP keys on the following Windows Operating systems:

• Windows 7 (32- and 64-bit)


• Windows 8.1 (32- and 64-bit)


DIGITAL TELEPHONE REFRESH

The Digital Telephone refresh redesigns the existing 8568 and 8528 Digital Telephone hardware to eliminate the potential for obsolete parts with the new redesigned Digital Telephones.

No new features are added to the refreshed series of Digital Telephones. To the end-user there are no differences in the installation and operation between the refreshed Digital Telephones and the previous versions.

The refreshed MiVoice 8568 and 8528 Digital Telephones are supported on all Mitel 5000 CP / MiVoice Office 250 systems that currently support these phones.

VIRTUALIZED PS1

The PS1 can now be deployed as a virtual appliance running in a VMware environment. Please see the MiVoice Office 250 Installation Manual for details regarding the installation and upgrade to virtualized PS1.

VIDEO SUPPORT FOR SIP TO SIP CALLS

MiVoice Office 250 Release 6.1 supports video capable SIP endpoints connected to the system when one SIP endpoint tries to establish a video call or escalate an established audio call to video. The peer-to-peer (P2P) functionality enables new SIP Peer endpoints to send and receive video packets directly to another IP device. It also allows support for Mid-Call escalation of SIP-to-SIP calls using signaling from voice calls to video.

In Database Programming, Peer-to-Peer Audio has been renamed Peer-To-Peer Media.

NOTE

Debian is a Linux distribution that is composed primarily of free and open-source software, most of which is under the GNU General Public License, and packaged by a group of individuals known as the Debian project. The Debian Stable distribution is one of the most popular for personal computers and network servers, and has been used as a base for several other Linux distributions.

New Features

7

BT CLIP CLID ENHANCEMENTS

MiVoice Office 250 Release 6.1 allows the decoding of Calling Line Identification (CLID) information in countries which use variations of BT CLIP (British Telecom Calling Line Identification Presentation) signaling on analog trunks.

BT CLIP supports the delivery of Caller ID information on analog trunks in the UK, and the standard is commonly known as BT CLIP. BT CLIP is an on-hook capability that provides the user with information about the caller before actually answering a call.

A new trunk Service Type and timer parameters are added in MiVoice Office 250 Database Programming to support BT CLIP.

SIP ACD AGENT LOGIN

SIP endpoints - including a SIP DECT, SIP Softphone, and a Mitel Phone Manager Softphone to login in to and logout from an ACD queue is supported.

SIP phones can be added to a Basic or UCD Hunt Group member list. Calls routed to a particular Hunt Group may be directed to a SIP phone.

For a SIP phone to use the ACD features (Log in, Log out, Wrap-Up Terminate), an OAI application must be used in conjunction with the SIP phone. Mitel Phone Manager is the recommended application.

The ACD feature codes cannot be dialled using a SIP phone.


DTMF DECODING PAYLOAD

SIP Peers in Database Programming can be configured with the necessary custom payload type value, depending on the supported payload type of the far end SIP provider. In previous releases, this value was configured in IP Settings. In Release 6.0 SP2 and up, the DTMF Decoding Payload is now located in the individual configurations for SIP Phone Groups, SIP Trunk Groups, and SIP Voice Mails. The payload type is typically negotiated between carriers and devices, but in some cases the SIP service provider does not accept requests. The DTMF Payload Type is made available by the SIP trunk service provider.

Database Programming now allows the DTMF Payload type to be selected for SIP Peers, from a range of values that correspond to those supported by the SIP service provider.

The default DTMF Decoding Payload value applies to North American, UK, and Australian systems.

PROPAGATION OF ORIGINAL CALLER ID TO P-ASSERTED IDENTITY

In Database Programming, a new flag, Do not use Out-bound Username in REGISTER, was added. This flag controls the registration of a SIP call when upgrading to Release 6.0 SP2.


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If the flag is set to No, the to/from/contact headers in the SIP register correspond to the out-bound username. If the flag is set to Yes, the to/from/contact headers in SIP register correspond to the trunk group Caller ID.

DO NOT USE OUT-BOUND USERNAME IN REGISTER

In Database Programming, a new flag, Do not use Out-bound Username in REGISTER, was added. This flag controls the registration of a SIP call when upgrading to Release 6.0 SP2.

SUPPORT FOR NEW DDM16 AND DEM16 CIRCUIT BOARD REVISIONS

New hardware revisions of the DDM16 and DEM16 circuit boards are fully supported by the Release 6.0 SP2 software.

MCD INTEROPERABILITY SUPPORT

The Mitel 5000 Release 6.0 SP2 maintains all current support with the published MCD Interoperability.

MICROSOFT LYNC INTEGRATION SUPPORT

The Mitel 5000 Release 6.0 SP2 now has support for Microsoft Lync via the Micollab 6.0 product. Refer to the MiCollab 6.0 documentation for details on this integration.

SUPPORT FOR NEW DEVICES

The Mitel 5000 Release 6.0 SP2 supports the following new devices:

• The 5610 IP Dect Phone is now supported via a Media Border Gateway connection to the Mitwl MiVoice Office 250.

• New models of the 5320e phone have a backlit display. The Mitel 5000 v6.0 SP2 contains new firmware for the 5320e IP phones.

• Apple IOS support via a Media Border Gateway connection to the MiVoice Office 250. Refer to the Media Border Gateway and IOS documentation for details.

• Mitel Phone Manager (formerly by Xarios).

UPS MONITORING SUPPORT CHANGES ON 6.0 SP2 AND LATER SOFTWARE

Effective Release 6.0 SP2, monitoring capabilities for previously approved suppliers of UPS units cannot be guaranteed to be compatible with the UPS Monitoring feature. See the MiVoice Office 250 Installation Manual for details.


ENGINEERING GUIDLEINES

The Engineering Guidlines (this document) is now a stand-alone document. It was previously contained as an appendix in the MiVoice Office 250 Installation Manual.

New Features

9

MID CALL FEATURES

Mid-Call Features provide a way for a mobile user to perform hold, transfer, and conferencing features when the 5000 places a call via Dynamic Extension Express (DEE).

MBG SIP SERVICES

The MBG (Mitel Border Gateway) is a platform for the secure deployment of multiple network connectivity services in a number of network edge scenarios. New MGB SIP features include:

• SIP device support

• UCA application support

NOTEMid-Call Features are only supported with other networked MiVoice Office 250 nodes running Release v6.0 SP1 and higher software.


10

Chapter 2

ENGINEERING GUIDELINES


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INTRODUCTION

The MiVoice Office 250 is a versatile system that can be scaled to address the needs of a wide variety of small and medium-size businesses. To achieve this flexibility, the MiVoice Office 250 offers two optional pieces of hardware, a Processor Expansion Card (PEC-1) and a Processing Server (PS-1), that may or may not be recommended based on the customer’s specific installation needs.

Starting with the v4.0 software release and its elimination of the 5200, 5400, and 5600 designation (for both the CS Controller and HX Controller platforms), it is now possible to add a PS-1 without first having to add a PEC-1. This means that a PEC-1 and a PS-1 can be added to a system independently of each other.

This document provides engineering guidelines and configuration recommendations on the following topics:

• “When to Add a Processor Expansion Card“ on page 13

• “When to add a Processing Server“ on page 20

• “OAI Applications“ on page 26

• “UVM Coexisting with NuPoint UM“ on page 28

• “Determining the Correct Compact Flash-Type Card Size“ on page 29

• “Basic vs. Advanced Ad Hoc Conferencing“ on page 31

• “Mid-Call Features“ on page 40

• “System Calculator“ on page 42


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WHEN TO ADD A PROCESSOR EXPANSION CARD

The MiVoice Office 250 Base Server has a built-in Processor Module that provides the system with a limited base number of digital signal processor (DSP) resources. It is important to note that DSP resources can be used for IP resources (e.g., VoIPs), as well as for other various non-IP features. These DSP resources can provide the following functionality:

• Connect local IP phones to the system

• Connect remote IP phones to the system

• Connect IP trunks to the system

• Connect Unified Voice Messaging (UVM) ports to the system

• Connect SIP-based voice mail ports (e.g., NuPoint UM) to the system

• Connect IP networking trunks to the system

• T.38 Faxing

• File-Based Music-On-Hold

• Configuration Assistant

• Conference Assistant

• Dynamic Extension Express human-answer-supervision

• Caller ID receivers and transmitters

• Meet-Me Conferencing

• Ad Hoc Conferencing plus other conferencing-related features (Record-A-Call, Agent Help and Station Monitor).

• MId Call Features

For installations that require more DSP resources, adding an optional PEC-1 to the system “additively” increases the pool of available DSP resources. Note that adding a PS-1 does not increase the size of the pool. Also note that adding a PEC-1 to a system does not increase the call processing power of the system (i.e., the ability to handle high call traffic).

Depending on how a system is configured and/or the actual call traffic experienced on the system, a PEC-1 may or may not be required. The DSP resources required by a particular installation are dependent on several factors, including:

• The quantity and types of devices

• The number of concurrent calls


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For systems that are predominantly configured with digital telephones, a PEC-1 is not typically required. On the other hand, systems that contain a large number of IP phones, IP trunks, IP networking, or voice mail ports may require a PEC-1 to increase the pool of available DSP resources.

Beginning with the v5.0 software release, the system has the ability to provide built-in Meet-Me Conferencing functionality. This functionality is provided by the Processor Module and the PEC-1 through DSP resources. The system can provide up to 40 conference ports with a maximum conference size of 20 parties. The administrator can also configure the Ad Hoc Conference Type setting under the System\Conference-Related Information folder in DB Programming. This setting determines whether or not Ad Hoc Conferencing, Record-a-Call, Station Monitor, and Agent Help use the Advanced DSP resources or the traditional Basic DSP resources. When set to Advanced, the Ad Hoc Conference features share the 40 conference ports with the Meet-Me Conference feature, thus possibly reducing the number of available Meet-Me Conference ports; however, the Ad Hoc Conferencing party limit also increases from four parties to twenty parties. When set to Basic, the Ad Hoc Conferencing features share 20 ports of conferencing using the traditional Basic DSP resources, and the Meet-Me Conferencing feature has 40 ports of conferencing using the Advanced DSP resources; however, an Ad Hoc Conference is limited to four parties.

OVERSUBSCRIPTION

Historically, PBXs provided dedicated switching hardware and wiring for every device. Although this allowed all devices to be in use at the same time, much of the hardware was idle most of the time. IP telephony borrows the shared resource approach of data networks to switch calls more cost effectively. Looking back, even traditional PBXs shared things like trunks, voice mail ports, etc., so sharing is nothing new. In fact, more efficient sharing of resources typically results in direct cost savings. Note that a properly engineered system can be “oversubscribed” yet never encounter contention for shared resources.

Technically, an oversubscribed system means that there are not enough resources available for every device in the system to acquire a resource at the same time. To determine whether or not a system is oversubscribed, we have to consider the worst-case scenario of every IP device being on a call at the same time, as well as using other call-related features like File-Based Music-On-Hold or T.38 Faxing. Because there are a lot of variables that go into this calculation, determining the precise point at which a PEC-1 is required is not trivial. However, in most scenarios, the decision can be made fairly easily.

IP phones, IP trunks, IP networking, and IP voice mail typically consume IP resources only when they are actively on a call. Therefore, making certain that a system is not oversubscribed may not be necessary because most sites will unlikely use or need to use all IP devices simultaneously. However, sites that have call-critical scenarios (e.g., hospitals or call centers) are generally not good candidates for deploying oversubscription.

Some features of the system consume DSP resources without an active call. For example, the background music and paging features on an IP phone each use a DSP resource for the duration of the feature; therefore, if the site uses the background music feature for company conference calls or paging to a large number of IP phones, then the site might not be a good candidate for oversubscription.

It is important to note that the system does not guarantee that 40 ports of conferencing are always available. These ports are provided by the DSP resource pool; therefore, if the site


15

requires that 40 ports of conferencing are always available, then the site might not be a good candidate for oversubscription.

Note that although the MiVoice Office 250 provides peer-to-peer audio to allow one IP device to talk directly to another IP device, this peer-to-peer audio does not decrease the number of DSP resources necessary for a call. This is because the MiVoice Office 250 allocates a DSP resource for each IP device on a call, as the call might require an audio path back to the PBX when performing certain features (e.g., the hold feature). Therefore, using peer-to-peer audio does not help prevent oversubscription.

If a site is configured with a large number of IP devices, the System Calculator is the best way to calculate whether or not a PEC-1 is required. The System Calculator takes into account all types of DSP resources to provide an indication of how much, if any, the site is oversubscribed.

DSP RESOURCE COSTS AND LIMITS

Each type of DSP resource has a cost associated with it. DSP resources with a higher cost consume more of the DSP resource pool than DSP resources with a lower cost. The cost of a resource is based on several factors including the functionality that the resource provides and whether or not the resource requires compression.

Typically, local (on-site) IP phones do not require compression, so they have a lower resource cost than remote (off-site) IP phones. Therefore, a system can connect more local IP phones than remote IP phones. If a system has both local and remote IP phones, then the System Calculator may be required to determine whether a PEC-1 is necessary based on the total number of IP phones.

Unified Voice Messaging (formerly Basic Voice Mail) that resides on a MiVoice Office 250 not equipped with a PS-1 requires compression in order to expand its recording capacity on the system’s compact flash-type memory card. UVM that resides on a PS-1 and NuPoint UM do not require compression due to the storage capacity of their hard drives, so they each have a lower resource cost than UVM that resides on a MiVoice Office 250 not equipped with PS-1. Because of this factor, a system equipped with a PS-1 may have a lower percentage of oversubscription compared to a system that is not equipped with a PS-1.

IP trunks, SIP trunks, and IP networking typically require compression, so they all have a higher resource cost. An IP trunk or IP networking call that is consuming a T.38 Fax resource has an even higher resource cost.

File-Based Music-On-Hold has the same resource cost as a local IP phone; therefore it has a lower resource cost. Note that File-Based Music-On-Hold allocates a resource from the resource pool on power-up and never releases the resource back to the pool, regardless of whether or not there is a device listening to the MOH.

A conference port has a resource cost that is similar to a local IP phone; therefore a Meet-Me Conference with 20 parties is equivalent to 20 local IP phones active on a call.

The following table summarizes the DSP resource costs for various types of DSP resources.


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Mitel recommends consulting the System Calculator if one of the following is true:

• If the site has more than 50 devices that consume a resource with a lower cost.

• If the site has more than 25 devices that consume a resource with a higher cost.

• If the site count of lower-cost and higher-cost devices total more than 30.

SYSTEM CALCULATOR

The System Calculator can help to determine whether or not a system requires a PEC-1. It can also be used to determine how close the system is to requiring a PEC-1 to accommodate future growth.

IP PHONE PERCENTAGE

For most systems, it is not cost effective to have a one-to-one ratio of phones to trunks.

Apart from an inbound call center or outbound dialing center, a typical system does not require a large percentage of phones to be on an outside call at the same time. This same principle is carried over to the DSP resources.

Table 1: DSP Resource Costs

DSP RESOURCE RESOURCE COST

Local IP Phones + UVM (on PS-1) +NuPoint UM +File-Based MOHConference Ports

Lower

Remote IP Phones +UVM (on Base Server) +IP Trunks +SIP Trunks +IP Networking +T.38 Fax

Higher

Mid Call FeaturesMedia Stream Resources (MSRs) used for DTMF and IVR. (Note that DTMF MSR may be allocated on either the Main Processor Module or PEC-1; the IVR MSR can be allocated on the main processor module only, not on a PEC-1.)

Lower


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In determining whether or not a system requires a PEC-1, the System Calculator uses the Phone-to-Trunk percentage plus the required percentage of concurrent intercom (IC) calls to determine a recommended percentage of IP phones that can be on a call at the same time. Note that the required percentage for concurrent IC calls can be adjusted in the System Calculator to match the needs of the specific site.

HIGH PRIORITY VS. SHARED

In order to calculate the recommended percentage of IP phones that need an available resource, the System Calculator places the IP devices and features into two categories: high priority and shared. The high priority category contains devices that are required to have minimal to no contention for resources. The shared category contains the devices and feature that are more dynamic.

HIGH PRIORITY DEVICES

The system allocates File-Based MOH ports on system power-up and are never released; therefore, they inherently fall into the high priority category because they are guaranteed to require a DSP resource. IP trunks and voice mail ports are in the high priority category as well, because they need to handle times of peak system traffic with minimal to no contention for resources. This requirement is based on the fact that if a system is configured with 20 trunks and four ports of voice mail, then the system should be able to handle a peak load of 20 trunk and four voice mail calls.

SHARED DEVICES

IP phones, advanced conferencing features, and other features that use DSP resources fall into the shared category because during times of peak system traffic they are typically not all in use. This is because they are already bound by the number of configured trunks and voice mail ports. For example, a system with 20 trunks and 80 phones cannot place more than 20 outside calls. If all of the phones attempted to place a call, then 60 phones would camp on waiting for a trunk to free up.

Keep in mind that the MiVoice Office 250 does not actually reserve DSP resources for the high priority devices (except for File-Based MOH). The resource pool categories within the System Calculator exist only to determine the amount of resources the IP phones share among themselves and the remaining features.


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PHONES TO DEVICES PERCENTAGE

The System Calculator displays the percentage of phones to various devices to give the administrator an idea of the amount of oversubscription that inherently exists in the system. In the following example, 25% of the phones at most can be on an outside call, 10% of the phones at most can place a call to voice mail, 8% of the phones and trunks can call voice mail, and 23% of the phones and voice mail ports at most can be on an outside call.

IP PHONES TO SHARED RESOURCES

The IP phones and the system features are competing for DSP resources from the shared portion of the resource pool. Keep in mind that the pool does not actually reserve the high priority portion of the resource pool, but for the purposes of this calculation are removed to reduce the chance of having unavailable resources for trunks or voice mail during times of peak system traffic.

The following example is directly linked to the input data and Phone-to-Trunk figures from the previous section. In this case, the system does not require a PEC-1 for the HX Controller platform because 72% of the IP phones can be on a call at a given time. This percentage is more than the recommended 44%. The chart indicates that an HX Controller platform with a PEC-1 can support 100% of the IP phones on a call at a given time.


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It is important to note that because the System Calculator does not take every factor into account, the system may still require a PEC-1 to provide enough DSP resources for certain features (e.g., paging, all-ring hunt groups, background music, advanced conferencing features, etc.). These features need to be considered when making the final decision of whether or not the system requires a PEC-1.


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WHEN TO ADD A PROCESSING SERVER

The MiVoice Office 250 provides enough call control processing power to handle most typical-size systems with average call traffic. However, some installations may benefit from the additional processing power provided by a Processing Server (PS-1), formerly known as the CS-5600. Unlike the PEC-1 that is concerned primarily with voice channels and DSP resources, the PS-1 provides additional call control processing power. Systems that have one or more of the following are good candidates for a PS-1:

• High call traffic

• Hunt groups

• All-ring hunt groups

• All-page groups

• Large number of digital telephones

• Large number of trunks

• Large number of IP devices

• System is the hub (main) node in a networked system

• Large number of voice mail ports

• Large number of UVM mailboxes with an E-mail Synchronization level of Enhanced Integration

The 5000 CP v4.0 software release dissolved the former distinction between the system types 5200, 5400 and 5600 and provided a more flexible model such that a PEC-1 is no longer required with a PS-1. This means that a PEC-1 and a PS-1 can each be added to a system (HX Controller platform or CS Controller platform) independently of each other. A system can have neither, either, or both.

HIGH CALL TRAFFIC

If a site is a call center and/or handles a high volume of call traffic, then it should be equipped with a PS-1. Call traffic analysis should be based on the busiest hour of the day. The other factor to consider is that the system load is based on the quantity of distinct calls, rather than the duration of the calls. For example, ten one-minute calls require ten times as much call control processing as one ten-minute call. The following table defines the three ranges of call traffic:

Table 2: Call Traffic Examples

TRAFFIC CALLS PER HOUR EXAMPLES/CHARACTERISTICS

Low 0-99 • Small office (15-25 phones)

• Medical / Law office

Medium 100-499 • Medium size office (25-100 phones)

• Requires more than 25 trunks

High 500+ • Large size office (100+ phones)

• Call Center

• Requires more than 50 trunks


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HUNT GROUPS

If a site has a large number of hunt groups used concurrently (e.g., in a call center environment), then the site should be equipped with a PS-1. The number of hunt groups, combined with medium to high call traffic that rings into a hunt group, greatly increases the need for a PS-1.

ALL-RING HUNT GROUP MEMBERS

If a site has all-ring hunt groups containing more than 30 members, then the site should be equipped with a PS-1. The number of all-ring hunt groups, combined with medium to high call traffic that rings into an all-ring hunt group, greatly increases the need for a PS-1.

ALL-PAGES

If a site performs a lot of paging to page zones containing more than 30 destinations, then it should be equipped with a PS-1.

PHONES

If a site has more than 175 phones with medium call traffic, then it is a good candidate for a PS-1. If a site has more than 150 phones with high call traffic, then it should be equipped with a PS-1. This applies to digital telephones, IP phones, SIP phones, and single-line phones.

TRUNKS

If a site uses more than 50 trunks concurrently, then this could be a good indication that the site has high call traffic; therefore, the site should be equipped with a PS-1.

IP DEVICES

If a site has more than 150 total IP devices with medium call traffic, then it may be a good candidate for a PS-1. If a site has more than 100 total IP devices with high call traffic, then it should be equipped with a PS-1. IP devices include IP phones, SIP phones, IP trunks, IP networking, SIP trunks, and SIP voice mail ports.

HUB NODE IN A NETWORKED SYSTEM

If a site is equipped with multiple nodes, the hub (main) node may be a good candidate for a PS-1. A node is considered to be a hub node if it is a PSTN gateway for the other nodes in the network, or the node provides access to the rest of the networked nodes for more than one other node. If the call traffic into the hub node is medium to high, then the node should be equipped with a PS-1.


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In the preceding example, nodes 2-4 use node 1 as a PSTN gateway. Also, when nodes 2-4 call nodes 2-4, the call must go through node 1 to reach the other node. This typically would be the case if the site used PRI trunks instead of IP networking for private networking.

VOICE MAIL PORTS

The v5.0 software release increases the maximum number of UVM ports from 16 to 32 as follows:

• 24 UVM ports (maximum) on a standalone CS Controller (without a PS-1)

• 32 UVM ports (maximum) on a standalone HX Controller (without a PS-1)

• 32 UVM ports (maximum) on a CS or HX Controller equipped with a PS-1

NuPoint UM can also now co-exist with UVM. If a site has a combined voice mail port count of up to 32 with medium call traffic, Mitel recommends that the system be equipped with a PS-1. If a site has a combined voice mail port count of 24 or more with high call traffic, Mitel recommends that the system be equipped with a PS-1.

Due to the limited number of compressed IP resources available on the Processor Module in a CS Controller platform, Mitel recommends a PS-1 when equipping 32 ports of UVM. Note that the HX Controller platform has more compressed IP resources so it can provide enough resources for 32 ports of UVM; however, based on other factors like the number of IP phones, the HX Controller platform may still require a PS-1.

ENHANCED INTEGRATION

If a site has more than 100 mailboxes with the Enhanced Integration enabled, Mitel recommends that the system be equipped with a PS-1. Note that in this scenario, the UVM resides on the PS-1 to provide more call control processing power.

Mitel recommends that when choosing the E-mail Client Message Format or the Download Format for Mobile Web Page settings for E-mail Synchronization to use the default of value of WAV instead of the MP3 because the MP3 encoding requires more processing time. This can result in a delay in delivering the e-mail with the attached voice mail message when multiple messages need to be delivered at the same time. For systems requiring a large amount of MP3


23

encoding for e-mail synchronization, Mitel recommends that the system be equipped with a PS-1 to reduce the delay in delivering the message via e-mail.

PROCESSING SERVER GUIDELINES

The following table defines the system limits that require a PS-1 based on the system configuration or characteristics combined with the system call traffic.

USING THE SYSTEM CALCULATOR TO DETERMINE THE NEED FOR A PROCESSING SERVER

The System Calculator can recommend whether or not a system requires a PS-1 based on the input provided to the System Calculator. Please note that because the System Calculator does not know every possible detail of the system, the user of the calculator needs to consider all factors listed in the Engineering Guidelines when deciding whether or not to configure a system with a PS-1. Unlike the IP Resource Calculator, the new System Calculator has information about digital and analog devices and call load, so that it can better recommend whether or not a system requires a PS-1.

The System Calculator attempts to determine whether or not the system requires a PS-1 based on the number of configured devices and the call load (calls per hour). Remember that it is the total number of calls placed on the system and not the concurrent calls that matters most (i.e., one ten-minute call requires 1/10 of the processing power as ten one-minute calls).

The following table indicates the system limits that the System Calculator uses to determine whether or not Mitel recommends a PS-1.

Table 3: PS-1 Guidelines

SYSTEM CONFIGURATION (CALL TRAFFIC) LIMIT TO REQUIRE A PS-1

High Call Traffic 500+ Calls per hour

Number of Hunt Groups (medium/high) (30+/20+) hunt groups

All-Ring Hunt Group Members (medium/high) (30+/20+) members

Page Zone Members 30+ members

Phones (medium/high) (175+/150+) phones

Trunks (medium/high) (75+/50+) trunks

IP Devices (medium/high) (150+/100+) IP devices

Hub-Node in a Networked System (medium/high) 3+ nodes

Voice Mail Ports (medium/high) (32+/24+) combined VM ports

Enhanced Integration Mailboxes (medium/high) (100+/50+) Enhanced Integration



Phones (medium/high) (175+/150+) phones

Trunks (medium/high) (75+/50+) trunks

IP Devices (medium/high) (150+/100+) IP devices


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If the total number of phones and trunks exceeds 400, then Mitel recommends that the system be split into a multi-node system.

The following table indicates the system limits that the administrator should follow when determining whether or not to configure a system with a PS-1. Note that the System Calculator does not have this information to provide a recommendation.

CALL LOAD (CALLS PER HOUR)

The Call Load setting within the System Calculator provides information about how busy the system will be at peak times. The System Calculator uses this setting to determine the number of devices that Mitel recommends. In the following scenarios, changing the Call Load setting from Medium to High changes the PS-1 recommendation based on the number of IP phones and the call load.

Voice Mail Ports (medium/high) (32+/24+) combined UVM and NuPoint UM ports



Number of Hunt Groups (medium/high) (30+/20+) hunt groups

All-Ring Hunt Group Members (medium/high) (30+/20+) members

Page Zone Members 30+ members

Hub Node in a Networked System (medium/high) 3+ nodes

Enhanced Integration UVM Mailboxes (medium/high) (100+/50+) Enhanced Integration




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OAI APPLICATIONS

Open Architecture Interface (OAI) provides a great deal of flexibility by allowing an external application to customize system behavior for customer-specific behavior. With this flexibility comes the risk of an OAI application adversely impacting system performance.

Each OAI Application has a different effect on system performance based on the type of application. Combined with the OAI application type, the other factors that contribute to system performance are call traffic and the number of devices. For a system with very low call traffic or fewer than 30 total devices, system performance is not greatly affected regardless of the application type or number of applications. The following guidelines are for systems with more than 30 total devices or medium to high call traffic.

HEAVY APPLICATIONS

Some applications (e.g., Attendant Console, Unified Communicator Advanced, and Customer Service Manager) monitor the statuses of almost every device in the system or deal with moving calls throughout the system. These types of applications are considered heavy applications and can affect system performance with medium to high call traffic.

LIGHT APPLICATIONS

Some applications (e.g., UC Express) only monitor and control a single phone. These types of applications are considered light and do not affect system performance by themselves.

CT GATEWAY

CT Gateway is an application that runs on an external PC and provides the system with the following functionality:

• CT Gateway can connect to multiple systems in a multi-node deployment to make a multi-node system appear as a single node to a connecting OAI application. This function-ality eliminates the need for an OAI application to be node aware so that it does not have to manage the connections to multiple nodes and manage multi-node scenarios.

• CT Gateway offloads OAI processing from the MiVoice Office 250, thus allowing for more heavy applications to connect to a single system.

OAI APPLICATIONS GUIDELINES

The following table defines the recommended number of OAI applications based on the application type and call traffic. If the system exceeds the recommended number of connecting applications, then the system should be equipped with either a CT Gateway or a PS-1.

Table 6: OAI Applications Guidelines

OAI APPLICATION TYPE

MAXIMUM NUMBER OF APPLICATIONS

LOW CALL TRAFFIC

MEDIUM CALL TRAFFIC

HIGH CALL TRAFFIC

Heavy Applications (e.g., CSM, UC Advanced, and Attendant Console)

5 3 3


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Mitel recommends that no more than 30 heavy OAI applications connect to a system equipped with a PS-1.

Light Applications (e.g., UC Express) 500 500 250

Table 6: OAI Applications Guidelines

OAI APPLICATION TYPE

MAXIMUM NUMBER OF APPLICATIONS

LOW CALL TRAFFIC

MEDIUM CALL TRAFFIC

HIGH CALL TRAFFIC


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UVM COEXISTING WITH NUPOINT UM

The v3.0 software release added support for NuPoint UM. Since then Mitel has recommended that UVM and NuPoint UM not coexist on the same MiVoice Office 250. However, there is functionality that UVM can provide that is currently not offered on NuPoint UM, as well as functionality on NuPoint UM that currently is not offered on UVM. Depending on how the system is configured, the functionality that is only provided by UVM and NuPoint UM can coexist on the same MiVoice Office 250.

Below is a list of functionality that the UVM can provide while coexisting with NuPoint UM:

• Call Routing Announcement (CRA) – A UVM CRA can provide the following functionality for Hunt Group Announcements. Note that the CRA can provide digit translation function-ality, but Mitel recommends that the digit translation not be transferred to a mailbox.

- Queue Position – This allows the caller in an ACD hunt group queue to know their position in the queue.

- Approximate Wait Time – This allows the caller in an ACD hunt group queue to know their approximate wait time.

• Scheduled Time-Based Application Routing (STAR) – The UVM STAR can provide System Day/Night Mode routing. Note that STAR can route calls to a CRA that resides on NuPoint UM.

When UVM and NuPoint UM coexist on the same MiVoice Office 250, Mitel recommends that only NuPoint UM provide the voice mail, mailbox and auto attendant functionality. Below is a list of functionality that is not recommended when UVM and NuPoint UM coexist on the same system:

• Mailboxes on UVM

• Voice Mail Application on UVM

• Message Notification/Retrieval on UVM

• Auto Attendant on UVM

• Digit Translation to a Mailbox on UVM

• Networking UVM and NuPoint UM


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DETERMINING THE CORRECT COMPACT FLASH-TYPE CARD SIZE

The MiVoice Office 250 contains a compact flash-type memory card that provides storage for the various system features and functionality. The size of the card is directly proportional to the capacity of UVM voice data; however, the UVM voice data competes with the following for memory card space:

• Software/Firmware – The software/firmware is a fixed size and typically increases in size with each release of software to support new devices and functionality.

• UVM Language Prompts – UVM has default language prompts loaded during installation. Adding languages takes up more space on the card. You should upload only the languages that are required for a particular installation. Each set of language prompts consumes approximately four MB of storage.

• Database Backup – The database backup is variable in size and is based on the size of the database (e.g., the number of configured phone, trunks, off-node devices, etc.). The database backup consumes approximately four MB of storage.

• Log Files – The system has various log files that are stored on the memory card. Each log file has a maximum size, and some have a quantity associated with it (e.g., the Call Pro-cessing System log is five files of 102400 bytes each). Over time, each set of log files may reach the maximum size and quantity. Once the maximum size and quantity is reached, the system automatically deletes the oldest file and starts a new file. When debugging an issue, Mitel Technical Support may instruct the technician to bump up the size and quantity of certain log files to create a deeper history to help track down an issue. It is good practice to return the logging options back to their original values once deeper logs are no longer needed. A System Administration & Diagnostics tool can remove certain log files to help increase the amount of available compact flash-type card space.

• Diagnostic Freezes – Each time the system performs a diagnostic freeze, it creates a single zip file that contains all the log files combined with a history queue and device information. The system stores the last freeze on the card. Each diagnostic freeze con-sumes a variable amount of storage based on the log file sizes. If the diagnostic freeze contains audio captures, it requires a much larger amount of storage.

• File-Based MOH files – The number of files and the size of each file are based on the files the administrator uploads to the system. Because File-Based MOH files can potentially take up a large amount of space, upload only the necessary files to conserve storage space.

• Embedded Reporting Data – The Embedded Reporting feature added in v5.0 collects various information for a fixed amount of time (e.g., over a seven-day period). This infor-mation can be viewed through System Administration & Diagnostics. The size of the reporting data depends on three main factors: number of hunt groups, the number of mail-boxes, and the number of PSTN calls. The system automatically removes the oldest day’s data if the maximum number of records is reached. Also, once the maximum number of days is reached, the system deletes the data from the oldest day. The embedded reporting data can consume as much as four MB of data on a busy system.

• People Application Data – For the People application (for 5340/5360 phones) added in the v5.0 software release, the size of this data depends on how many phones are using


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the feature and the number of contacts created on each phone. Each phone can store up to 50 contacts.

The following chart provides an estimate of the amount of available UVM voice data storage based on the size of the compact flash-type card and the compression type. Note that the card size does not increase the available voice mail storage on a system equipped with a PS-1 because UVM uses the hard drive on the PS-1 to store the voice mail data. Note that the system may not be able to provide the stated hours for a compact flash-type card if the other dynamic storage data (e.g., File-Based MOH, log files, etc.) consumes more storage than normal.

These values are based on normal compact flash-type card usage of the areas mentioned on the previous page. Extreme use of File-Based MOH, system logging, Embedded Reporting, or a large database could result in voice mail storage that is less than the above stated values.

In the v5.0 software release, the amount of UVM voice data that the system can provide per compact flash-type card size has been adjusted from the v4.0 specifications to provide the system the ability to support the new v5.0 functionality and to dynamically grow in the areas mentioned above without impacting the size of the UVM voice data. Note that when upgrading from v4.0 to v5.0, these new specifications need to be considered. If 100 hours of voice mail storage will not supply enough storage for the customer, then the site must upgrade to the 2 GB memory card.

When the UVM voice data has reached 80% capacity, the system generates an alarm. If the UVM voice data reaches 100% capacity, the system generates an alarm, and UVM does not allow any new recordings to be created. A caller attempting to leave a message hears a prompt indicating that the mailbox is full.

Due to functionality added in the v5.0 software release, the MiVoice Office 250 requires a minimum 1 GB compact flash-type memory card. The v5.0 software release also supports a 2 GB card.

Table 7: Voice Mail Storage

COMPACT FLASH-TYPE CARD

SIZE

VOICE MAIL STORAGE (NON PS-1) USING G.729

(DEFAULT)VOICE MAIL STORAGE

(NON PS-1) USING G.726VOICE MAIL STORAGE

(WITH PS-1) USING G.711

1 GB 6000 minutes(100 hours)

1,500 minutes(25 hours)


2 GB 18,000 minutes(300 hours)




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BASIC VS. ADVANCED AD HOC CONFERENCING

The Meet-Me Conferencing feature provides voice-guided prompts that allow users to join a conference based on an access code. Meet-Me Conferencing uses the “Advanced” DSP resources to provide better voice quality and up to a 20-party conference. Each system can provide up to 40 conference ports for Meet-Me Conferencing.

Traditional Ad Hoc Conferencing uses the “Basic” DSP resources to create up to a four-party conference. Each system can provide 20 conference ports via the Basic DSP resources. Note that Ad Hoc Conferencing is not the only feature that uses the Basic DSP resources; Record-A-Call, Agent Help, and Station Monitor all share the pool of 20 Basic DSP resources.

Unlike the Advanced DSP resources that can dynamically provide DSP functionality (e.g., connecting IP devices, file-based MOH, advanced conferencing, etc.), the Basic DSP resources statically provide 20 ports of conferencing for features like Ad Hoc Conferencing, Record-A-Call, Station Monitor, and Agent Help.

The Ad Hoc Conference Type is a new database field that determines which set of DSPs the Ad Hoc Conferencing features use. When the Ad Hoc Conference Type is set to Basic, Ad Hoc Conferencing, Agent Help, and Station Monitor use the Basic DSP resources. When the Ad Hoc Conference Type is set to Advanced, these features use the Advanced DSP resources. The Advanced DSP resources provide better voice quality and up to a 20-party Ad Hoc Conference. The down side of using Advanced is that Meet-Me Conferencing, Ad Hoc Conferencing, Agent Help, Record-A-Call, and Station Monitor all share the pool of 40 conference ports via the Advanced DSP resources. The following table lists the advantages and disadvantages of Basic vs. Advanced.

The default value of the Ad Hoc Conference Type is Basic. Therefore, if the system requires an Ad Hoc Conference with more than four parties or better voice quality, and if sharing the DSP resources is not an issue, then set the Ad Hoc Conference Type to Advanced.

If a user is already in a Meet-Me Conference and a feature like Record-A-Call, Agent Help, or Station Monitor is started, the system always uses the Advanced DSP resources even if the Ad Hoc Conference Type is set to Basic. Therefore, it is still possible for the system to reduce the number of Meet-Me Conference ports available for parties in the conference.

Table 8: Ad Hoc Conference -- Basic vs. Advanced

BASIC ADVANCED

ADVANTAGE DISADVANTAGE ADVANTAGE DISADVANTAGE

20 ports for traditional conferencing-related features

40 ports for Meet-Me Conferencing

Average voice quality for traditional featuresa

a. The voice quality of Basic (traditional) conferencing in v5.0 is the same as previous versions of theMiVoice Office 250. The Advance conferencing via DSP resources has built-in conference controls likea dominant talker feature and echo cancellation that can provide better voice quality for conferences witha larger number of participants.

Better voice quality for traditional features

40 ports shared among all traditional conferencing- related features and Meet-Me Conferencing

Four-party Ad Hoc Conferences

20-party Ad Hoc Conferences


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VIDEO SUPPORT

This section provides general guidelines for video device deployment on the MiVoice Office 250, for release 6.1 and above. This section also describes the basic principle of how to deploy video devices on a single MiVoice Office 250 node.

PREREQUISITES

You should become familiar with the MiVoice Office 250, and if required, the MiVoice Conference/Video Phone (previously referred to as the UC360) and the MiVoice Border Gateway products’ installation and administration instructions.

All documents are available from http://edocs.mitel.com.

SUPPORTED DEVICES

MiVoice Office 250 video is support on following devices:

• MiVoice Conference/Video Phone designed for deployment in conference rooms.

• CounterPath Bria, a commercial SIP softphone.

• CounterPath X-Lite, a free of charge version of SIP softphone with limited video codec support.

PERIPHERAL DEVICES FOR MIVOICE CONFERENCE/VIDEO PHONE

Depending on the application, the MiVoice Conference/Video Phone may be connected to a number of peripheral devices such as HDMI displays, HDMI projectors, video cameras, keyboards and mice.

HIGH DEFINITION MULTIMEDIA INTERFACE DISPLAYS

The MiVoice Conference/Video Phone has a High Definition Multimedia Display (HDMI) Type 'A' interface that allows the device to be connected to HDMI flat panel display or an HDMI projector.

The MiVoice Conference/Video Phone supports wide screen displays and projectors with resolutions of up to WUXGA (1920 x1080p).

ETHERNET CAMERAS

When the MiVoice Conference/Video Phone is installed as a Video Conferencing Bridge solution, a local video camera must be provided. The MiVoice Conference/Video Phone is designed to interface to a HDTV 1 Megapixel (720p) camera via an Ethernet LAN connection. This camera will capture a video image from the conference room.

Ethernet Camera Requirements

Cameras that are suitable for use with the MiVoice Conference/Video Phone should meet the following requirements:

http://edocs.mitel.com


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• The camera should be IEEE 802.3af PoE compliant, allowing the camera to receive its power over the LAN connection.

• The camera must be ONVIF (Open Network Video Interface Forum) compliant. ONVIF defines a protocol for the exchange of information between different video devices. ONVIF compliance is intended to ensure interoperability between devices.

• The camera must support the H.264 video compression standard and it must also support video streaming via RTP over UDP.

• The camera should support video resolutions of up to 720p @ 30 fps (HDTV).

• The camera must provide a 10BASE-T/100BASE-TX Ethernet network interface.

For details on how to configure the camera and the L2 switch networking parameters, see LAN Connection Guidelines.

For details on settings, see the MiVoice Conference/Video Phone Engineering Guidelines.

CODECS COMPATIBILITY MATRIX

In order to establish a video call, both parties have to support a common video codec.

The table below shows the list of supported phones and codecs.

BANDWIDTH USAGE CONSIDERATIONS

Since all media will go from one SIP device to another, there will not be any extra resource/bandwidth usage, Therefore, there will not be a need to run the system resource calculation tools. In case of a P2P failure, the call will be handled as an Audio-only call, and existing resource calculations will provide the proper results for maximum number of supported calls, and bandwidth requirements.

VIDEO PHONE BANDWIDTH LIMITATION

These settings are found under the Cable/DSL menu under Settings -> Advanced -> System Settings -> Video Settings.

X-LITE BRIAMIVOICE VIDEO PHONE

X-Lite H.263

H.263+ (1998)

H.263

H.263+ (1998)

No common codec. No Video Call

Bria H.263

H.263+ (1998)

H.264 (base profile)

VP8

H.264 (base profile)

MiVoice Conference/Video Phone

H.264(base profile)

H.264(high profile)


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Three different bandwidth settings can be independently applied to the uplink and downlink connections:

• High – limits the bandwidth to 1.5 Mb/s

• Medium – limits the bandwidth to 1.0 Mb/s

• Low – limits the bandwidth to 512 kb/s

NETWORK CONFIGURATION AND BANDWIDTH

For more details please refer to MiVoice Conference/Video Phone Engineering Guidelines.

The following table provides peak bandwidth values for some configurations.

CONFIGURATION

AVERAGE BANDWIDTH REQUIREMENTS

PEAK REQUIREMENTS COMMENT

IP camera to MiVoice Conference/Video Phone

6 MB/s 100 Mb/s A real high bandwidth user. It’s a good idea to have 1 GB switch in the conf room in order to connect camera to the IP video phone

IP phone to IP phone with HD quality

2 x 1.5 Mb/s of bandwidth

12Mb/s Reference to camera config page

Teleworker 2 x 1.5 Mb/s to/from MBG

6Mb both directions You need to make sure your MBG has the correct resources in accordance to its engineering guidelines

4 party Video Conference

7 Mb/s (average) x Both directions (up and down)

30 Mbs x Both directions to conf. unit

Please refer to Video Bandwidth Required for a Four-Party Conference section in MiVoice Conference/Video Phone Eng. Guide.


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BANDWIDTH REQUIREMENTS FOR MIVOICE CONFERENCE/VIDEO PHONE

The following table lists the bandwidth requirements for the MiVoice Conference/Video Phone, (shown as the UC360).

GENERAL INFORMATION ON IP NETWORKING

For details on how to setup IP networking settings, see the MiVoice Conference/Video Phone Engineering Guidelines.

VIDEO CONFIGURATION

When placing SIP devices into a P2P Network Group, it should be noted that the MiVoice Office 250 assumes that all devices in the same Network Group have to use the same set of codecs. If a Network Group contains two devices that use different codecs, then these calls can result in no media. In this case, P2P media should not be configured.

The figure below shows an example of the MiVoice Office 250 configured with an MBG to create successful video calls.

CONFIGURATIONBANDWIDTH REQUIREMENTS(BASED ON 10 CALLS WITH EACH CALL LASTING 1 MINUTE)

UC360 + UC360, High quality 3,081 Mbps per participant

UC360 + UC360, Low quality 1,238 Mbps per participant

UC360 + UC360 + UC360, High quality 5, 574 Mbps for bridge,

2,920 Mbps per participant

UC360 + UC360 + UC360, Low quality 2,334 Mbps for bridge,

1, 150 Mbps per participant

UC360 + UC360 + UC360 + UC360, High quality

8,860 Mbps for bridge,


UC360 + UC360 + UC360 + UC360, Low quality

3,585 Mbps for bridge,



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DATABASE PROGRAMMING

For more information on configuring settings on the MiVoice Office 250, see the MiVoice Office 250 Features and Programming Guide.

SIP CLIENTS CONFIGURATION

To configure the SIP clients, obtain the SIP Server address name, User Name, and Login Name from the User and Device Configuration form.

Program the following if you are using MBG:

• For User Name, enter the SIP-side username.

• For Password, enter the SIP-side password.

• For SIP Registrar or Domain, enter the public IP address of MBG.

• For SIP Proxy (if applicable), enter the public IP address of MBG.

SOFT CLIENT CAMERA SETTINGS TO CONTROL BANDWIDTH

Based on bandwidth availability, you can optimize the amount of video data for the video phone to send over IP. To do this, you need to configure the device settings and select the appropriate resolution.

The following figure shows an example for Bria/Xlite. In the Preferences dialog, in Devices->Other Devices, select the camera you will use for video capture, then select the resolution. Bandwidth requirements for particular softphone codec/resolution combinations can be found in the softphone documentation.


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MIVOICE CONFERENCE/VIDEO PHONE SETTINGS

Before configuring SIP settings, ensure that MiVoice Office 250 programming is completed.

1. Launch Advanced Settings using the phone interface.

2. Press Settings.

3. Press Advanced.

4. Enter the password.

5. Select System settings->SIP settings.

6. Obtain the SIP Server address name, User Name, and Login Name from the User and Device Configuration form.

Program the following if you are using MBG:

• For User Name, enter the SIP-side username.

• For Password, enter the SIP-side password.

• For SIP Registrar or Domain, enter the public IP address of MBG.

• For SIP Proxy (if applicable), enter the public IP address of MBG.

The Video Phone provides the same functionality as the Conference Phone while also enabling remote presentation and multi-party video conferencing. This variant supports video conferences with a maximum of four parties natively within the Video Phone (that is, the Video Phone, plus three external participants). Make sure that MiVoice Office 250 Database Programming has Maximum Number of Calls = 4.


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For details on how to setup the MiVoice Conference/Video Phone, see the MiVoice Conference/Video Phone Engineering Guidelines and the MiVoice Conference/Video Phone Administration Guide.

MIVOICE BORDER GATEWAY CONFIGURATION

The following figure shows an example of a MiVoice Office 250 configuration with MBG and SIP clients.

For information about ICP configuration and SIP devices provisioning, see the MiVoice Border Gateway Installation and Maintenance Guide.

TELEWORKER CONNECTION

For more information about teleworker connections and configuration, see the MiVoice Border Gateway Installation and Maintenance Guide.

BANDWIDTH REQUIREMENTS

With average bandwidth requirements for single call around 1.5 Mb/s and peak at 12 Mb/s, teleworker sets have to have enough bandwidth available through to the local ISP, and QoS policy set on the local router, so that background e-mail download will not affect video quality.

When more than one teleworker wants to participate in a video conference, the person with the phone deployed on premises should organize the video conference call first. If not, the teleworker who hosts the meeting could run out of bandwidth.

Bandwidth required for particular softphone codec/resolution combinations could be found in the softphone documentation.


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RECOMMENDATIONS

When using video on teleworker devices, start with the device in standard resolution. It is not as bandwidth intensive as HD/HD Widescreen.

The phone deployed on premises must be the video conference leader.

For best video quality your LAN must support VLAN tagging and QoS. Your service provider must provide a SLA which guaranties upload and download speeds large enough to account for burst.


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MID-CALL FEATURES

Mid-Call Features (MCF) provide a way for a mobile user to perform hold, transfer, and conferencing features when the MiVoice Office 250 places a call via Dynamic Extension Express (DEE).

IMPACTS TO SYSTEM PERFORMANCE

MCF does not impact system performance, and requires the following two types of resources:

• DTMF receiver. Where there is a Dual T1/E1 module, an embedded DTMF receiver is required. A DSP resource is consumed for other modules.

• IVR prompts. A DSP resource is required when playing IVR prompts.

96 pre-defined DSP channels are available which can be used simutaneously with no impact to other tasks. With MCF enabled, the expectation is that the majority of office-bound users will not answer calls on their mobile phones. Where users are away from their workplace, and calls answered on their mobile phone, the only resource used is the DTMF detector DSP with no impact to other system resources. Where Dual T1/E1 modules with an embedded DSP are used, no other system resources are required. Typically IVR prompts are of short duration (one to two prompts per call) and skipped over by users. As a result, there are no system resource impacts, and a PS-1 is not required.

SYSTEM LIMITATIONS AND CAPACITY

When MCF is enabled, a DTMF receiver is required to detect the DTMF MCF activation sequence. The receiver is connected upon the call being answered, and disconnected only when the call is dropped.

Where Dual T1/E1 modules are installed on the MiVoice Office 250, DTMF receiver resources are allocated on the module itself and no system resources are consumed.

A system-shared DSP is required when other module types are installed (e.g. SIP, LS, or PRI/BRI/E&M on Single T1 module). If MCF is enabled and there are not enough DSP resources, a DEE call is twinned to the mobile phone successfully but the MCF IVR menu prompts will not be activated when requested.

When the MCF menu is activated (by the default activation sequence (* #)), a Media Stream Resource (MSR) is required to play the IVR prompts. the MSR requires one DSP channel, and this channel can be allocated on the main processor module only, and not on the PEC-1 extension board. Note that this DSP is allocated across all trunk modules, including Dual T1/E1. Plan to add extra DSP channels for MCF calls on Dual T1/E1 modules.

Table 9 on page 41 specifies the number of DSP channels consumed by the MCF feature as per the call scenario listed. DSPs consumed by other devices, and conferencing are not included.

Important: When ARS is used to place outgoing calls, a DSP resourced is consumed on a node where the CO trunk is located, and not where the DEE user is configured. Perform system engineering carefully. Add all the required concurrent MCF calls to the Total IP Phone numbers on a node where CO trunks are configured!


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Legend:

• Simple Call - a call to DEE answered by the mobile phone, no IVR in progress.

• Simple Call to DEE, MCF IVR menu is being played - a call to DEE answered by the mobile phone, IVR prompts are played.

• Consult Call/Transfer - a call to DEE answered by the mobile phone, MCF activated, and the call is put on hold. A new consult call is started. DSP is moved from held call to the new call to detect MCF DTMF sequence, to be able to confirm or cancel transfer.

• Ad hoc conference - a call to DEE answered by mobile phone, MCF is used to create an ad hoc conference; conference is successful, mobile phone is one of participants in this conference.

Table 9: DSP Channels Consumed by MCF

CALL SCENARIO DUAL T1/E1 SINGLE T1 LSM-2/4 BRM-2 SIP

Simple Call to DEE

0 1 1 1 1

Simple Call to DEE, MCF IVR menu is being played

1a

a. IVR VoIP DSP is used (main processor only).For all others: DTMF VoIP DSP is used (main processor board or PEC-1).0 in a cell means embedded Dual T1/E1 board resource is used.

1a 1a 1a 1a

Consult Call, Transfer Leg

0 1 1 1 1

Ad hoc conference

0 1 1 1 1


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SYSTEM CALCULATOR

With the v5.0 software release, the IP Resource Calculator has been revised and renamed as the System Calculator. Like the older IP Resource Calculator, the new System Calculator works in conjunction with the Sales Workbench or as a stand-alone application. The System Calculator, which is available on Mitel Online, recommends the appropriate system configuration based on the number of IP devices and other items configured in the system. Though this tool provides guidance on which system Mitel recommends based on the supplied input, it does not have knowledge of every aspect of the system, like call traffic (e.g., calls ring into a CRA or an all-ring hung group) and feature usage.

The System Calculator provides a system recommendation by determining whether or not a PEC-1 and/or a PS-1 is recommended. The System Calculator may even determine that the system exceeds the recommended limits of all configurations of the MiVoice Office 250. In this case, the site may require a multi-node system to supply the required system configuration and functionality.

SYSTEM CALCULATOR INPUTS

The new System Calculator has more inputs than the older IP Resource Calculator in order to provide better estimates. Below is a list of the all of the inputs:

• System Type – This is a drop-down list that indicates the platform type that is being con-figured: HX or CS.

• Call Load (calls per hour) – This is a drop-down that indicates the system call load (low, medium, or high). Mitel recommends using the medium setting unless enough knowledge of a system is known to choose low or high. This selection changes some of the system limits that determine whether or not a PS-1 or PEC-1 is recommended.

• Concurrent IC Calls – This is a drop-down list that indicates the required percentage of concurrent phones to be on an internal call (low 10%, medium 25%, or high 50%). Note that the System Calculator already assumes that the number of phones equal to the number of trunks can be on an outgoing call at the same time. This percentage is in addition to that number. Mitel recommends using the medium setting unless enough knowledge of a system is known to choose low or high. This selection changes the recommended IP Phone-to-Shared percentage that determines whether or not a PS-1 or PEC-1 is recommended.

• Total Phones – This is the total number of phones in the system. This value includes digital telephones, analog single-line phones, IP phones, and SIP phones. If you are using Audio and Web Conferencing (AWC), include the number of ports in this total.

• IP Phones (IP plus SIP)– This is the total number of IP phones. If you are using SIP phones and/or AWC, include the number of phones/ports in this total.

• Remote IP Phones (compressed) – This is the total number of IP phones that use com-pression. Normally this contains the number of IP phones that reside outside the customer’s LAN.

• Total Trunks – This is the total number of trunks in the system. This value includes all trunks: digital, analog, SIP, Networking, and IP Networking. The IP Networking trunks value is an estimate of the maximum number of concurrent IP Networking trunks that the system


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may use. If the data was imported from the Sales Workbench, make sure the IP Networking trunks are included in the total number of trunks.

• SIP Trunks – This is the total number of SIP trunks. If the system has MGCP trunks, then include them in this count.

• Maximum Concurrent T.38 Faxes – This is the estimated number of concurrent T.38 Faxes that the system may use. The maximum number of concurrent T.38 Faxes is six.

• Unified Voice Messaging Ports – This is the number of UVM ports configured in the system. (This number should not include the number of NuPoint UM ports.) Note that the maximum number of UVM ports has increased from 16 ports to 32 ports in the v5.0 software release. A PS-1 equipped system is recommended where lenghty voice messages are expected and email synchronization enabled.

• NuPoint Messenger Ports – This is the number of NuPoint UM ports configured in the system. Note that in v5.0 software release, UVM and NuPoint UM can coexist (with some restrictions) on the same MiVoice Office 250. The maximum number of NuPoint UM ports is 32.

• Filed-Based Music-On-Hold – This is the number of File-Based MOH ports configured on the system. The maximum number of File-Based MOH ports is five.

• Advanced Conference Ports – This is the maximum number of concurrent Advanced conference ports. The system offers conference functionality via the Advanced DSP re-sources, allowing up to 20 parties in an Ad Hoc Conference and up to 20 parties in a Meet-Me Conference. The maximum number of conference ports is 40.

• Maximum concurrent UWP sessions – This is the maximum number of User Web Portal (UWP) sessions that can occur concurrently. Based on the assumption that a UWP page can load, on average, in approximately 20 seconds, the Mitel 5000 supports up to 8 con-current UWP users, or a single administrator session with a background traffic load of up to 10 simultaneous calls. If an increase in background traffic load or number of concurrent UWP users is needed, then a PS-1 equipped platform is recommended. With a background traffic load of 50 simultaneous calls on a PS-1 equipped platform, the average UWP page load time is less than 4 seconds. The added PS-1 will not affect any call processing-related activities.

The following figure shows an example of the available inputs for the System Calculator.


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SYSTEM CALCULATOR OUTPUTS

When determining which configuration meets the customer’s needs, there are many factors that need to be considered. The System Calculator uses the input data to best produce a recommended system configuration. Because there are other factors that may go into this decision that are not supplied to the System Calculator (e.g., hunt group information and paging), the System Calculator can only supply a recommendation based on supplied data. The System Calculator also provides charts to graphically display the usages of various resources in the system.

SYSTEM CALCULATOR RESULTS

The System Calculator produces three basic outputs: a system configuration recommendation, the percentage of Phones-to-Devices chart and the percentage of IP Phones-to-Shared chart per system type.

SYSTEM RECOMMENDATION

The system recommendation indicates which system configuration Mitel recommends and which system or systems are not recommended.

In the following example, Mitel recommends the Mitel HX w/ PEC-1 system. The Mitel HX w/ PS-1 and Mitel HX w/ PEC-1 & PS-1 systems are also valid systems. The Mitel HX is not recommended.


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PHONES TO DEVICES PERCENTAGE

The System Calculator displays the percentage of phones to various devices to give the administrator an idea of the amount of oversubscription that may exist in the system. In the following example, 33% of the phones at most can be on an outside call at the same time, 27% of the phones at most can place a call to voice mail, 20% of the phones and trunks can call voice mail, and 26% of the phones and voice mail ports can be on an outside call.

IP PHONES TO SHARED PERCENTAGE

The IP phones and the system features are competing for Advanced DSP resources from the shared piece of the resource pool (see “Shared Devices“ on page 17). Keep in mind that the pool does not actually reserve the dedicated portion of resource pool, but for the purposes of this calculation are removed to help guarantee that a contention for resources does not occur for a trunk or voice mail call. The following example corresponds to the input data and the Phone-to-Trunk data on the previous page. In this case, the system requires a PEC-1 for the HX Controller platform because the System Calculator requires 50% of the IP phones be on a call at a given time and the non-PEC system can only provide 48%. The HX Controller platform with a PEC-1 can support 100% of the IP phones on a call at the same time.

It is important to note that because the System Calculator does not take every factor into account, the system may still require a PEC-1 to allow certain features to work properly (e.g., paging, all-ring hunt groups, advanced conferencing features, etc.). These features need to be considered when making the final decision on whether or not the system requires a PEC-1.

The System Calculator also provides the IP Phone/Shared chart for the CS Controller platform configurations. The Processor Modules on the HX and CS provide a different amount of DSP resources to the pool; therefore, the two charts will have different results. Note that the PEC-1 provides the same amount of DSP resources to the pool for each of the two platforms.


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EXAMPLES

This section contains examples from the System Calculator to demonstrate how the results are displayed based for various configurations.

DIGITAL HX

This following sample configuration is a fully digital system with 20 conferencing ports and eight UVM ports.


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In this example, the HX Controller platform is the perfect choice. The system does not require a PS-1 because the device count is low. The IP Phone/Shared chart indicates that features like Meet-Me Conferencing and voice mail can be used without any chance of resource issues because the chart indicates 100% for the HX Controller platform.

IP PHONE HX

This sample configuration contains some IP phones with 20 conferencing ports and eight UVM ports. This system does not require a PEC-1, but it does have a fair amount of IP phones.


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The device count is not high enough to require a PS-1, and the non-PEC-1 allows 87% of the IP phones and conference ports to be in use at the same time. This value is more than the recommended 50%; therefore, the HX Controller platform is the recommended system.

IP TRUNKS

This sample configuration does not contain IP phones, but it has some SIP trunks with 20 conference ports and eight UVM ports. This system barely requires a PEC-1 because the SIP trunks use compression and the need for 20 conference ports.


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Because there are no IP phones, the percentage of IP Phone-to-Shared consist of only the conference ports. Keep in mind that SIP trunks and voice mail ports are not part of the IP Phone-to-Shared percentage, but they both use compression. If the system could use the Concurrent IC Calls setting of Low (i.e., 10%), then this configuration could be deployed without a PEC-1.

IP PHONES/SIP TRUNKS HX W/ PEC-1

This sample configuration consists of IP phones, SIP trunks, and eight UVM ports. This system will require a PEC-1.


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The percentage of Phones-to-Trunks indicates that 25% of the phones can be on an outside call at the same time. The HX’s IP Phones-to-Share percentage is only 20%; therefore, this option is not recommended. The HX w/ PEC-1 system indicates that 100% of the phones can be on a call at the same time; therefore, the HX w/ PEC-1 is the recommend system.

LARGE NUMBER OF IP PHONES, TRUNKS AND UVM PORTS HX W/ PEC-1

This sample system consists of a large number of IP phones, trunks, and UVM ports.


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The DSP resources are oversubscribed, but they are still within the recommend percentage of 50% with a PEC-1. Because voice mail ports have a higher DSP resource cost, this configuration utilizes more of the high priority portion of the DSP resource pool leaving a very small resource pool for the non-PEC-1 configurations. This system is on the borderline of requiring a PS-1 based on the IP phone, trunk, and voice mail port count. If this system plans to grow or has high call load, then Mitel recommends the site be configured with a PS-1.

DIGITAL HX W/ PS-1

This sample configuration does not have IP phones or SIP trunks. It does not require a PEC-1, but it does require a PS-1 due to the large number of digital telephones.


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The percentage of IP Phones-to-Shared is 100% for the HX Controller platform; therefore, the system does not require a PEC-1. However, because of the 200 digital telephones, Mitel recommends that the system be equipped with a PS-1. Note that the MiVoice Office 250 no longer requires a PEC-1 when adding a PS-1.

DIGITAL AND IP PHONES HX W/ PS-1

This sample configuration consists of a large number of digital and IP phones. There are not enough IP phones to require a PEC-1, but the total number of phones requires a PS-1.


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Because the recommended system is an HX PS-1, the UVM ports reside on the PS-1, thus eliminating the need for compression. This allows the IP Phones-to-Shared percentage of 53% to be over the recommended percentage of 40%. If this system requires a greater percentage concurrent IC calls, then an HX PEC-1/PS-1 is recommended.

FULLY-LOADED CONFIGURATION

This sample system consists of a large number of IP phone, digital telephones, trunks, and UVM. This system requires both a PEC-1 and a PS-1.


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Based on the number of devices and the number of IP devices, this system clearly needs a PEC-1 and a PS-1. Because UVM resides on the PS-1, the percentage of IP Phones-to-Shared is slightly greater than the HX PEC-1 configuration.

EXCEEDS ALL SYSTEM REQUIREMENTS

In this example, the configuration consists of values that exceed the recommended limits of the MiVoice Office 250. In this case, Mitel recommends a multi-node configuration.


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