Avaya BCM 2.5 IP Telephony Configuration Guide

Return to Menu

Part No. P0937663 02

Business Communications Manager 2.5

IP Telephony Configuration Guide

The information in this document is subject to change without notice. The statements, configurations, technical data, and recommendations in this document are believed to be accurate and reliable, but are presented without express or implied warranty. Users must take full responsibility for their applications of any products specified in this document. The information in this document is proprietary to Nortel Networks NA Inc.

Trademarks

NORTEL NETWORKS is a trademark of Nortel Networks.

Microsoft, MS, MS-DOS, Windows, and Windows NT are registered trademarks of Microsoft Corporation.

All other trademarks and registered trademarks are the property of their respective owners.

P0937663 02.0

Contents

Preface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

Before you begin . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

Symbols used in this guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14

Text conventions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14

Acronyms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15

Related publications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

How to get help . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18

Chapter 1

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

IP telephones and VoIP trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

IP telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

VoIP trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

The IP telephony network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

Business Communications Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

M1-ITG . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

IP Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

Public Switched Telephone Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

Key IP Telephony Concepts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

Jitter Buffer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

QoS routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

Traditional telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

IP telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

H.323 terminals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

WAN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

LAN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

Chapter 2

Prerequisites checklist . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Network diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Network devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Network assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

Resource Assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

Keycodes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

Business Communications Manager System Configuration . . . . . . . . . . . . . . . . . . . . 29

IP Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

IP Telephony Configuration Guide

Chapter 3

Installing IP Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31

Preparing your Business Communications Manager for IP telephone registration . . . 31

Choosing a codec . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

Choosing a Jitter Buffer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

Installing and Configuring i2004 Internet Telephones . . . . . . . . . . . . . . . . . . . . . . . . . 34

Before Installing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

Using a 3-port switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

Connecting the i2004 Internet telephone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

Configuring the i2004 telephone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Troubleshooting an IP telephone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

Configuring DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Modifying settings for Nortel IP telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

Download Firmware to an i200X telephone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43

Deregistering DNs for online IP telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44

Deregistering offline DNs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

Moving IP telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46

Configuring the i2050 Software Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46

H.323 devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47

Adding H.323 devices to the system . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48

Chapter 4

VoIP Trunk Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51

Installing keycodes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51

Setting the Published IP Address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51

Determining the published IP address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52

Configuring media parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53

Configuring codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54

Setting Silence Compression . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54

Setting Jitter Buffers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55

Outgoing call configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56

Putting VoIP lines into a line pool . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57

Configuring the Access code for the Line Pool . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

Configuring DNs to access the VoIP line pool . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

Configuring a remote gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59

Configuring PSTN fallback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60

Setting up a VoIP schedule . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

Configuring routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63

Creating a destination code . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Configuring digits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67

Activating the VoIP schedule . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67

P0937663 02.0

Turning on QoS monitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68

Enabling PSTN fallback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70

Incoming call configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70

Assign a target line to the DN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70

Example configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71

On Business Communications Manager Ottawa . . . . . . . . . . . . . . . . . . . . . . . . . 72

On Business Communications Manager Santa Clara . . . . . . . . . . . . . . . . . . . . . . 73

Making calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74

Connecting an i200X telephone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75

Connecting an i200X telephone on the LAN . . . . . . . . . . . . . . . . . . . . . . . . . 75

Configuring NetMeeting clients . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75

Quality of Service Monitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77

Quality of Service Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77

Updating the QoS Monitor data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77

Port settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77

Using firewalls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78

Port settings for legacy networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79

Using a gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80

The call signalling method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80

Alias names . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80

Modifying the call signalling method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81

Gatekeeper call scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82

Chapter 5

Typical applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85

M1 to Business Communications Manager over IP trunks . . . . . . . . . . . . . . . . . . 85

MCDN functionality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86

Multiple Business Communications Manager systems

across VoIP trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86

Multi-location chain with call center . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87

Business Communications Manager to IP Telephones . . . . . . . . . . . . . . . . . . . . . 89

Appendix A

Networking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

Determining the bandwidth requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

Determining WAN link resources . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

Link utilization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

Network Engineering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92

Bandwidth Requirements on Half Duplex Links . . . . . . . . . . . . . . . . . . . . . . . . . . 93

Bandwidth Requirements on Full Duplex Links . . . . . . . . . . . . . . . . . . . . . . . . . . . 94

LAN Engineering Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95

WAN engineering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96

IP Telephony Configuration Guide

Additional feature configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96

Setting Non-linear processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96

Determining network loading caused by IP telephony traffic . . . . . . . . . . . . . 96

Other intranet resource considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

Implementing the network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

LAN engineering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

Getting the best results from your system . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

Further network analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

Components of delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

Reduce link delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100

Reducing hop count . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100

Routing issues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102

Post-installation network measurements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102

Appendix B

Silence compression . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103

Silence compression on Half Duplex Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103

Silence compression on Full Duplex Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105

Comfort Noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

Appendix C

Network Performance Utilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109

Ping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109

Traceroute . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109

Sniffer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109

Appendix D

Interoperability . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111

Speech Path Setup Methods . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112

Media Path Redirection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112

Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

Asymmetrical Media Channel Negotiation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

No feedback busy station . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

Symbol Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114

Appendix E

Quality of Service. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115

Setting QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115

Measuring Intranet QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116

P0937663 02.0

Measuring end-to-end network delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116

Measuring end-to-end packet loss . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117

Recording routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117

Adjusting ping measurements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118

Measurement procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118

Other measurement considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

Implementing QoS in IP networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120

Traffic mix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120

TCP traffic behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121

Business Communications Manager Router QoS Support . . . . . . . . . . . . . . 121

Network Quality of Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121

Network Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122

Quality of Service parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122

Fallback to PSTN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123

Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127

IP Telephony Configuration Guide

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Figures

Figure 1 Network diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

Figure 2 Set registration properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Figure 3 Global options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Figure 4 DHCP summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

Figure 5 DHCP range . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

Figure 6 IP Terminal status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41

Figure 7 Configuration menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42

Figure 8 Terminal status dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42

Figure 9 IP Terminal Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43

Figure 10 IP Terminal Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44

Figure 11 Deregister Offline DN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

Figure 12 i2050 Communications server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47

Figure 13 i2050 Switch type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47

Figure 14 H.323 Terminal list . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48

Figure 15 H.323 Terminal list dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49

Figure 16 Global IP settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52

Figure 17 Setting the Published IP address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53

Figure 18 Media parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54

Figure 19 Media Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55

Figure 20 Media parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56

Figure 21 Trunk/Line data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57

Figure 22 Line pool . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

Figure 23 Remote gateway list . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59

Figure 24 Remote gateway dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60

Figure 25 PSTN fallback diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

Figure 26 VoIP schedule . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62

Figure 27 VoIP Routing Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63

Figure 28 Route list . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64

Figure 29 Add route dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64

Figure 30 Add route dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Figure 31 Add destination code dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Figure 32 Normal schedule . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66

Figure 33 VoIP schedule . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67

Figure 34 Remote Gateway list . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69

Figure 35 Remote Gateway dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69

Figure 36 Example PSTN fallback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71

Figure 37 NetMeeting options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76

Figure 38 NetMeeting advanced options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76

Figure 39 Port Ranges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78

Figure 40 Port ranges dialog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79

IP Telephony Configuration Guide

Figure 41 Local gateway IP interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81

Figure 42 Business Communications Manager systems with a gatekeeper . . . . . . 82

Figure 43 M1 to Business Communications Manager network diagram . . . . . . . . . 85

Figure 44 Multiple Business Communications Manager systems network diagram 87

Figure 45 M1 to Business Communications Manager network diagram . . . . . . . . . 88

Figure 46 M1 to Business Communications Manager network diagram . . . . . . . . . 89

Figure 47 Calculating network load with IP telephony traffic . . . . . . . . . . . . . . . . . . 97

Figure 48 One Call on a Half Duplex Link Without Silence compression . . . . . . . . 104

Figure 49 One Call on a Half Duplex Link With Silence compression . . . . . . . . . . 104

Figure 50 Two Calls on a Half Duplex Link With Silence compression . . . . . . . . . 105

Figure 51 One Call on a Full Duplex Link Without Silence compression . . . . . . . . 106

Figure 52 One Call on a Full Duplex Link With Silence compression . . . . . . . . . . 107

Figure 53 Two Calls on a Full Duplex Link With Silence compression . . . . . . . . . . 108

Figure 54 Relationship between users and services . . . . . . . . . . . . . . . . . . . . . . . 115

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Tables

Table 1 Network diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Table 2 Network device checklist . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

Table 3 Network assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

Table 4 Resource Assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

Table 5 Keycodes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

Table 6 Business Communications Manager system configuration . . . . . . . . . . . 30

Table 7 IP telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

Table 8 Settings for IP telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Table 9 QoS status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77

Table 10 Voice over IP Transmission Characteristics for

Table 11 Bandwidth Requirements per BCM Gateway port for half-duplex links. . 94

Table 12 Bandwidth Requirements per BCM Gateway port for Full-duplex links. . 95

Table 13 Link capacity example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98

Table 14 Business Communications Manager 2.5 Product

Table 15 Engineering specifications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111

Table 16 Supported voice payload sizes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112

Table 17 Name comparison . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113

Table 18 Quality of voice service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116

Table 19 Site pairs and routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117

Table 20 Computed load of voice traffic per link . . . . . . . . . . . . . . . . . . . . . . . . . . 118

Table 21 Delay and error statistics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

unidirectional continuous media stream . . . . . . . . . . . . . . . . . . . . . . . . . . 93

Interoperability Summary 111

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Preface

This guide describes IP Telephony functionality for the Business Communications Manager 2.5 system. This includes information on Nortel IP terminals such as the i2004 phone and the i2050 software phone, H.323 terminals and H.323 trunks.

Before you begin

This guide is intended for installers and managers of a Business Communications Manager 2.5 system. Prior knowledge of IP networks is required.

Before using this guide, you must install and configure a Business Communications Manager 2.5 system.

This guide assumes:

• You have planned the telephony and data requirements for your Business Communications Manager 2.5 system.

• The Business Communications Manager 2.5 is installed and initialized and the hardware is working. External lines and terminals and sets are connected to the appropriate media bay modules on the Business Communications Manager 2.5.

• Configuration of lines is complete.

• Operators have a working knowledge of the Windows operating system and graphical user interfaces.

• Operators who manage the data portion of the system are familiar with network management and applications.

Refer to Chapter 2, “Prerequisites checklist,” on page 27 for more information.

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14 Preface

Symbols used in this guide

This guide uses these symbols to draw your attention to important information:

Caution: Caution Symbol Alerts you to conditions where you can damage the equipment.

Danger: Electrical Shock Hazard Symbol Alerts you to conditions where you can get an electrical shock.

Warning: Warning Symbol Alerts you to conditions where you can cause the system to fail or work improperly.

Note:

Alerts you to important information.

Tip: Tip Symbol Alerts you to additional information that can help you perform a task.

Text conventions

This guide uses these following text conventions:

angle brackets (< >) Represent the text you enter based on the description

bold Courier text

inside the brackets. Do not type the brackets when entering the command.

Example: If the command syntax is

ping <ip_address> ping 192.32.10.12

Represent command names, options and text that you need to enter.

Example: Use the dinfo Example: Enter

, you enter

command.

show ip {alerts|routes}

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italic text Represents terms, book titles and variables in command

syntax descriptions. If a variable is two or more words, the words are connected by an underscore.

Example: The command syntax

show at <valid_route>, valid_route

is one variable and you substitute one

value for it.

Preface 15

plain Courier text

Acronyms

This guide uses the following acronyms:

ATM Asynchronous Transfer Mode

BCM Business Communications Manager

CIR Committed Information Rate

DID Direct Inward Dialing

DOD Direct Outward Dialing

DIBTS Digital In-Band Trunk Signalling

DSB DIBTS Signalling Buffer

ITU International Telecommunication Union

IXC IntereXchange Carrier

IP Internet Protocol

Represents command syntax and system output, such as prompts and system messages.

Example: Set Trap Monitor Filters

ISDN Integrated Services Digital Network

LAN Local Area Network

LATA Local Access and Transport Area

LEC Local Exchange Carrier

MOS Mean Opinion Score

PCM Pulse Code Modulation

PPP Point-to-Point Protocol

PRI Primary Rate Interface

PSTN Public Switched Telephone Network

QoS Quality of Service

RTP Real-time Transfer Protocol

SNMP Simple Network Management Protocol

TCP Transmission Control Protocol

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16 Preface

UDP User Datagram Protocol

UTPS UNISTEM Terminal Proxy Server

VoIP Voice over Internet Protocol

WAN Wide Area Network

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Related publications

Documents referenced in the Business Communications Manager 2.5 IP Telephony Configuration Guide, include:

• Business Communications Manager 2.5 Installation and Maintenance Guide

• Business Communications Manager 2.5 Software Keycode Installation Guide

• Business Communications Manager 2.5 Telephone Features Guide

Preface 17

IP Telephony Configuration Guide

18 Preface

How to get help

Your local distributor can provide technical support for your Business Communications Manager system or have access to that information through a Technical Service Center (TSC).

If you require non-technical support, contact 1-800-4NORTEL (1-800-466-7835), choose option 3, Sales or Pre-Sales Support)

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Chapter 1 Introduction

IP Telephony provides the flexibility, affordability and expandability of the Internet to the world of voice communications. Business Communications Manager for VoIP gives you several critical advantages:

• Cost Savings. IP networks can be significantly less expensive to operate and maintain than traditional networks. The simplified network infrastructure of an Internet Telephony solution cuts costs by connecting IP telephones over the LAN wiring system and eliminates the need for dual cabling. Internet Telephony can also eliminate toll charges on site-to-site calls via global four-digit dialing. And, by using the extra bandwidth on your WAN for IP Telephony, you leverage the untapped capabilities of your data infrastructure to maximize the return on your current network investment.

• Portability and flexibility. Employees can be more productive because they are no longer confined by geographic location. IP telephones work anywhere on the network, even over a remote connection. With Nortel Networks wireless e-mobility solutions, your phone, laptop, or scanner can work anywhere on the network where a Nortel Networks Access Point is installed. Network deployments and reconfigurations are simplified, and service can be extended to remote sites and home offices over cost-effective IP links.

• Simplicity and consistency. A common approach to service deployment allows further cost-savings from the use of common management tools, resource directories, flow-through provisioning, and a consistent approach to network security. As well, customers can centrally manage a host of multimedia services and business-building applications from a central point via a Web-based browser. The ability to network existing PBXs using IP can bring new benefits to your business. For example, the ability to consolidate voice mail onto a single system, or to fewer systems, making it easier for voice mail users to network.

• Compatibility. Internet Telephony is supported over a wide variety of transport technologies. A user can gain access to just about any business system, through an analog line, Digital Subscriber Line, a LAN, frame relay, asynchronous transfer mode, SONET or wireless connection.

• Scalability. A future-proof, flexible, and safe solution combined with high reliability lets your company focus on customer needs, not network problems. Nortel Networks Internet Telephony solutions offer hybrid environments that leverage existing investments in Meridian and Norstar systems.

• Increased customer satisfaction. Breakthrough e-business applications help deliver the top-flight customer service that leads to success. By providing your customers rapid access to sales and support personnel via phone, the Web, and e-mail, your business can provide better customer service than ever before.

IP telephones and VoIP trunks

This guide describes two similar applications for IP telephony on the Business Communications Manager system: IP telephones and VoIP trunks. These applications can be used separately or together as a network voice/data solution.

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20 Chapter 1 Introduction

IP telephones

IP telephones offer the functionality of regular telephones, but do not require a hardwire connection to the Business Communications Manager. Instead, they must be plugged into an IP network which is connected to the LAN or WAN card on the Business Communications Manager. Calls made from IP telephones can pass over VoIP trunks or across a Public Switched Telephone Network (PSTN).

VoIP trunks

VoIP trunks allow voice signals to travel across IP networks. A gateway within the Business Communications Manager converts the voice signal into IP packets, which are then transmitted through the IP network. The device at the other end reassembles the packets into a voice signal.

The IP telephony network

This section explains the components of the Business Communications Manager system and the devices it interoperates with to create a network. Figure 1: Network diagram on page 21 shows components of the Business Communications Manager system.

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Inspect FORWARD Callers

MXP

Inspect FORWARD Callers

MXP

Figure 1 Network diagram

Chapter 1 Introduction 21

Router

Internet

Inspect FO

ARD Ca

llers

H.323 Device A

Gatekeeper

LAN A

IP set A

WAN

BCM A

Router

PSTN

Norstar set A

I2050 set A

Router

LAN B

BCM B

In spect FO

RWA

RD Callers

MXP

M1-ITG

Meridian set A

H 323 Device B

IP set B

Business Communications Manager

The Business Communications Manager is a key building block in creating your network. It interoperates with many devices including the M1 and any H.323 device. In this network diagram, the Business Communications Manager system is connected to devices through multiple IP networks as well as the PSTN. Multiple Business Communications Manager systems can be linked together on a network.

In Figure 1: Network diagram on page 21, BCM A is connected to a LAN (via a LAN card), a WAN (via a WAN card), and a PSTN (via Media Bay Modules). Through these networks, the system accesses other systems and network equipment connected to the network.

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22 Chapter 1 Introduction

M1-ITG

The Meridian 1 Internet Telephony Gateway lets the M1 communicate with H.323 devices including the Business Communications Manager. In Figure 1: Network diagram on page 21, telephones on the M1 such as Meridian set A can initiate and receive calls with the other telephones on the system across IP networks.

Telephones

The Business Communications Manager system can communicate using several different types of telephones, ranging from traditional analog and digital telephones to IP telephones and H.323 terminals.

Traditional telephones

While traditional telephones cannot be connected to the Business Communications Manager system with an IP connection, they can use VoIP trunks to make calls to other locations on a network.

IP telephones

Nortel IP telephones include the i2004, i2002 and i2050. These devices connect to the Business Communications Manager across an IP network through either a LAN or a WAN.

H.323 terminals

H.323 terminals include computers that have Microsoft NetMeeting installed, or other third-party devices. These terminals connect through the network to either the LAN or WAN card on the Business Communications Manager.

Gatekeeper

A gatekeeper tracks IP addresses of specified devices, and provides authorization for making and accepting calls for these devices. A gatekeeper is not required for the Business Communications Manager system, but can be useful on networks with a large number of devices. In Figure 1:

Network diagram on page 21, for example, when Norstar set A wants to call an H.323 device, and

Norstar set B is under the control of the gatekeeper, Norstar set A sends a request to the gatekeeper. The gatekeeper, depending on how it is programmed, provides Norstar set A with the information it needs to contact Norstar set B.

IP Network

In the network shown in Figure 1: Network diagram on page 21 several LANs and a WAN are shown. When planning your network, be sure to consider your other needs for a data network.

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Chapter 1 Introduction 23

WAN

A Wide Area Network (WAN) is a communications network that covers a wide geographic area, such as state or country. For Business Communications Manager, a WAN is any IP network connected to a WAN card on the Business Communications Manager system. This may also be a direct connection to another Business Communications Manager system.

LAN

A Local Area Network (LAN) is a communications network that serves users within a confined geographical area. For Business Communications Manager, a LAN is any IP network connected to a LAN card on the Business Communications Manager system. Often, the LAN can include a router that forms a connection to the Internet.

Public Switched Telephone Network

The Public Switched Telephone Network (PSTN) can play an import role in IP Telephony communications. In many installations, the PSTN forms a fallback route, so that if a call across a VoIP trunk does not have adequate voice quality, the call is routed across the PSTN instead. The Business Communications Manager also serves as a gateway to the PSTN for users on the system.

Key IP Telephony Concepts

In traditional telephony, the voice path between two telephones is circuit switched. This means that the analog or digital connection between the two telephones is dedicated to the call. The voice quality is usually excellent, since there is no other signal to interfere.

In IP telephony, voice quality between IP telephones can vary significantly from call to call and time of day. When two IP telephones are on a call, each IP telephone encodes the speech at the handset microphone into small data packets called frames, and sends the frames across the IP network to the other telephone where the frames are decoded and played at the handset receiver. If some of the frames get lost while in transit, or are delayed too long, the receiving telephone experiences poor voice quality.

Codecs

The algorithm used to compress and decompress voice is embedded in a software entity called a codec (COde-DECode).

Two popular Codecs are G.711 and G.729. The G.711 Codec samples voice at 64 kilobits per second (kbps) while G.729 samples at a far lower rate of 8 kbps.

Voice quality is better when using a G.711 CODEC, but more network bandwidth is used to exchange the voice frames between the telephones.

If you experience poor voice quality and suspect it is due to heavy network traffic, you can get better voice quality by configuring the IP telephone to use a G.729 CODEC.

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24 Chapter 1 Introduction

Jitter Buffer

Voice frames are transmitted at a fixed rate, because the time interval between frames is constant. If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many cases, however, some frames can arrive slightly faster or slower than the other frames. This is called jitter, and degrades the perceived voice quality. To minimize this problem, configure the IP telephone with a jitter buffer for arriving frames. This is how the jitter buffer works:

Assume a jitter buffer setting of five frames.

• The IP telephone firmware places first five arriving frames in the jitter buffer.

• When frame six arrives, the IP telephone firmware places it in the buffer and sends frame one

to the handset speaker.

• When frame seven arrives, the IP telephone buffers it and send frame two to the handset speaker.

The net effect of using a jitter buffer is that the arriving packets are delayed slightly in order to ensure a constant rate of arriving frames at the handset speaker. The disadvantage of using a jitter buffer is that the speech arrives delayed by the number of frames in the buffer. For one-sided conversations, this is not an issue. For conversations where one party tries to interrupt the other speaking party, it is annoying because by the time the voice of the interrupter reaches the interruptee, the interruptee has spoken (2*jitter size) frames past the intended point of interruption. In cases where very large jitter sizes are used, some users revert to saying "OVER" when they wish the other party to speak.

Possible jitter buffer settings, and corresponding voice packet latency (delay) for the Business Communications Manager system IP telephones are:

• None

• Small (.06 seconds)

• Medium (.12 seconds)

• Large (.18 seconds)

QoS routing

When it sends a voice frame onto the network, the IP telephone firmware places some header information on the frame.

The header contains the network address of the sending and receiving IP telephones, and a TOS (Type Of Service) byte, which contains a routing priority.

The IP telephone firmware establishes the TOS byte to the highest possible priority so that as the voice frame travels through the network, the routers it encounters give it higher routing priority than competing data frames resulting from file transfers, WEB downloads, e-mails, etc. This process of prioritizing data frames is Quality of Service (QoS) routing.

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Chapter 1 Introduction 25

The Business Communications Manager system does QOS routing, but if one or more routers along the network route do not support QOS routing, this can impact voice quality. Business Communications Manager system QoS can also be configured so that the system reverts to a circuit-switched line if a suitable QoS cannot be guaranteed.

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26 Chapter 1 Introduction

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Chapter 2 Prerequisites checklist

Before you set up VoIP trunks or IP telephones on a Business Communications Manager, complete the following checklist to ensure that the system is set up properly. Some questions do not apply to all installations.

Network diagram

To aid in installation a Network Diagram is needed to provide a basic understanding of how the network is configured. Before you install VoIP functionality, you must have a network diagram that captures all of the information described below. If you are configuring IP telephones but not IP trunks, you do not need to answer 1.d and 1.e.

Tabl e 1 Network diagram

Prerequisites Yes No

1.a Has a network diagram been developed?

1.b Does the network diagram contain any routers, switches or bridges with corresponding IP addresses and bandwidth values for WAN or LAN links?

1.c Does the network diagram contain IP Addresses and network locations of all BCMs?

1.d Answer this if your system will use IP trunks, otherwise, leave it blank: Does the network diagram contain IP Addresses of any other Voice over IP gateways desired to connect to?

1.e Answer this only if your system will use a gatekeeper, otherwise, leave it blank: Does the network diagram contain alias for any Gatekeeper that may be used?

Network devices

This section of the checklist contains questions about devices on the network such as firewalls, NAT devices, and DHCP servers.

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28 Chapter 2 Prerequisites checklist

Tabl e 2 Network device checklist

Prerequisites Yes No

2.a This section of the checklist contains questions about devices on the network such as

firewalls, NAT devices, and DHCP servers.

2.b Is the network using private IP addresses? If the network uses public IP addresses, complete 2.c. If the network uses private IP addresses, complete 2.d to 2.e.

2.c Are there enough public IP addresses to accommodate all IP telephones and the Business Communications Manager?

2.d Does the system have a firewall/NAT device, or will the BCM be used as a firewall/ NAT device?

2.e The Business Communication Manager has limited space fore firewall rules. If the Business Communications Manager is to be used as a firewall/NAT device, do the firewall rules fit within 32 input rules and 32 output rules?

2.f A hub-based core will not have suitable performance for IP Telephony. Does the network use a non-hub solution at its core?

Network assessment

This section ensures that the network is capable of handling IP Telephony and that existing network services are not adversely affected.

Tabl e 3 Network assessment

Prerequisites Yes No

3.a Has a network assessment been completed?

3.b Has the number of switch/hub ports available and used in the LAN infrastructure been calculated?

3.c Have the used and available IP addresses for each LAN segment been calculated?

3.d Has DHCP usage and location been recorded?

3.e Has the speed and configuration of the LAN been calculated?

3.f Has the estimated latency values between network locations been calculated?

3.g Bandwidth/CIR utilization values for all WAN links.

3.h Has the quality of service availability on the network been calculated?

Resource Assessment

You must allocate sufficient resources on the Business Communications Manager for IP telephony.

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Chapter 2 Prerequisites checklis t29

Tabl e 4 Resource Assessment

Prerequisites Yes No

4.a Has a Business Communications Manager Resource Assessment been performed

using the resource questionnaire in the Programming Operations Guide?

4.b Has an analysis been done to determine which DS-30 split is appropriate for the system, and has the DS-30 split been changed to 5/3 if necessary?

4.c Have all necessary media resources for IP trunks, clients, vmail or WAN dialup been assigned or dedicated?

Keycodes

All elements of VoIP trunks and IP telephony are locked by the Business Communications Manager keycode system.

Tabl e 5 Keycodes

Prerequisites Yes No

5.a Complete this question only if you are using VoIP trunks: Do you have enough VoIP

keycodes?

5.b Complete this question only if you are using IP telephones: Do you have enough IP client keycode?

Business Communications Manager System Configuration

Several sections of the the Business Communications Manager must be properly configured prior to activatio n o f I P telephony.

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30 Chapter 2 Prerequisites checklist

Tabl e 6 Business Communications Manager system configuration

Prerequisites Yes No

6.a Is the LAN functioning properly with the Business Communications Manager?

6.b Is the WAN functioning properly with the Business Communications Manager?

6.c Has a dialing plan been created, taking into account special considerations for IP telephony?

6.d Has a preference been established between pre-configured DNs and auto-assigned DNs? If the preference is for auto-assigned DNs, complete 6.e.

6.e Complete this question only if you are not using Auto-assign DNs. Have set DNs been programmed for the corresponding IP clients?

IP Telephones

Complete this section only if you have IP telephones.

Tabl e 7 IP telephones

Prerequisites Yes No

7.a Are IP connections and IP addresses available for all IP telephones?

7.b If DHCP is not being used, has all telephone configuration been documented and made available for telephone installers?

7.c Has telephone powering been provisioned?

7.d Do computers that will be using the i2050 soft phone meet the minimum system requirements, including headset?

7.e Are IP connections and IP addresses available for all IP telephones?

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Chapter 3 Installing IP Telephones

An IP telephone converts the voice signal into data packets and sends these packets directly to another IP telephone or to the Business Communications Manager. If the destination is an IP telephone, the arriving voice packets are converted to a voice stream which is routed to a speaker or headset. If the destination is the Business Communications Manager, the voice stream is routed to a circuit switched connection (telephone or line), or some form of gateway.

Note: IP telephones require an IP network to reach the Business Communications Manager. However, they do not need to use VoIP trunks to communicate beyond the Business Communications Manager. They can use any type of trunk, just as any other phone on the Business Communications Manager can.

Before setting up IP clients, you must enable keycodes for IP telephony. For information on entering keycodes, see the Keycode Installation Guide.

Preparing your Business Communications Manager for IP telephone registration

If this is the first time you are installing an IP telephone on this Business Communications Manager, you must activate terminal registration on the Business Communications Manager.

Note: For the simplest installation possible, set telephone Registration and Auto Assign DNs to ON, and leave Password blank. IP Telephones installed on the system will connect and boot-up without manual registration.

1 In Unified Manager, open Services, IP Telephony, and Nortel IP Terminals. Select the

General tab:

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32 Chapter 3 Installing IP Telephones

Figure 2 Set registration properties

2 Set Registration to ON to allow new IP clients to register with the Business Communications

Manager.

Note: For security reasons, set the Registration to Off when you are not registering telephones.

3 In the Password box, type a password. This is the password that installers must enter from the

IP terminals to connect to the Business Communications Manager. If this field is left blank, no password is needed to register an IP terminal.

Note: The default password is “bcmi”. It can be changed to any alphanumeric string that is 10 characters in length or less.

4 Set the Auto Assign DN box. If Auto Assign DNs is set to ON, the Business Communications

Manager system assigns a free DN to a set being registered instead of prompting the installer for the set DN. If registration and Auto Assign DNs are both set to ON, and the Registration password is blank, first-time-connected IP clients will be assigned a DN without requiring installer intervention. The system selects this number from the Norstar Digital Telephone Range.

5 In the Advertisement/Logo box, type a string of text characters. This message is displayed on

the first line of the IP terminal’s display. This string can be a maximum of 24 characters.

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Chapter 3 Installing IP Telephones 33

6 From the Codec menu, select a default Codec, or leave the Default Codec at Auto. This is the

Codec that is used if an IP telephone has not been configured with a preferred codec. For information on this, see “Choosing a codec” on page 33.

7 From the Jitter Buffer menu, select a Jitter Buffer level. For information on choosing a Jitter

Buffer, see “Choosing a Jitter Buffer” on page 33.

Choosing a codec

The default codec is used when an IP client has not been configured to use a preferred Codec (see the next section for individual IP client Codec settings). If the default Codec is set to AUTO, the Business Communications Manager will choose the appropriate CODEC when an IP client goes on a call. For example, if both endpoints of the call are I20XX telephones on the same subnet, the Business Communications Manager chooses G.711 for maximum voice quality. If the telephones are on different subnets, the Business Communications Manager will choose G.729 to minimize network bandwidth consumption by voice data packets.

For IP telephones, the Business Communications Manager supports both A and MU law variants of the G.711 CODEC, as well as the G.729 and G.723 CODECS.

• The G.711 CODEC samples the voice stream at a rate of 64Kbps (Kilo bits per second), and is the CODEC to use for maximum voice quality.

• The G.729 CODEC samples the voice stream at 8Kbps. The voice quality is slightly lower using a G.729 but it reduces network traffic by approximately 80%.

• The G.723 CODEC should be used only with third party devices that do not support G.729 or G.711.

Choosing a Jitter Buffer

A jitter buffer is used to prevent the jitter associated with arriving (Rx) voice packets at the IP telephones. The jitter is caused by packets arriving out of order due to having used different network paths, and varying arrival rates of consecutive voice packets.The greater the size of the jitter buffer, the "better sounding" the received voice is. However, voice latency (delay) also increases. Latency is very problematic for telephone calls, as it increases the time between when one user speaks and the user at the other end hears the voice. The administrator can adjust the default jitter buffer size to the following values:

• NONE: Minimal latency, best for short-haul networks with good bandwidth.

• AUTO: Business Communications Manager will dynamically adjust the size.

• SMALL: Business Communications Manager will adjust the buffer size, depending on

CODEC type and number of frames per packet to introduce a 60 millisecond delay.

• MEDIUM: 120 millisecond delay

• LARGE: 180 millisecond delay

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34 Chapter 3 Installing IP Telephones

Installing and Configuring i2004 Internet Telephones

The telephone can be configured by the end user or by the administrator. If the end user is configuring the telephone, the administrator must provide the user with parameters to configure.

A maximum of 90 IP telephones, softphones and H.323 units can be connected on the Business Communications Manager system.

Before Installing

Before installing the i2004 telephone, ensure that:

• the installation site has a 120V AC (60 Hz) outlet

• the installation site has a 10/100 BaseT Ethernet connection

• If you are not using the 3-port switch, you must have an additional 10/100 BaseT Ethernet

connection connecting you to the local Ethernet.

Using a 3-port switch

In an office environment where a LAN network already exists, most computers will already be connected to a LAN line. Using the Nortel Networks 3-port switch, the i2004 phone can be installed on the network along with the computer, without needing to add an extra cable to a hub. For more information, consult the i2004 documentation.

Connecting the i2004 Internet telephone

Caution: Do not plug the telephone into an ISDN connection. This can cause

severe damage to the telephone. Plug the telephone into only a 10/100 BaseT Ethernet connection

To connect the i2004 Internet telephone:

1 Connect one end of the handset cord to the handset jack on the telephone base. Connect the

other end of the handset cord to the handset.

2 Connect one end of a Cat-5 line cordwith RJ45 connectors to the line cord jack on the

telephone base. Connect the other end of the line cord to the Ethernet connection.

3 Plug the AC Power adapter into the base of the telephone, and plug the adapter into the AC

outlet.

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Chapter 3 Installing IP Telephones 35

Configuring the i2004 telephone

Depending on how you set up terminal registration, you may be prompted for the password and a DN for the telephone the first time the telephone is booted. For information on setting the registration settings, see “Preparing your Business Communications Manager for IP telephone

registration” on page 31.

1 If the telephone is not yet registered, and if a password was entered in the Terminal

Registration screen, the telephone prompts you for that password. Type the password.

2 If you set Auto Assign DN on the Business Communications Manager to OFF, the telephone

prompts you for a DN. Choose a DN for the telephone. Entering ‘0’ will cause the system to automatically assign a DN to the telephone.

Once registration has been completed, you do not need to go through the registration steps described above unless you deregister a terminal.

To access the local configuration menu on a i2004 telephone:

1 Restart the telephone by disconnecting then reconnecting the power. After about four seconds,

the top light flashes and ‘NORTEL NETWORKS’ appears on the screen.

2 When the greeting appears, press the four softkeys (located under the telephone’s display) one

at a time from left to right. These keys must be pressed one after the other within 1.5 seconds, or the telephone will not go into configuration mode. If ‘

Manual Cfg DHCP(0 no, 1 yes)

’ appears on the screen, you successfully accessed the

configuration mode.

Manual Cfg DHCP(0 no, 1 yes)

If ‘

’ does not appear, disconnect then reconnect the

power, and try to access the configuration mode again.

3 For each parameter, use the keypad to define values. Refer to Table 8 on page 35 for

information on each parameter. To type a decimal, press *.

Tabl e 8 Settings for IP telephones

DHCP 0 if not using a DHCP server to disperse IP addresses

1 if using a DHCP server If you choose to use a DHCP server rather than allocating static IP

addresses for the IP telephones, skip the remainder of this section. For information on setting up a DHCP server, see “Configuring DHCP”

on page 38.

SET IP The set IP must be a valid and unused IP address.

NETMASK This is the subnet mask

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36 Chapter 3 Installing IP Telephones

DEF GW Default Gateway on the network (i.e., the nearest router to the

S1 IP This is the Published IP address of the Business Communications

S1 PORT Set this to 7000.

S1 ACTION Set this to 1.

S1 RETRY COUNT Set this to 255.

S2 IP This is the Published IP address of the Business Communications

S2 PORT Set this to 7000.

S2 ACTION Set this to 1.

S2 RETRY COUNT Set this to 255.

telephone. The router for IP address W.X.Y.Z is usually at W.X.Y.1)

If there are no routers between the telephone and the Business Communications Manager network adaptor to which it is connected, (for example a direct HUB connection), then enter the Published IP address of the Business Communications Manager as the DEF GW.

If the IP telephone is not connected directly to the Published IP address network adaptor, set the DEF GW to the IP address of the network adaptor the telephone is connected to. For information on setting the published IP address of the Business Communications Manager, see

“Setting the Published IP Address” on page 51.

Manager.

After you have configured a telephone, it attempts to connect to the Business Communications Manager. The message ‘Locating Server’ appears on the display. If the connection is successful, the message changes to ‘Connecting to Server’ after about 15 seconds.

The IP telephone is completely booted once the date/time string is displayed on the telephone. Do not attempt to use the telephone while the string "Connecting to server" is displayed (initialization of the telephone may take several minutes).

Troubleshooting an IP telephone

If the system is not properly configured, several messages can appear:

• SERVER: NO PORTS LEFT--The Business Communications Manager has run out of ports. This message will remain on the display until a port becomes available and the telephone is powered down then powered up.To obtain more ports, you may need to install additional keycodes. See the Keycode Installation Guide.

• Invalid Server Address--The S1 is incorrectly configured with the IP address of a Business Communications Manager network adapter other than the published IP address.

• NEW SET--The telephone has not been connected to the Business Communications Manager before, and must be registered.

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Chapter 3 Installing IP Telephones 37

• Registration Disabled--The Registration on the Business Communications Manager is set to OFF.

Note: To see the configuration information of a telephone connected to the Business Communications Manager:

When the telephone is not on a call, press the EXPAND key (blue key at the bottom-right of the telephone) followed by the SERVICES key (key with image of a globe).

To see the Codec data for a telephone while it is on a call: Press the EXPAND key (blue key at the bottom-right of the telephone) followed by the SERVICES key (key with image of a globe).

Use the following strategies for troubleshooting an IP telephone:

• If an IP telephone does not boot, use VNC to access the Business Communications Manager. Find this file:

E:\NORTEL NETWORKS\Logs\Nnu\UTPS.log.

If the telephone is properly configured to reach the Business Communications Manager, you should see this message in the log: "Opening signaling channel for set index X [at A.B.C.D]" where A.B.C.D is the telephone's IP address.

If this entry is not present, the IP telephone is not connected to the Business Communications Manager. Double check the telephone configuration parameters. VNC into the Business Communications Manager, open a DOS window and try pinging the telephone. Check the configuration settings of any NAT server, DHCP server, firewall and routers between the telephone and the Business Communications Manager. For information on using Ping, see Appendix C,

“Network Performance Utilities,” on page 109.

• While signaling between the IP telephones and the Business Communications Manager uses

Business Communications Manager port 7000, voice packets are exchanged using the default RTP ports 28000 through 28255 at the Business Communications Manager, and ports 51000 through 51200 at the IP telephones. If these ports are blocked by the firewall or NAT, you will experience one-way or no-way speech paths.

• If the LAN traffic in your network environment is heavy, you may experience dropped voice packets. If this occurs, connect the Business Communications Manager and the telephones to a local network hub to avoid the network traffic.

• If an IP telephone does not display the text "Connecting to server" within two minutes after power up, the telephone was unable to establish communications with the Business Communications Manager. Double check the telephone's IP configuration, and IP connectivity to the Business Communications Manager (cables, hubs, etc.).

• When an IP telephone is connected for the first time, the contrast level is set to the default setting of 1. Most users find this value is too low. Therefore, after the telephone is operational, you can increase the contrast level by pressing Feature *7 at the telephone.

• If the connection between the IP client and the Business Communications Manager is slow (ISDN, dialup modem), change the preferred CODEC for the telephone from G.711 to G.729. (See “Settings for IP telephones” on page 35.)

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38 Chapter 3 Installing IP Telephones

Configuring DHCP

An alternative to manually configuring the IP sets is to use Distributed Host Control Protocol (DHCP) to automatically assign IP addresses to the IP sets. Before setting up DHCP using the information below, see the Business Communications Manager 2.5 Programming Operations Guide for more extensive information on DHCP.

Note: Do not enable DHCP on the BCM if you have another DHCP server on the

Do not ddfdf

To set up DHCP to work with IP terminals, ensure that DHCP is set up with the following settings:

1 In Unified Manager, navigate to Services/DHCP, and select the Global Options tab. The

2 Set the NORTEL IP Terminal Information box to:

Nortel-i2004-A, iii.jjj.kkk.lll:7000,1,250;iii.jjj.kkk.lll:7000,1,250.

network

Global Options screen appears.

iii.jjj.kkk.lll represents the published IP address. Be sure to include the period at the end of the string (1,250.). In Figure 3, 47.65.82.131 is the published IP address:

Figure 3 Global options

3 Click the Summary tab. The summary screen appears:

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Figure 4 DHCP summary

Chapter 3 Installing IP Telephones 39

4 Set the Status box to Enabled.

5 Open Services/DHCP/Local Scope/LANX, where LANX is a LAN that contains IP sets that

use DHCP. The Scope Specific Options Screen appears:

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40 Chapter 3 Installing IP Telephones

Figure 5 DHCP range

6 Set each of the following:

• Scope Status: set this to Enabled

• Default Gateway Field: set this to Published IP Address.

• Address Range: set the range of IP addresses the DHCP server dispenses.

7 Repeat Step 6 for every network adapter that IP sets use to connect to the Business

Communications Manager.

8 Restart all of the IP telephones.

Note: Whenever changes are made to the DHCP settings, telephones will retain

the old settings until they are restarted.

If the DHCP server is not properly configured with the Published IP address, the telephones will display “Invalid Server Address”. You must then correct the DHCP settings, and restart the telephones.

Modifying settings for Nortel IP telephones

Settings such as jitter buffers and codecs for the Nortel IP telephones including the i2050, i2002 and i2004 can be modified through the Unified Manager:

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Chapter 3 Installing IP Telephones 41

1 In the Unified Manager, open Services, IP Telephony, and click on Nortel IP Terminals. The

IP Terminal summary appears.

2 Click on the IP Terminal Status tab. Every IP telephone currently connected to the Business

Communications Manager occupies a row in the IP Terminal Status table.

Figure 6 IP Terminal status

3 Select the IP Terminal that you want to change the properties for.

4 Open the Configuration menu, or right-click anywhere on the listing for the terminal to bring

up the menu.

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42 Chapter 3 Installing IP Telephones

Figure 7 Configuration menu

5 From the menu, select Modify Codec/Jitter Buffer. The Terminal dialog box appears:

Figure 8 Terminal status dialog

6 From the Codec menu, select a Codec. Specifying a non-default CODEC for a telephone

allows you to override the general setting. You might for example want to specify a low bandwidth CODEC (g.729) for a telephone that is on a remote or busy sub-net.

7 From the Jitter Buffer menu, select a jitter buffer value. For a telephone that has poor network

connectivity to the Business Communications Manager, increase the jitter buffer size.

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Chapter 3 Installing IP Telephones 43

Download Firmware to an i200X telephone

Firmware is the software stored in the telephone. When the Business Communications Manager is upgraded with a new IP telephone firmware load, this firmware load will automatically be downloaded into the IP telephones when they next connect to the Business Communications Manager. You can use the "Force firmware download" option to force immediate download to a telephone, or if you suspect that a particular telephone has corrupted firmware.

To download firmware to a telephone:

1 In the Unified Manager, open Services, IP Telephony, and click on Nortel IP Terminals. The

IP Terminal summary appears.

2 Click on the IP Terminal Status tab.

Figure 9 IP Terminal Status

3 Select the IP telephone that you want to download firmware to.

4 Open the Configuration menu, or right-click anywhere on the listing for the terminal to bring

up the menu.

5 Select Force Firmware Download. A dialog appears asking if you want to confirm that you

want to proceed.

6 Click the Ye s button. The firmware download begins.

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44 Chapter 3 Installing IP Telephones

The system drops any active call on that telephone, and downloads a new firmware load into the selected telephones. The telephones will be unusable until the download is completed and the telephones have reset.

Note: In order not to saturate the IP network with download packets, the system will only download up to five IP telephones at any given time. Telephones requiring download will show a Unified Manager status of and the Unistem Trunk Proxy Server (UTPS) will initiate download as resources become available.

Deregistering DNs for online IP telephones

This command will deregister the selected telephones(s) from the Business Communications Manager, and force it to go through the registration process again. Any active calls are dropped.

To deregister a DN for a phone that is online:

1 In the Unified Manager, open Services, IP Telephony, and click on Nortel IP Terminals. The

IP Terminal summary appears.

Download Pending

2 Click on the IP Terminal Status tab:

Figure 10 IP Terminal Status

3 Select the IP Terminal with the DN you want to deregister.

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Chapter 3 Installing IP Telephones 45

4 Open the Configuration menu, or right-click anywhere on the listing for the terminal to bring

up the menu.

5 Click Deregister DN.

The Deregister DN option deregisters the selected telephone from the Business Communications Manager, and forces it to go through the registration process again. It will cause any active call to be dropped.

Deregistering offline DNs

To deregister the DN of an IP client that is offline:

1 In Unified Manager, open Services, IP Telephony, and click on Nortel IP Terminals. The IP

Terminal summary appears.

2 Click the Deregister Offline DN tab. The Deregister Offline DN screen appears:

Figure 11 Deregister Offline DN

3 In the Deregister DN field, type the number of the DN you want to deregister.

4 Press Enter.

The DN is deregistered.

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46 Chapter 3 Installing IP Telephones

Moving IP telephones

IP telephones retain their DN when they are moved to a new location. The following instructions apply to I200X phones, computers with the I2050 softphone installed, and H.323 devices.

To move an IP telephone without changing the DN:

1 Disconnect the power from the IP telephone or 3-port switch.

2 Disconnect the network connection.

3 At the new location, reconnect the network location and the power connection.

4 If the new location is on a different LAN or WAN from the old location, the subnet mask,

default gateway IP, S1 IP, and S2 IP may change. If this is the case, you must change the settings for the telephone. To do this, see “Connecting the i2004 Internet telephone” on page

34. Do not change the Set IP Address.

To move an IP telephone and change the DN:

1 Deregister the DN, using the instructions in “Deregistering DNs for online IP telephones” on

page 44.

2 Disconnect the network connection and the power connection from the telephone.

3 Reinstall the phone at the new location. For information on this, see “Connecting the i2004

Internet telephone” on page 34.

Configuring the i2050 Software Phone

The i2050 Software Phone allows a computer equipped with a soundcard, microphone, and headset to function as an IP terminal on the Business Communications Manager system. The i2050 Software Phone uses the computer’s IP network connection to connect to the Business Communications Manager. When you install the i2050 Software Phone, on-screen documentation walks you through the steps for installing the software.

To configure the i2050 Software Phone to connect to the Business Communications Manager:

1 Follow the installation steps for installing your i2050 Software Phone until the software

configuration utility appears, or select the software configuration utility from your Windows Start Menu. The utility opens to the Communications Server tab.

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Chapter 3 Installing IP Telephones 47

Figure 12 i2050 Communications server

2 Enter the Published IP address of the Business Communications Manager in the IP address

field.Iin the Port drop down menu, select BCM .

3 Select the Switch Type tab:

Figure 13 i2050 Switch type

4 Click on the BCM option.

5 If using a USB headset, enable it in the Select Sound Devices tab.

To further configure this device through Unified Manager, see “Modifying settings for Nortel IP

telephones” on page 40.

H.323 devices

An H.323 device is any device that uses the H.323 protocol. This is a protocol for Internet communications that uses audio, video or data signals.

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48 Chapter 3 Installing IP Telephones

Adding H.323 devices to the system

The Business Communications Manager can support devices that use the H.323 protocol. To install one of these devices:

1 In the Unified Manager, open Services, IP Telephony, and click on H.323 Terminals. The

H.323 terminal list appears.

Figure 14 H.323 Terminal list

2 In the Configuration menu, click Add DN. The H.323 Terminal List dialog appears.

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Chapter 3 Installing IP Telephones 49

Figure 15 H.323 Terminal list dialog

3 Type a valid DN, password, and the IP address of the terminal.

4 Click the Save button.

5 Connect the H.323 device to the network, and use the documentation available with that device

to install it.

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Chapter 4 VoIP Trunk Configuration

This chapter explains how to configure VoIP trunks on a Business Communications Manager. A VoIP trunk allows you to establish communications between a Business Communications Manager and a remote system across an IP network.

Note: VoIP trunks can be used for calls originating from any type of phone on the Business Communications Manager system. They are not strictly for use with IP terminals.

Configuring a VoIP trunk requires the following actions:

• Installing keycodes

• Configuring media parameters

• Outgoing call configuration

• Incoming call configuration

Note: If you are using the Business Communications Manager with an M1, you

must set up the system to be compatible with the M1. See Appendix D,

“Interoperability,” on page 111.

Installing keycodes

Before you can use VoIP, you must obtain and install the necessary keycodes. See the Keycode Installation Guide for more information.

Setting the Published IP Address

The published IP address is the IP address used by computers on the public network to find the Business Communications Manager. For example, if a Business Communications Manager has a LAN interface (LAN1) that is connected only to local office IP terminals and a WAN interface (WAN1) that is connected to the public network, then WAN1 should be set to the published IP address.

To set the published IP address:

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52 Chapter 4 VoIP Trunk Configuration

1 In Unified Manager, open Services and click on IP Telephony. The Global settings tab

appears.

Figure 16 Global IP settings

2 From the Published Address menu, select the appropriate network interface.

Determining the published IP address

Use the flowchart below to determine which card should be set as the published IP address. The flowchart makes reference to public and private IP addresses. The public and private IP addresses are concepts relating to Network Address Translation (NAT). The decision also depends on whether a Virtual Private Network (VPN) is enabled. For information on NAT, see the Business Communications Manager 2.5 Programming Operations Guide.

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Figure 17 Setting the Published IP address

Start

Chapter 4 VoIP Trunk Configuration 53

Is NAT enabled?

Is the Business Communications

Manager expected to connect to

devices on the public side?

Are all of your public side

devices using a VPN?

Do you anticipate the most VoIP

traffic on your public or private side?

Private

Set the network interface with

Public

the most anticipated VoIP traffic

as the Published IP address

Set the network interface on the

private side as the

published IP address

Set the network interface on the

public sideas the published IP address

Set the network interface on the

public sideas the published IP address

Set the network interface on

the private side as the

published IP address

If you using IP telephones on the network, they must be set to have the IP address of the network card they are connected to for their Default Gateway, and the Published IP address as the S1 IP address. For more information about this, see “Configuring the i2004 telephone” on page 35.

Configuring media parameters

There are three steps to configuring media parameters:

• Configuring codecs

• Setting Silence Compression

• Setting Jitter Buffers

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54 Chapter 4 VoIP Trunk Configuration

Configuring codecs

This section explains how to select the codecs that are used for VoIP trunks. For an explanation of codecs, see “Codecs” on page 23.

To configure the codecs:

1 In Unified Manager, open Services, IP Telephony, and click on H.323 trunks.

2 Click on the Media Parameters tab. The Media Parameters dialog appears.

Figure 18 Media parameters

3 Click the First Preferred Codec menu. Select the codec you want to use as the first preferred

codec. This is the most preferred codec to be used on VoIP trunks.

4 For each preferred codec, use the process described in step 3.

Setting Silence Compression

This section explains how to set silence compression on VoIP trunks.

The silence compression feature identifies periods of silence in a conversation, and stops sending IP speech packets during those periods. In a typical phone conversation, most of the conversation is half-duplex, meaning that one person is speaking while the other is listening. If silence compression is enabled, no voice packets are sent from the listener’s end. This greatly reduces bandwidth use.

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Chapter 4 VoIP Trunk Configuration 55

G.723.1 and G.729 support silence compression. If a conversation is using G.711, silence compression does not occur.

To set the silence compression:

1 In Unified Manager, open Services, IP Telephony, and click on H.323 trunks.

2 Click on the Media Parameters tab. The Media Parameters dialog appears:

Figure 19 Media Parameters

3 Click the Silence Compression drop-down menu, and select either Enabled or Disabled. If

you select Enabled, silence compression is only used when a G.729 or G.723.1 codec is in use.

Setting Jitter Buffers

This section explains how to select the jitter buffer size used on VoIP trunks.

Jitter buffers are explained in detail in “Jitter Buffer” on page 24.

To set the jitter buffer size for VoIP trunks:

1 In Unified Manager, open Services, IP Telephony, and click on H.323 trunks.

2 Click on the Media Parameters tab. The Media Parameters dialog appears:

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56 Chapter 4 VoIP Trunk Configuration

Figure 20 Media parameters

3 In the Voice Jitter Buffer, select a value.

Outgoing call configuration

This section explains how to set up your system to place calls through VoIP trunks. The system at the other end of the call must be set up to receive VoIP calls. For information on this, see

“Incoming call configuration” on page 70.

Before telephones on the Business Communications Manager can use VoIP trunks services, the VoIP trunks must be placed in a line pool. Outgoing call configuration consists of the following steps:

• Putting VoIP lines into a line pool

• Configuring the access code for the line pool

• Configuring DNs to access the VoIP line pool

• Configuring a remote gateway

• Configuring PSTN fallback

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Chapter 4 VoIP Trunk Configuration 57

Putting VoIP lines into a line pool

Lines 001 to 060 are reserved for VoIP trunks. However, they can be used only if you have entered the appropriate keycodes to activate them.

When putting VoIP trunks into a line pool, choose a line pool that is not used for any other type of line.

To put your lines into a line pool:

1 In Unified Manager, open Services, Telephony Services, Lines, VoIP lines, Line 001, and

click on Trunk/Line Data. The Trunk/Line Data screen appears.

Figure 21 Trunk/Line data

2 In the Line type field, set a line pool that is not used by any non-VoIP lines.

Repeat this procedure for as many trunk lines as you have keycodes for. You can use the same line pool for all VoIP lines.

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58 Chapter 4 VoIP Trunk Configuration

Configuring the Access code for the Line Pool

An access code is a number that users dial from their telephones to access the line pool containing VoIP trunks.

Note: Set up an access code for the line pool only if you are NOT planning to use PSTN fallback.

To set up access codes:

1 In Unified Manager, open Telephony Services, General Settings, Access Codes, Line Pool

Codes, and click on the line pool that you selected as the VoIP line pool.

The Pool screen appears.

Figure 22 Line pool

2 Enter a unique access code for this line pool. Ensure that no other line pools use this access

code.

Configuring DNs to access the VoIP line pool

Each DN that is to use the VoIP lines must be authorized to use the VoIP line pool:

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Chapter 4 VoIP Trunk Configuration 59

1 In Unified Manager, open Services, Telephony Services, System DNs, Assigned DNs, DN

XXX, Line Access, and click Line Pool Access. DN XXX is any DN that you want to allow to

use VoIP trunking.

2 Click Add.The Add Line Pool Access dialog appears.

3 Type the letter of the VoIP line pool.

4 Click Save.

Repeat this procedure for every DN you want to allow to use VoIP trunks.

Configuring a remote gateway

This section explains how to configure the Business Communications Manager to communicate with other Business Communications Managers and/or other VoIP gateways such as Meridian ITG. The remote gateway list must contain an entry for every remote system to which you want to make VoIP calls.

To add an entry to the remote gateway list:

1 In Unified Manager, open Services, IP Telephony, H.323 Trunks, and click on Remote

Gateway. The remote gateway tab appears.

Figure 23 Remote gateway list

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60 Chapter 4 VoIP Trunk Configuration

2 In the Configuration menu, select Add a new entry. The Remote Gateway window appears

Figure 24 Remote gateway dialog

3 In the Name field, type a name for the remote system.

4 In the Destination IP field, enter the IP address of the system.

5 In the Destination Digits field, set the leading digits which callers can dial to route calls

through the remote gateway. Ensure that there are no other remote gateways currently using this combination of destination digits. If multiple leading digits map to the same remote gateway, separate them with a space. For example, 7 81 9555.

6 Set the QoS monitor option. If you intend on using fallback to a PSTN line, set the QoS

monitor to enabled. Otherwise, set it to disabled. For information on enabled QoS, see

“Turning on QoS monitor” on page 68.

7 Click Save.

Configuring PSTN fallback

By enabling PSTN fallback, you allow the system to check the availability of suitable bandwidth for a VoIP call. If the bandwidth is adequate, it connects the call across the VoIP line. If the bandwidth is not adequate, it routes the call across the PSTN.

In a network configured for PSTN fallback, there are two connections between a Business Communications Manager and a remote system. One connection is a VoIP trunk, the other is a T1, BRI, PRI or analog line. When a user dials the destination code, the system checks first to see if the connection between the two systems can support an appropriate level of QoS. If it can, the call proceeds as normal. If the minimum acceptable level of QoS is not met, the call is routed over a different route, commonly a PSTN.

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Figure 25 PSTN fallback diagram

Chapter 4 VoIP Trunk Configuration 61

IP network

BCM A

PSTN

BCM B

Before configuring fallback, you must have both your VoIP trunk and your fallback line properly configured and line pools created for each. For information on creating a VoIP line pool, see

“Putting VoIP lines into a line pool” on page 57. You create a PSTN line pool in the same manner.

Ensure that you use a different line pool for PSTN lines than for VoIP lines.

Setting up PSTN fallback includes:

• Setting up a VoIP schedule

• Configuring routes

• Creating a destination code

• Configuring digits

• Activating the VoIP schedule

• Turning on QoS monitor

Setting up a VoIP schedule

Setting up a VoIP schedule allows you to create two different call routes for a single destination code. One of these is a VoIP route. The other is typically a PSTN route. The PSTN route uses the ‘Normal’ schedule which runs when no other schedule is activated.

To set up the VoIP route:

1 In the Unified Manager, open Services, Telephony Services, Scheduled Services, Common

Settings, Schedule Names:

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62 Chapter 4 VoIP Trunk Configuration

Figure 26 VoIP schedule

2 Select a schedule that will not be used for another purpose, typically Schedule 4.

3 Change the schedule name to Vo I P.

4 Open Services, Telephony Services, Scheduled Services, Routing Service, and click on

Vo I P.

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Figure 27 VoIP Routing Service

Chapter 4 VoIP Trunk Configuration 63

5 Change the Service setting to Manual.

6 Change the Overflow setting to Y.

Configuring routes

Configuring routes allows you to set up access to the VoIP and the PSTN line pools. These routes can be assigned to destination codes using schedules. You must configure your PSTN and VoIP line pools before continuing.

Note: If you already have routes for your PSTN or VoIP line pools configured, you do not need to configure new routes, unless you cannot match the dialed digits. See “Configuring digits” on page 67.

To configure the PSTN route:

1 Open Services, Telephony Services, Call Routing and click on Routes.

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64 Chapter 4 VoIP Trunk Configuration

Figure 28 Route list

2 Click the Add button. The Add Routes dialog appears.

Figure 29 Add route dialog

3 Type a number between 001 and 999 to define this route. Only numbers not otherwise

assigned will be allowed by the system.

4 Click the route you just created.

5 In the Use Pool box, type the letter of the line pool for the fallback lines.

To configure the VoIP route:

1 Open Services, Telephony Services, Call Routing and click on Routes.

2 Click the Add button. The Add Routes dialog appears.

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Chapter 4 VoIP Trunk Configuration 65

Figure 30 Add route dialog

3 Type a number between 001 and 999 to define this route.Only numbers not otherwise assigned

will be allowed by the system.

4 Click the route you just created.

5 In the Use Pool field, type the letter of the line pool for the VoIP lines.

Creating a destination code

Creating a destination code allows you to combine the two routes created in “Configuring routes”

on page 63 under a single dialed code. When this code is dialed, the Business Communications

Manager will select the VoIP line if possible, and if not, will fall back to the PSTN line.

1 Open Services, Telephony Services, Call Routing and highlight Destination Codes.

2 Click Add. The Destination Code dialog appears.

Figure 31 Add destination code dialog

3 Type a digit for this destination code and click Save.

4 Click on the destination code you just created, click on the key beside Schedules, and click

Normal.

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66 Chapter 4 VoIP Trunk Configuration

Figure 32 Normal schedule

5 Change Use Route to the route you configured for your PSTN fallback line.

6 Under the Schedules list, highlight VoI P.

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Figure 33 VoIP schedule

Chapter 4 VoIP Trunk Configuration 67

7 Change Use Route to the route you configured for your VoIP line.

Configuring digits

For PSTN fallback to work, you must ensure that the digits the user dials will be the same regardless of whether the call is going over the VoIP trunk or the PSTN. In many cases, this involves configuring the system to add (inserting) or remove (absorbing) digits. For information of inserting and absorbing digits, see the Business Communications Manager 2.5 Programming Operations Guide.

For example: a caller dials 7 to access the VoIP line, 55 to select the remote gateway, and then 224 to dial a DN on the remote Business Communications Manager. The dialing plan must be set up so that calls are treated the same way by the VoIP and PSTN routes. In the example above, dialing 755224 must access the same DN even if the call is routed across the PSTN and not through VoIP.

Activating the VoIP schedule

Before activating the VoIP schedule, calls using the destination code are routed over the PSTN. This is because the system is set to use the ‘Normal’ schedule, which routes the call over the PSTN.

Once the VoIP schedule is activated, calls are routed over the VoIP trunk.

The VoIP line must be activated from the control set for the schedule. For information about control sets, see the Business Communications Manager 2.5 Programming Operations Guide.

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68 Chapter 4 VoIP Trunk Configuration

To activate the VoIP lines:

1 Dial Feature 873 from the control set for the VoIP trunk. The phone prompts you for a

password.

2 Type the password. Press OK. The first schedule appears.

3 Scroll down the list until VoIP is selected.

Press OK.

The VoIP schedule stays active even after a system reboot, and can only be deactivated manually.

To deactivate the VoIP line:

1 Dial Feature # 873. The phone prompts you for a password.

2 Type the password. Press OK. The system restores Normal schedule.

Turning on QoS monitor

For fallback to function, the QoS monitor must be enabled:

1 In Unified Manager, open Services, IP Telephony, H.323 Trunks, and click on Remote

Gateways:

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Figure 34 Remote Gateway list

Chapter 4 VoIP Trunk Configuration 69

2 Select the Remote Gateway for which you want to enable QoS Monitoring.

3 From the Configuration menu, click Modify Entry.

Figure 35 Remote Gateway dialog

4 For the QoS Monitor field, select Enabled.

5 Set the Transmit Threshold and Receive Threshold to a value between 0 and 5. This marks

the level of quality that the gateway must be able to support before transmitting a call. In most cases, the transmit threshold and receive threshold should be the same. On a line where

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70 Chapter 4 VoIP Trunk Configuration

communications in one direction are more important than in the other direction, you can set up asymmetrical thresholds.

For information about using the QoS monitor, see “Quality of Service Monitor” on page 77.

Enabling PSTN fallback

To enable PSTN fallback:

1 Open Services, IP Telephony and click on H.323 trunks.

2 Click the Fallback to Circuit-Switched menu and select Enabled-All or

Enabled-TDM-only. Enabled-TDM-only enables fallback for calls originating on TDM telephones. This is useful if your IP telephones are connected remotely, on the public side of the Business Communications Manager network, because PSTN fallback is unlikely to result in better quality of service in that scenario.

Incoming call configuration

This section explains how to receive incoming calls placed over a VoIP network. For information about setting up your Business Communications Manager to place outgoing VoIP calls, see

“Outgoing call configuration” on page 56.

Assign a target line to the DN

A target line routes incoming calls to specific telephones (DNs) depending on the incoming digits. This process is independent of the trunk over which the call comes in. The mapping of the target lines involves two steps:

• The incoming digits (e.g. 321) are mapped to a target line (e.g. 241) by setting the Received Number to the incoming digits (e.g. to line 241 to 321).

• The target line is mapped to a telephone by assigning the line (241) to the telephone.

For a detailed explanation of target lines, see the Business Communications Manager 2.5 Programming Operations Guide.

To assign a target line to a DN:

1 In Unified Manager, open Services/Telephony Services/System DNs/Active Set DNs.

2 Choose the DN of the telephone where you want the line to be directed.

3 Choose Line Access/Line assignment and click the Add button.

4 Enter the number of an available target line (241-412). Click the Save button.

5 Click on the line number you just created. The DN/Line screen appears in the right frame.

6 Ensure that Appearance type is set to Appr & Ring.

7 Go to Services/Telephony Services/Lines/Ta rg et L ine /<Target line number from step 4>.

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8 Click the key beside Trunk/line data.

9 Click on Received number.

10 In the Public number field, enter the DN.

The telephone assigned to that DN can now receive calls routed across VoIP trunks.

Example configuration

This section walks through a sample Business Communications Manager configuration. In this scenario, two Business Communications Managers in different cities are connected to a WAN. One Business Communications Manager resides in Ottawa; the other in Santa Clara. The systems already communicate through a PRI line, which will be configured to be used for fallback. Both systems already have all keycodes installed for eight VoIP lines, and resources properly allocated for VoIP trunking. For information about keycodes, see the Business Communications Manager

2.5 Keycode Installation Guide. For information on Resource Allocation, see Configuring the MSC

Resources in the Business Communications Manager 2.5 Programming Operations Guide.

Chapter 4 VoIP Trunk Configuration 71

Figure 36 Example PSTN fallback

IP network

BCM Santa Clara

PSTN

Business Communications Manager Ottawa

• Private IP address: 10.10.4.1

• Public IP address: 47.62.54.1

• Phones 221-231

• From this system, dial 9 to get onto PSTN

BCM Ottawa

Business Communications Manager Santa Clara

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72 Chapter 4 VoIP Trunk Configuration

• Private IP address: 10.10.5.1

• Public IP address: 47.62.84.1

• Phones 321-331

• From this system, dial 9 to get onto PSTN

On Business Communications Manager Ottawa

This procedure details actions that the installer performs to set up the Business Communications Manager Ottawa.

1 The installer obtains keycodes for eight VoIP lines, and inputs the keycodes into the system.

Each Business Communications Manager has 10 phones, VoIP lines are available for eight phones, but if all 10 phones try to place or receive VoIP calls at the same time, two of the calls will be rerouted to the PSTN. The installer sets up 221 as the Control set for each line, so that the VoIP schedule can be manually activated. This setup is necessary for PSTN fallback.

2 The installer sets the published IP address.

Because the public IP network is connected to the LAN 2 connection, the installer sets the published IP address to LAN 2. This is the address that devices on the Packet Data Network (PDN) will use to locate the system.

3 The installer configures the media for the system, using the following settings:

• The first preferred codec is set to G.729. The installer chooses this setting due to the

unique requirements of this installation.

• Silence Compression is turned on.

• Jitter Buffer is set to medium.

4 The installer puts eight VoIP lines into line pool O. Any line pool can be used as long as all of

the lines in the pool are VoIP. The installer does not set an access code for the line pool, because the access code does not work with fallback. Instead, the line pool will be accessed using destination digits after the installer sets up PSTN fallback.

5 For each telephone on the system (DNs 221 to 231), the installer gives the DN access to line

pool O.

6 The installer sets up a remote gateway for the Santa Clara Business Communications Manager,

using the following settings:

• Destination IP: 47.62.84.1 This is the published IP address of the Santa Clara Business

Communications Manager.

• QoS Monitor: Enabled This must be enabled for PSTN fallback to function.

• Transmit Threshold: 3.0 This is a Mean Opinion Score (MOS) value that ensures that the

VoIP lines are used as long as the system can provide moderate quality.

• Receive Threshold: 3.0 This is a MOS value that ensures that the VoIP lines are used as

long as the system can provide moderate quality.

• Destination Digits: 44 This means that callers dial ‘44’ to call phones on the Santa Clara

Business Communications Manager after dialing the destination digit.

7 The installer sets up a schedule called VoIP with these settings:

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• Service: Manual

• Overflow: Y

Chapter 4 VoIP Trunk Configuration 73

8 The installer defines a new route called Route 003, and sets it to use line pool

9 The installer defines a new route called Route 100, and sets it to use line pool O. This is the

line pool that contains the VoIP lines.

10 The installer creates a destination code of 7.

• Under the Normal schedule, the installer assigns Route 003, which uses line pool PRI-A.

• Under the VoIP schedule the installer assigns Route 100, which uses the VoIP lines in line

pool O.

11 The installer configures the call digits. They must be configured so that calls will still be

processed correctly if routed over PSTN. For information about configuring dialing plans, see the Business Communications Manager 2.5 Programming Operations Guide.

12 From the control set 221, the installer dials Feature 873 and selects the VoIP schedule. VoIP is

now activated. At this point, the system is configured to make outgoing calls, but it is not set up to receive incoming calls.

13 The installer programs target line 241 with the received digits 221, then assigns this target line

to DN 221. This means that calls received containing the digits 221 are delivered to DN 221. The installer repeats this procedure for each DN, using different received digits and target lines.

The Ottawa Business Communications Manager is now set to handle calls sent to and from a remote VoIP gateway. However, the Santa Clara Business Communications Manager must be set up before any calls can be made.

PRI-A.

On Business Communications Manager Santa Clara

This procedure details actions that the installer performs to set up the Business Communications Manager Santa Clara.

1 The installer obtains keycodes for eight VoIP lines, and inputs the keycodes into the system.

The installer sets up 321 as the Control set for each line, so that the VoIP route can be manually activated.

2 The installer sets the published IP address. Because the public data network (PDN) is

connected to the LAN 2 connection, the installer sets the published IP address to LAN 2. This is the address that devices on the PDN will use to locate the system.

3 The installer configures the media for the system, using the following settings:

• The first preferred codec is set to G.729.

• Silence Compression is turned on.

• Jitter Buffer is set to medium.

4 The installer puts the first eight VoIP lines into line pool O. Any line pool can be used as long

as all of the lines in the pool are VoIP. The installer does not set an access code for the line

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74 Chapter 4 VoIP Trunk Configuration

pool, because the access code would not work with fallback. Instead, the line pool will be accessed using destination digits after the installer sets up PSTN fallback.

5 For each set on the system (DNs 321 to 331), the installer gives the set access to line pool O.

6 The installer sets up a remote gateway for the Santa Clara Business Communications Manager,

using the following settings:

• Destination IP: 47.62.54.1 This is the published IP address of the Ottawa Business

Communications Manager.

• QoS Monitor: Enabled This must be enabled for PSTN fallback to function.

• Transmit Threshold: 3.0 This is a MOS value that ensures that the VoIP lines are used as

long as the system can provide moderate quality.

• Receive Threshold: 3.0 This is a MOS value that ensures that the VoIP lines are used as

long as the system can provide moderate quality.

• Destination Digits: 55 This means that callers dial 55 to call phones on the Ottawa

Business Communications Manager after dialing the destination digit.

7 The installer sets up a schedule called VoIP with these settings:

• Service: Manual

• Overflow: Y

8 The installer defines a new route called Route 003, and sets it to use

9 The installer defines a new route called Route 100, and sets it to use line pool O. This is the

line pool that contains the VoIP lines.

10 The installer creates a destination code of 7.

• Under the Normal schedule, the installer assigns Route 003, which uses PRI-A.

• Under the VoIP schedule the installer assigns Route 100, which uses the VoIP lines in line

pool O.

11 The installer configures the call digits. They must be configured so that calls will still be

processed correctly if routed over PSTN. For information about configuring dialing plans, see the Business Communications Manager 2.5 Programming Operations Guide.

12 The installer dials Feature 873 and selects the VoIP schedule. VoIP is now activated. At this

point, the system is configured to make outgoing calls, but it is not set up to receive incoming calls.

13 The installer programs target line 241 with the received digits 321, then assigns this target line

to DN 321. This means that calls received containing the digits 321 are delivered to DN 321. The installer repeats this procedure for each telephone using different received digits and target lines.

PRI-A.

Making calls

From a set on Business Communications Manager Ottawa, a caller dialling a set on Business Communications Manager Santa Clara must dial the destination code, the destination digits for the Business Communications Manager Santa Clara remote gateway and the DN of the set. For example, dialing 744322 would connect as follows:

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Chapter 4 VoIP Trunk Configuration 75

• 7 is the destination code. If a suitable level of QoS is available, the call is routed through the VoIP trunks.

• 44 is the remote gateway. The call is sent across the PDN using the IP address of the Santa Clara Business Communications Manager.

• 322 is linked to the target line associated with DN 322. The call arrives at the phone with the DN 322.

If a user in Santa Clara wanted to make a local call in Ottawa, they would dial 7559 followed by the local Ottawa number. The digit 7 accesses the VoIP line pool and 55 accesses the Ottawa Business Communications Manager, while 9 accesses an Ottawa outside line.

Connecting an i200X telephone

This section takes the example above and uses it to demonstrate how an installer would configure an i2004 phone on the system. For information on configuring i200X telephones, see

“Installing IP Telephones,” on page 31.

Note: IP clients require an IP network to reach the Business Communications Manager. However, they do not need to use VoIP trunks to communicate beyond the Business Communications Manager. They can use any type of trunk, just as any other phone on the Business Communications Manager can.

Chapter 3,

Connecting an i200X telephone on the LAN

In this case, the Santa Clara administrator wants to connect an i2004 phone using the LAN 1 network interface.

1 The installer sets up the Business Communications Manager to handle IP telephone by turning

Registration to ON, and Auto Assign DNs to ON.

2 The installer connects the telephone to the LAN, and sets it up using the following settings:

• Set IP address:10.10.5.10

• Default GW: 10.10.5.1 This is the IP address of the default gateway on the network, which

is the nearest router to the telephone.

• S1 IP address: 47.62.84.1 This is the published IP address of the Business

Communications Manager.

3 The Business Communications Manager automatically assigns the telephone the DN of 348.

4 The installer sets up a target line for DN 348, using the Received Digits 348.

This phone would follow all of the same dialing rules as the other telephones on the Santa Clara Business Communications Manager. A caller could dial 321 to connect with telephone 321, dial 9 to access the PSTN, or dial 755241 to access a telephone on the Ottawa system.

Configuring NetMeeting clients

NetMeeting is an application available from Microsoft which uses the H.323 protocol.

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To use NetMeeting:

1 Install NetMeeting on the client computer.

2 In the Tools menu, click Options. The options dialog appears.

Figure 37 NetMeeting options

3 Click Advanced Settings. The advanced settings dialog appears.

Figure 38 NetMeeting advanced options

4 Under Gateway settings, select the Use a gateway... option. In the Gateway field, type the

published IP address of the Business Communications Manager.

5 Add a remote gateway to your system as explained in “Configuring a remote gateway” on page

59. When prompted for the IP address of the remote gateway, type the IP address of the client

computer.

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Repeat this procedure for every NetMeeting client you want to set up.

Quality of Service Monitor

The Quality of Service Monitor is an application that monitors the quality of the IP channels. It does this by performing a check every 15 seconds. The QoS Monitor determines the quality of the intranet based on threshold tables for each codec. If the QoS Monitor is enabled, and it determines that the quality of service falls below the set threshold, it will trigger fallback to PSTN. For information about setting up the system to use QoS and fallback to PSTN, see “Configuring PSTN

fallback” on page 60.

Quality of Service Status

The QoS Status displays the current network quality described as a Mean Opinion Score (MOS) for each IP destination. A pull-down menu allows the administrator to view the MOS mapping.

Table 9 shows a sample QoS Monitor.

Chapter 4 VoIP Trunk Configuration 77

Tabl e 9 QoS status

47.192.5.2 Enabled 4.50 4.50 4.00 4.30 4.75 4.70 4.80 4.90

47.192.5.6 Disabled N/A N/A N/A N/A N/A N/A N/A N/A

Note: For the QoS monitor and PSTN fallback to function, both Business Communications Managers must list each other as a Remote Gateway and QoS Monitor must be enabled on both systems.

Updating the QoS Monitor data

To update the table with the most current values:

From the View menu, select Refresh.

Port settings

QoS

Monitor

G.723.1 6.3

G.729 G.711

Tx Rx Tx Rx Tx Rx Tx Rx

kbit/s

G.723.1 5.3

kbit/s

In some installations, you may need to adjust the port settings for the Business Communications Manager can work with other devices.

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78 Chapter 4 VoIP Trunk Configuration

Using firewalls

Firewalls can interfere with communications between the Business Communications Manager and another device. The port settings must be properly configured for VoIP communications to function properly. Using the instructions provided with your firewall, ensure that communications using the ports specified for VoIP are allowed.

An i2004 telephone uses ports between 51000 and 51200 to communicate with the Business Communications Manager.

The Business Communications Manager by default uses ports 28000 to 28255 to transmit VoIP packets. To modify these settings:

1 In Unified Manager, open Services, IP Telephony, Port Ranges.

Figure 39 Port Ranges

2 Select the Port Range you want to modify.

3 From the Configure menu, select Modify PortRanges. The Modify PortRanges dialog

appears.

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Chapter 4 VoIP Trunk Configuration 79

Figure 40 Port ranges dialog

4 Change the port settings.

5 Click the Save button.

Port settings for legacy networks

Business Communications Manager 2.5 uses UDP port ranges to provide high priority to VoIP packets in existing legacy IP networks. You must reserve these same port ranges and set them to high priority on all routers that an administrator expects to have QoS support. You do not need to reserve port ranges on DiffServ networks.

You can select any port ranges that are not used by well-known protocols or applications.

Each H.323 or VoIP Realtime Transfer Protocol (RTP) flow uses two ports for each direction. The total number of UDP port numbers to be reserved depends on how many concurrent RTP flows are expected to cross a router interface. In general:

• Backbone routers reserve more ports than edge routers.

• The port ranges on edge routers are a subset of the backbone router port ranges.

• Include port number UDP 5000 in the reserved port ranges, for the QoS monitor.

• The port ranges reserved in a Business Communications Manager 2.5 system are also reserved

by the remote router.

• You must reserve two ports for each voice call you expect to carry over the WAN link.

• You can reserve multiple discontinuous ranges. Business Communications Manager 2.5

requires that each range meet the following conditions:

— Each range must start with an even number.

— Each range must end with an odd number.

— You cannot have a total of more than 256 ports reserved.

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Using a gatekeeper

The Business Communications Manager supports the use of an ITU-H323 gatekeeper. A gatekeeper is a third-party software application residing somewhere on the network, which provides services such as:

• address translation

• admission control (ARQ)

• bandwidth control

• zone management

H.323 endpoints such as the Business Communications Manager are configured with one or more alias names that are registered with the gatekeeper. The gatekeeper stores the alias-IP mapping internally and uses them to provide alias to IP address translation services. Later if an endpoint's IP address changes, that endpoint has to re-register it with the gatekeeper.

See the gatekeeper software documentation for information about changing IP addresses.

The call signalling method

The call signalling method defines how the Business Communications Manager prefers call signalling information to be directed. Call signaling establishes and disconnects a call. The Business Communications Manager can use three types of call signalling:

• Direct: Under the direct call signalling method, call signalling information is passed directly between endpoints. The remote gateway table in the Unified Manager contains a mapping of phone numbers which the Business Communications Manager uses to perform DN to IP address resolution.

• Gatekeeper Resolved: Gatekeeper Resolved signalling uses a gatekeeper for call permission and address resolution. All call signalling occurs directly between H.323 endpoints. In effect, the gateway requests that the gatekeeper resolves the phone numbers into IP addresses, but the gatekeeper is not involved in call signalling.

• Gatekeeper Routed: Gatekeeper Routed signalling uses a gatekeeper for call permission and address resolution. In this method, call signalling is directed through the gatekeeper.

For information on changing the call signalling method, see “Alias names” on page 80.

Note: The Business Communications Manager can request a method for call signaling, but whether this request is granted depends on the configuration of the gatekeeper. Ultimately, the gatekeeper decides which call signalling method to use.

Alias names

One or more alias names may be configured for a Business Communications Manager. Alias names are comma delimited, and may be one of the following types:

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Chapter 4 VoIP Trunk Configuration 81

• E.164 — numeric identifier containing a digit in the range 0-9 (commonly used since it fits into dialing plans). Identified by the keyword "TEL:"

• H323Identifier

—

alphanumeric strings representing names, e-mail addresses, etc. Identified

by the keyword "NAME:"

• Transport Address — IP Address. Identified by the keyword "TA:"

In the following example the Business Communications Manager is assigned an E.164 and an H323 Identifier alias:

Alias Names: tel:76, name:bcm10.nortel.com

Modifying the call signalling method

To modify the call signalling method:

1 In the Unified Manager, open Services, IP Telephony, and click on H.323 trunks:

Figure 41 Local gateway IP interface

2 In the Call Signalling list, select the appropriate setting. For information about the settings,

see “The call signalling method” on page 80.

If selecting GateKeeperRouted or GateKeeperResolved, in the Gatekeeper IP box type the IP address of the machine that is running the gatekeeper.

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If selecting GateKeeperRouted or GateKeeperResolved, in the Alias Names box type one or more alias names for the gateway. For information on setting alias names, see

“Alias names” on page 80.

Gatekeeper call scenarios

This section explains what must be set up, and how a call would be processed for the two types of gatekeeper configurations. Figure 42 on page 82 shows a network with three Business Communications Managers and a gatekeeper.

Figure 42 Business Communications Manager systems with a gatekeeper

gatekeeper

IP:10.10.10.17

BCM Ottawa

IP:10:10:10:18

DN 321

BCM Santa Clara

IP:10:10:10:19

DN 421

BCM Calgary

IP:10:10:10:20

DN 521

IP network

This example explains how a call from DN 321 in Ottawa would be made to DN 421 in Santa Clara. It assumes that call signalling is set to Gatekeeper Resolved and no pre-granted ARQ has been issued:

1 Business Communications Manager Ottawa sends an AdmissionRequest (ARQ) to the

gatekeeper for DN 421.

2 The gatekeeper resolves DN 421 to 10.10.10.19 and returns this IP in an AdmissionConfirm to

the Business Communications Manager Ottawa.

3 Business Communications Manager Ottawa sends the call Setup message for DN 421 to the

gateway at 10.10.10.19, and the call is established.

If call signalling is set to Gatekeeper Routed and no pre-granted ARQ has been issued:

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Chapter 4 VoIP Trunk Configuration 83

1 Business Communications Manager Ottawa send an AdmissionRequest to the gatekeeper for

DN 421.

2 The gatekeeper resolves DN 421 to 10.10.10.17.

3 Business Communications Manager Ottawa sends the call Setup message for DN421 to the

gatekeeper (10.10.10.17), which forwards it to the gateway at 10.10.10.19, and the call is established.

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Chapter 5 Typical applications

This section explains several common installation scenarios and provides examples for how to use VoIP trunks and IP telephony to enhance your network.

M1 to Business Communications Manager over IP trunks

The Business Communications Manager 2.5 can connect with a Meridian 1 to extend an existing network. In this type of installation, capabilities of the M1 are available to phones on the Business Communications Manager 2.5. For example, a company that has a M1 in a Head Office may want to set up a warehouse in another region. The warehouse can then call the head office across VoIP lines, bypassing long-distance tolls.

Figure 43 M1 to Business Communications Manager network diagram

Head Office

Warehouse

Meridian Telephone

PSTN (fallback route)

company server

Intranet VoIP trunk

To set up this system:

1 Make sure the M1 ITG meets the following requirements:

• ITG Kit [NTZC44BA] Delta 24.24

• Rls25.30

• S/W Packages 57, 58, 59, 145, 147, 148, 160

BCM

Norstar telephone

i2004 telephone

2 Set up the M1 ESN programming (CDP/UDP). For information on this, see your M1

documentation.

3 On the Business Communications Manager 2.5 Unified Manager:

• Set up outgoing call configuration for the VoIP gateway.

• Set up a remote gateway for the Meridian 1.

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86 Chapter 5 Typical applications

• Set phones to receive incoming calls through target lines.

• Configure the PSTN fallback and enable QoS on both systems.

• Activate and program CDP or UDP. For information on CDP and UDP, see the Business

Communications Manager 2.5 Programming Operations Guide.

MCDN functionality

To use MCDN functionality over PRI, set up:

• MCDN PRI settings on the M1. For information on this, see your M1 documentation.

• SL-1 (MCDN) keycodes on the Business Communications Manager 2.5.

Multiple Business Communications Manager systems across VoIP trunks

This system allows multiple offices to connect across the company Intranet. This installation allows for CallPilot to direct calls throughout the system. Full toll bypass occurs through the tandem setup, meaning that any user can call any DN without long distance charges. Users have full access to system users, applications, PSTN connections, Unified Messaging. The network diagram shows two Business Communications Managers, but additional base units can be added.

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Chapter 5 Typical applications 87

Figure 44 Multiple Business Communications Manager systems network diagram

Head Office

Warehouse

Norstar Telephone

Company server

To set up this system:

Business Communications Manager

i2004 telephone

Remote Office

PSTN (fallback route)

Intranet VoIP trunk

remote i2004

Business Communications Manager

Norstar

Telephone

i2050 software phone

i2004 telephone

1 Ensure that the existing network can support the additional VoIP traffic.

2 Coordinate a dialing plan.

3 On each Business Communications Manager 2.5 system:

• Set up outgoing call configuration for the VoIP gateway.

• Set up a remote gateway for other Business Communications Managers or NetMeeting

users.

• Set phones to receive incoming calls through target lines.

• Configure the PSTN fallback and enable QoS on both systems.

4 Reboot each system.

This system uses fallback to PSTN so that if traffic between the Business Communications Manager systems becomes too heavy, calls are routed across the PSTN.

Multi-location chain with call center

In this installation, one Business Communications Manager runs a Call Center and passes calls to the appropriate branch offices, each of which have a Business Communications Manager. A typical use of this would be a 1-800 number that users world-wide can call, who are then directed the the remote office best able to handle their needs.

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88 Chapter 5 Typical applications

Figure 45 M1 to Business Communications Manager network diagram

Call Center

PSTN (fallback route)

Intranet VoIP trunk

Branch Offices

i2004 telephone

To set up this system:

1 Ensure that the existing network can support the additional VoIP traffic.

2 Coordinate a dialing plan.

3 On each Business Communications Manager 2.5 system:

• Set up outgoing call configuration for the VoIP gateway.

• Set up a remote gateway for other Business Communications Managers.

• Set phones to receive incoming calls through target lines.

• Configure the PSTN fallback and enable QoS on both systems.

4 Reboot each system.

5 Set up a Call Center on the central Business Communications Manager.

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Chapter 5 Typical applications 89

Business Communications Manager to IP Telephones

This system allows for home-based users or Call Center agents to use the full capabilities of the Business Communications Manager, including system users, applications and PSTN connections. This system does not require VoIP trunk configuration. This system functions in a similar mannger to the system described in “Multi-location chain with call center” on page 87. This system is less expensive and on a smaller scale. However, it does not offer PSTN fallback.

Figure 46 M1 to Business Communications Manager network diagram

Norstar Telephone

Central Office

Intranet VoIP trunk

i2050 Software Phone

i2004 telephone

Home-based users or Call Center agents

To set up this system:

1 Ensure that each remote user has a network connection capable of support VoIP traffic, such as

DSL or cable.

2 On the Business Communications Manager, set up the system to support IP telephones.

3 At the remote user’s location, install and configure i20XX telephone.

4 Register each phone and provide it with a DN.

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90 Chapter 5 Typical applications

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Appendix A Networking

This appendix provides information on making your network run more efficiently.

Determining the bandwidth requirements

The design process starts with the an IP telephony bandwidth forecast. The bandwidth forecast determines the following:

• LAN requirements: LAN must have enough capacity for the number of calls plus the overhead

• WAN requirements: WAN must have enough capacity for the number of calls plus the

overhead

Determining WAN link resources

For most installations, IP telephony traffic travels over WAN links within the intranet. WAN links are the highest recurring expenses in the network and they are often the source of capacity problems in the network. WAN links require time to receive financial approval, provision and upgrade, especially inter-LATA (Local Access and Transport Area) and international links. For these reasons, it is important to determine the state of WAN links in the intranet before installing the IP telephony.

Link utilization

This procedure explains how to determine and adjust link utilization:

1 Get a current topology map and link utilization report of the intranet. A visual inspection of the

topology can indicate the WAN links anticipated to deliver IP telephony traffic.

2 Record the current utilization of the links that will be handling IP telephony traffic. For

example, the link utilization can be an average of a week, a day, or one hour. To be consistent with the considerations, get the peak utilization of the trunk.

3 Determine the available spare capacity. Business Communications Manager intranets are

subject to capacity planning controls that ensure that capacity use remains below a determined utilization level. For example, a planning control can state that the utilization of a 56 kbit/s link during the peak hour must not exceed 50%. For example, for a T1 link, the threshold is higher, for example at 85%. The carrying capacity of the 56 kbit/s link can be 28 kbit/s, and for the T1,

1.3056 Mbit/s. In some organizations the thresholds can be lower than those used in this example. In the event of link failures, spare capacity for rerouting traffic is required.

Some WAN links can exist on top of layer 2 services such as Frame Relay and Asychronous Transfer Mode (ATM). The router-to-router link is a virtual circuit, which is subject not only to a physical capacity, but also to a logical capacity limit. The installer or administrator needs to obtain the physical link capacity and the QoS parameters. The important QoS parameters are CIR (committed information rate) for Frame Relay, and MCR (maximum cell rate) for Asynchronous Transfer Mode (ATM).

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92 Networking

The difference between the current capacity and its acceptable limit is the available capacity. For example, a T1 link used at 48% during the peak hour with a planning limit of 85% has an available capacity of approximately 568 kbit/s.

Network Engineering

Engineer the network for worst-case numbers to indicate the spare bandwidth a LAN must have to handle peak traffic. It is important to plan that the LAN/WAN can handle the IP telephony traffic using the defined codec, without delay or packet loss. The installer or administrator must select one configuration and then set up the LAN/WAN so there is more bandwidth than the IP telephony output.

Table 10 on page 93 provides bandwidth characteristics for the transmission of voice over IP for

various link types given codec type and payload sizes. The bandwidths provided in Table 10 on

page 93 explain the continuous transmission of a unidirectional media stream.

The peak bandwidth and average bandwidth requirements for a normal two-way call must take into account the affects of full and half duplex links and the affects of silence suppression. See Table 11

on page 94 and Table 12 on page 95 for voice Gateway bandwidth requirements.

Peak bandwidth is the amount of bandwidth that the link must provide for each call. Considering voice traffic only, the number of calls a link can support is:

Usable Link bandwidth

Number of calls





peak bandwidth per call

Number of Calls = Usable Link Bandwidth / peak Bandwidth per call

The average bandwidth takes into account the affects of silence suppression, which over time tends to reduce bandwidth requirements to 50% of the continuous transmission rate. The affects of silence suppression on peak bandwidth requirements differ depending on whether the link is half-duplex or full-duplex. See Appendix B, “Silence compression,” on page 103 for more information.

When engineering total bandwidth requirements for LANs and WANs, additional bandwidth must be allocated for data. Refer to standard Ethernet engineering tables for passive 10BaseT repeater hubs. Refer to the manufacturer’s specification for intelligent 10BaseT layer switches. WAN links must take into account parameters such as normal link utilization and committed information rates.

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Networking 93

Table 10 Voice over IP Transmission Characteristics for unidirectional continuous media stream

Codec Type Payload Size

ms Bytes Bytes kbit/s kbit/s kbit/s

G.711

10 80 120 116.8 97.6 103.2

IP Packet Ethernet B/W

PPP B/W FR B/W

(64 kb/s)

20 160 200 90.4 80.8 83.6

30 240 280 81.6 75.2 77.1

G.729 (8 kb/s)

G.723.1 (6.3 kb/s)

G.723.1

10 10 50 60.8 41.6 47.2

20 20 60 34.4 24.8 27.6

30 30 70 25.6 19.2 21.1

30 24 64 24.0 17.6 19.5

30 20 60 22.9 16.5 18.4

(5.3 kb/s)

Notes:

1) Gray Background indicated payload sizes used by Business Communications Manager 2.5 for transmission. Other values listed indicate payload sizes that the Business Communications Manager 2.5 can receive.

2) Ethernet bandwidth includes the 14 byte Ethernet frame overhead plus a 12 byte inter-frame gap.

Bandwidth Requirements on Half Duplex Links

Table 11 on page 94 provides bandwidth requirements for normal two-way voice calls on a

half-duplex link for a variety of link protocols, codec types and payload sizes.

With no silence suppression, both the transmit path and the receive path continuously transmit voice packets. Therefore, the peak bandwidth requirement per call on half-duplex links is:

Peak Bandwidth per call 2 Continuous Transmission Rate()=

(Half Duplex links, No Silence Suppression)

On half-duplex links with silence suppression enabled, the half-duplex nature of normal voice calls allows the sender and receiver to share the same bandwidth on the common channel; while the sender is talking, the receiver is quiet. Since only one party is transmitting at a time, silence suppression reduces the peak bandwidth requirement per call on a half-duplex link to::

Peak Bandwidth per call 1 Continuous Transmission Rate()=

peak Band Width per call = 1 * Continuous Transmission Rate

(Half Duplex links, With Silence Suppression)

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94 Networking

Tabl e 11 Bandwidth Requirements per BCM Gateway port for half-duplex links.

Codec Type

G.711

64 kb/s)

(

G.729

Payload Size Ethernet B/W

No SP

peak peak Avg peak peak Avg peak peak Avg

ms kbit/s kbit/s kbit/s kbit/s kbit/s kbit/s kbit/s kbit/s kbit/s

10 233.6 233.63233.63195.2 195.23195.23206.4 206.43206.4

20 180.8 180.83180.83161.6 161.63161.63167.2 167.23167.2

30 163.2 163.23163.23150.4 150.43150.43154.2 154.23154.2

10 121.6 60.8 60.8 83.2 41.6 41.6 94.4 47.2 47.2

Silence Suppression No SP

PPP B/W FR B/W

Silence Suppression No SP

Silence Suppression

(8 kb/s)

20 68.8 34.4 34.4 49.6 24.8 24.8 55.2 27.6 27.6

30 51.2 25.6 25.6 38.4 19.2 19.2 42.2 21.1 21.1

G.723.1

30 48.0 24.0 24.0 35.2 17.6 17.6 39.0 19.5 19.5

(6.3 kb/s)

G.723.1

30 45.8 22.9 22.9 33.0 16.5 16.5 36.8 18.4 18.4

(5.3 kb/s)

Notes:

1) Gray background indicates payload sizes used by Business Communications Manager 2.5 for transmission. Other values listed indicate payload sizes that BCM can receive.

2) Ethernet bandwidth includes the 14 byte Ethernet frame overhead plus a 12 byte inter-frame gap.

3) G.711 does not support silence suppression.

Bandwidth Requirements on Full Duplex Links

Table 12 on page 95 provides bandwidth requirements for normal two-way voice calls on a

full-duplex link for a variety of link protocols, codec types and payload sizes. Bandwidths for full-duplex links are stated in terms of the individual transmit and receive channels. For instance a 64 kbits full duplex link (e.g. a DS0 on T1 link) has 64 kbits in the transmit direction and 64 kbits in the receive direction.

With no silence suppression, both the transmit path and the receive path continuously transmit voice packets. Enabling silence suppression on full-duplex links reduces the average bandwidth. However, since transmit and receive paths use separate channels, the peak bandwidth per call per channel does not change. Therefore, peak bandwidth requirements per channel (Rx or Tx) per call on a full-duplex link is::

Peak Bandwidth per channel per call 2 Continuous Transmission Rate()=

(Full Duplex links, With or Without Silence Suppression)

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The bandwidth made available by silence suppression on full-duplex links with continuous transmission rate – average bandwidth requirement, is available for lower priority data applications that can tolerate increased delay and jitter.

Table 12 Bandwidth Requirements per BCM Gateway port for Full-duplex links.

Networking 95

Codec Type

Payload Size Ethernet B/W

No SP

Silence Suppression

PPP B/W FR B/W

No SP

Silence

No SP

Suppression

Silence Suppression

peak peak Avg peak peak Avg peak peak Avg

msec kbit/s kbit/s kbit/s kbit/s kbit/s kbit/s kbit/s kbit/s kbit/s

G.711

64 kb/s)

(

G.729

10 116.8 116.8 116.8397.6 97.6 97.6

20 90.48 90.4 90.4

30 81.6 81.6 81.6

80.8 80.8 80.8

75.2 75.2 75.2

10 60.8 60.8 30.4 41.6 41.6 20.8 47.2 47.2 23.6

103.2 103.2 103.2

83.6 83.6 83.6

77.1 77.1 77.1

(8 kb/s)

20 34.2 34.4 17.2 24.8 24.8 12.4 27.6 27.6 13.8

30 25.6 25.6 12.8 19.2 19.2 9.6 21.1 21.1 10.6

G.723.1

30 24.0 24.0 12.0 17.6 17.6 8.8 19.5 19.5 9.8

(6.3 kb/s)

G.723.1

30 22.9 22.9 11. 5 16.5 16.5 8.3 18.4 18.4 9.2

(5.3 kb/s)

Notes:

1) Gray background indicates payload sizes used by Business Communications Manager 2.5 for transmission. Other values listed indicate payload sizes that Business Communications Manager can receive.

2) Ethernet bandwidth includes the 14 byte Ethernet frame overhead plus a 12 byte inter-frame gap.

3) G.711 does not support silence suppression. Therefore the average bandwidth is the same as the peak bandwidth.

4) Bandwidths stated per channel (Rx or Tx).

LAN Engineering Examples

Example 1: LAN engineering - voice calls

Consider a site with four Business Communications Manager IP telephony ports. Assume a preferred codec of G.729, which uses a voice payload of 20 ms. Silence compression is enabled. The Ethernet LAN is half-duplex. Ethernet LAN may also be full duplex.

Given the above, what is the peak traffic in kbit/s that IP telephony will put on the LAN?

From Table 11 on page 94, the peak transmission bandwidth for G.729 with silence suppression enabled on a half-duplex link is 34.4 kbit/s per call or 137.6 kbit/s for all four calls.

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96 Networking

WAN engineering

Wide Area Network (WAN) links are typically full-duplex links - both talk and listen traffic use separate channels. For example, a T1 link uses a number of 64 kbit/s (DS0) duplex channels allowing *64 kbit/s for transmit path and n*64 kbit/s for the receive path.

(WAN links may also be half-duplex.)

Example 1: WAN engineering - voice calls

Consider a site with four IP telephony ports and a full-duplex WAN link using PPP. The preferred codec is G.729 kbit/s, which uses a voice payload of 20 ms. Silence compression is enabled.

Given the above, what is the peak traffic in kbit/s that IP telephony will put on the WAN?

From Table 12 on page 95, the peak transmission rate for G.729 is 24.8 kbit/s per call or 99.2 kbit/ s in each direction for all 4 calls. In other words, in order to support four G.729 calls, the WAN link must have at least 99.2 kbit/s of usable bandwidth (in each direction).

From Table 12 on page 95, the average bandwidth for each call is 12.4 kbit/sec per channel or 49.4 kbit/s for all 4 calls for each channel. Low priority data applications can make use of bandwidth made available by silence suppression.

Additional feature configuration

This section contains additional information on configuring your network to run efficiently.

Setting Non-linear processing

Non-linear processing should normally be enabled.

To set non-linear processing:

1 In Unified Manager, open Services, IP Telephony, and click on H.323 settings. The H.323

parameters appear in the right window.

2 Click the Non-linear processing drop-down menu, and select either Enabled or Disabled.

Determining network loading caused by IP telephony traffic

At this point, the installer or administrator has enough information to load the IP telephony traffic on the intranet.

Consider the intranet has the topology as shown Figure 47 on page 97, and the installer or administrator wants to know in advance the amount of traffic on a specific link, R4-R5. Consider there are four IP telephony ports per site.

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Networking 97

Each site supports four VoIP ports. Assume the codex is G.729 Annex B, 20 ms payload. Assuming full-duplex links, peak bandwidths per call are from Table 12 on page 95 which is between 24.8 kbit/s and 27.6 kbit/s peak transmission or approximately 28 kbit/s.

Route R1-R2 needs to support 4 VoIP Calls. R4-R5 needs to support eight VoIP calls. The incremental peak bandwidth for VoIP traffic is therefore:

R1-R2 peak VoIP Load 4 28 kbit/s()112kbit/s==

R4-R5 peak VoIP Load 8 28 kbit/s()112kbit/s==

With BCM VoIP gateway bandwidth requirements and

Traceroute

measurements, the R4-R5 link is expected to support the Santa Clara/Richardson, Santa Clara/Tokyo and the Ottawa/Tokyo traffic flows. The other IP telephony traffic flows do not route over R4-R5. A peak of eight calls can be made over R4-R5 for the four IP telephony ports per site. R4-R5 needs to support the incremental bandwidth of 8 x 12 = 96 kbit/s.

To complete this exercise, the traffic flow from every site pair needs to be summed to calculate the load on each route and loaded to the link.

Figure 47 Calculating network load with IP telephony traffic

Ottawa

Tok y o

Santa Clara

Santa Clara/Richardson traffic Ottawa/Tokyo traffic

Business Communications Manager IP telephony

Router

Richardson

IP Telephony Configuration Guide

98 Networking

Enough link capacity

Table 13 on page 98 sorts the computations so that for each link, the available link capacity is

compared against the additional IP telephony load. For example, on link R4-R5, there is capacity (568 kbit/s) to allow for the additional 96 kbit/s of IP telephony traffic.

Table 13 Link capacity example

Link Utilization (%)

End Points

Incremental IP telephony

Available Capacity kbit/s Threshold Used Site pair

capacity

kbit/s

load

Traffic kbit/s

Enough capacity?

R1-R2 1536 85 75 154 Santa Clara/

Ottawa

Santa Clara/ Tok yo

R1-R3 1536

R2-R3 1536

R2-R4 1536

R4-R5 1536 85 48 568 Santa Clara/

Richardson

Ottawa/Tokyo

Santa Clara/ Tok yo

15.5 Yes

24 Yes

Some network management systems have network planning modules that determine network flows. These modules provide more detailed and accurate analysis as they can include correct node, link and routing information. They also help to determine network strength by conducting link and node failure analysis. By simulating failures, re-loading network and re-computed routes, the modules indicate where the network can be out of capacity during failures.

Not enough link capacity

If there is not enough link capacity, consider one or more of the following options:

• Use the G.723.1 codec. Compared to the default G.729 codec with 20 ms payload, the G.723.1 codecs use 29% to 33% less bandwidth.

• Upgrade the bandwidth for the links.

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Networking 99

Other intranet resource considerations

Bottlenecks caused by non-WAN resources do not occur often. For a more complete evaluation consider the impact of incremental IP telephony traffic on routers and LAN resources in the intranet. The IP telephony traffic moves across LAN segments that are saturated, or routers whose central processing unit (CPU) utilization is high.

Implementing the network

LAN engineering

To minimize the number of router hops between the systems, connect the gateways to the intranet. Ensure that there is enough bandwidth on the WAN links shorter routes. Place the gateway and the LAN router near the WAN backbone. This prevents division of the constant bit-rate IP telephony traffic from bursty LAN traffic, and makes easier the end-to-end Quality of Service engineering for packet delay, jitter and packet loss.

Getting the best results from your system

Further network analysis

This section describes how to examine the sources of delay and error in the intranet. This section discusses several methods for reducing one-way delay and packet loss. The key methods are:

• “Reduce link delay” on page 100

• “Reducing hop count” on page 100

• “Adjust the jitter buffer size” on page 101

Components of delay

End-to-end delay is the result of many delay components. The major components of delay are as follows:

• Propagation delay: Propagation delay is the result of the distance and the medium of links moved across. Within a country, the one-way propagation delay over terrestrial lines is under 18 ms. Within the U.S., the propagation delay from coast-to-coast is under 40 ms. To estimate the propagation delay of long-haul and trans-oceanic circuits, use the rule of thumb of 1 ms per 100 terrestrial miles. If a circuit goes through a satellite system, estimate each hop between earth stations adds 260 ms to the propagation delay.

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100 Networking

• Serialization delay: The serialization delay is the time it takes to transmit the voice packet one bit at a time over a WAN link. The serialization delay depends on the voice packet size and the link bandwidth, and is the result of the following formula:

• Queuing delay: The queuing delay is the time it takes for a packet to wait in the transmission queue of the link before it is serialized. On a link where packets are processed in a first come first served order, the average queuing time is in milliseconds and is the result of the following formula:

The average size of intranet packets carried over WAN links generally is between 250 and 500 bytes. Queueing delays can be important for links with bandwidth under 512 kbit/s, while with higher speed links they can allow higher utilization levels.

• Routing and hop count: Each site pair takes different routes over the intranet. The route taken determines the number and type of delay components that add to end-to-end delay. Sound routing in the network depends on correct network design.

serialization delay in ms 8

average IP packet size in bytes



queuing time in ms 8

------------ ------------- ------------ ----------- ------------- ------------ ----



IP packet size in bytes



------------- ------------ ------------- ----------- --------



link bandwidth in kbit/s

lp–()link bandwidth in kbit/s()

Reduce link delay

In this and the next few sections, the guidelines examine different ways of reducing one-way delay and packet loss in the network.

The time taken for a voice packet to queue on the transmission buffer of a link until it is received at the next hop router is the link delay. Methods to reduce link delays are:

• Upgrade link capacity to reduce the serialization delay of the packet. This also reduces the utilization of the link, reducing the queueing delay. Before upgrading a link, check both routers connected to the link for the upgrade and ensure correct router configuration guidelines.

• Change the link from satellite to terrestrial to reduce the link delay by approximately 100 to 300 ms.

• Put into operation a priority queueing rule.

• Identify the links with the highest use and the slowest traffic. Estimate the link delay of these

links using Traceroute. Contact your service provider for help with improving your QoS.

Reducing hop count

To reduce end-to-end delay reduce hop count, especially on hops that move across WAN links. Some of the ways to reduce hop count include:

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Avaya BCM 2.5 IP Telephony Configuration Guide

Specifications and Main Features

Frequently Asked Questions

User Manual