RealPresence Collaboration Server (RMX)

Administrator Guide
Version 8.6 | June 2015 | 3725-74301-000D
RealPresence® Collaboration
Server (RMX)
1800/2000/4000
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Contents
Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
About RealPresence Collaboration Server (RMX) Administrator Guide . . . . . . . . . . . . . . . . . . 1
Who Should Read This Guide? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
How This Guide is Organized . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
About the RealPresence Collaboration Server System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Network Services Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
IP Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
ISDN Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Card Configuration Modes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Viewing the Card Configuration Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
RealPresence Collaboration Server Supported Features . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Software Prerequisites and Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Conferencing Modes Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
AVC Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
Continuous Presence (CP) Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
Video Protocol Support in CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Video Switching (VSW) Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Video Switching (VSW) Conferencing Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
AVC Basic Conferencing Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
Supplemental Conferencing Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
SVC-based Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
SVC Conferencing Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
MCU Supported Resolutions for SVC Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 19
Mixed CP and SVC Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
MCU Resource Capacities for Mixed CP and SVC Conferences . . . . . . . . . . . . . . . . . . . 22
Using Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
Conferencing Parameters Defined in a Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
Conferencing Capabilities in the Various Conferencing Modes . . . . . . . . . . . . . . . . . . . . 24
Default Profile Settings in CP Conferencing Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
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Default Profile Settings in SVC Only Conferencing Mode . . . . . . . . . . . . . . . . . . . . . . . . . 27
Default Profile Settings in a Mixed CP and SVC Conferencing Mode . . . . . . . . . . . . . . . 28
Viewing the List of Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
Profiles Toolbar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
Modifying an Existing Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
Deleting a Conference Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
Defining New Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
Exporting and Importing Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Guidelines for Exporting and Importing Conference Profiles . . . . . . . . . . . . . . . . . . . . . . 34
Exporting Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Exporting All Conference Profiles from an MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Exporting Selected Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
Importing Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
Defining AVC-Based Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
Defining AVC CP Conferencing Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
Additional Information for Setting CP Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
Gathering Phase . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
Gathering Phase Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Gathering Phase Duration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
Enabling the Gathering Phase Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
Overlay Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
Guidelines for using the Overlay Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
Selecting the Overlay Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
Site Names Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
Shorten the Site Name Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
Site Names Display Position . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82
Sending Text Messages During a Conference Using Message Overlay . . . . . . . . . . . . . 84
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
Sending Messages to All Conference Participants using Message Overlay . . . . . . . 85
Sending Messages to Selected Participants Using Message Overlay . . . . . . . . . . . . 86
Selecting the Chinese Font for Text Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
Selecting the Chinese Font . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
Defining an AVC Video Switching Conference Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
H.264 High Profile Support in Video Switching Conferences . . . . . . . . . . . . . . . . . . . 91
Minimum Threshold Line Rate System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
Defining SVC and Mixed CP and SVC Conference Profiles . . . . . . . . . . . . . . . . 93
Defining SVC Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
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Defining Mixed CP and SVC Conferencing Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Video Protocols and Resolution Configuration for CP Conferencing . . . . . . . 107
Video Resolutions in AVC CP Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
Video Display with CIF, SD and HD Video Connections . . . . . . . . . . . . . . . . . . . . . . . . . 108
H.264 High Profile Support in CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
H.264 High Profile Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
HD1080p60 Resolution Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
CP Conferencing with H.263 4CIF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
H.263 4CIF Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
The CP Resolution Decision Matrix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
Video Resource Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
Default Minimum Threshold Line Rates and Resource Usage Summary . . . . . . . . . . . . . . . 112
Resolution Configuration for CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
Modifying the Resolution Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
Resolution Configuration - Basic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
Maximum CP Resolution Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
Resolution Configuration Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
Resolution Configuration - Detailed . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
Sharpness and Motion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
Resolution Configuration Sliders . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
Flag Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
Setting the Maximum CP Resolution for Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 120
Minimum Frame Rate Threshold for SD Resolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
Additional Video Resolutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
w448p Resolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
Packet Loss Compensation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
Enabling Support of the w448p Resolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
Collaboration Server System Flag Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
Additional Intermediate Video Resolutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
Sharing Content During Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
Content Control Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
Guidelines for Controlling Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
Supported Content Control Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
Content Sharing Using H.239 Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
Content Sharing Using BFCP Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
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Guidelines for Using SIP BFCP Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
Content Sharing Using People+Content Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126
Guidelines for Content Sharing Using People+Content Protocol . . . . . . . . . . . . . . . 126
Content Media Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
Content Transmission Methods . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
Content Video Switching . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
Highest Common . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
Fixed Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
Multiple Content Resolutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
Guidelines for Sharing Contents using Multiple Content Resolutions . . . . . . . . . . . 130
Content Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
Customized Content Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
MCU Usage Modes of Content Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
H.263 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
H.263 & H.264 Auto Selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
H.264 Cascade Optimized . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
H.264 HD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
H.264 Content Sharing Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Guidelines for Sharing Content Using H.264 HD . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
Content Sharing Related Issues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134
Sharing Content in Cascaded Environments . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134
Guidelines for Content Snatching in Cascaded Conferences . . . . . . . . . . . . . . . . . . 134
Sending Content to Legacy Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
Guidelines for Sending Content to Legacy Endpoints . . . . . . . . . . . . . . . . . . . . . . . 135
Content Display on Legacy Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
Sending Content to Legacy Endpoints in Telepresence Mode . . . . . . . . . . . . . . . . . 136
Exclusive Content Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
Guidelines for Sharing Content in Exclusive Content Mode . . . . . . . . . . . . . . . . . . . 137
Forcing Other Content Capabilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
Managing Noisy Content Connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
Useful Procedures in Content Sharing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
Defining Content Sharing Parameters for a Conference . . . . . . . . . . . . . . . . . . . . . . . . . 139
H.264 Cascade Optimized Content Sharing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
Selecting a Customized Content Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
Sharing Content in Multiple Content Resolutions Mode . . . . . . . . . . . . . . . . . . . . . . . . . 143
Giving and Canceling Token Ownership (AVC Participants) . . . . . . . . . . . . . . . . . . . . . 144
Stopping a Content Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
Content Sharing Reference Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
Resolutions and Content Rate Reference Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
H.263 Content Rate Table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
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H.264 Resolution per Content Rate Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
H.264 Highest Common Content Rates Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . 148
H.264 Cascade Optimized (Fixed) Content Rates Tables . . . . . . . . . . . . . . . . . . . . 150
Implementing Media Encryption for Secured Conferencing . . . . . . . . . . . . . . 152
Media Encryption Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
Mixing Encrypted and Non-encrypted Endpoints in one Conference . . . . . . . . . . . . . . . . . . 153
Direct Connection to the Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 154
Connection to the Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
Moving from the Entry Queue to Conferences or Between Conferences . . . . . . . . . . . . 156
Recording Link Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
Enabling Media Encryption for a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Setting the Encryption Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Enabling Encryption in the Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Enabling Encryption at the Participant Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
Monitoring the Encryption Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
Setting Conferences for Telepresence Mode (AVC CP) . . . . . . . . . . . . . . . . . . 161
Collaboration Server Telepresence Mode Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
System Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
Conference Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
Automatic Detection of Immersive Telepresence (ITP) Sites . . . . . . . . . . . . . . . . . . . . . 162
Horizontal Striping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Cropping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Gathering Phase with ITP Room Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
Aspect ratio for standard endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
Skins and Frames . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
RPX and OTX Video Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
Room Switch Telepresence Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
Telepresence Display Decision Matrix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
Guidelines for Managing the Room Switch Telepresence Layouts by the MCU . . . . . . . 168
Speaker Priority in CP Video Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Reserved Screens . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Grid Screens . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Video Layout Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Selecting Speaker Priority . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
Sending Content to Legacy Endpoints in Telepresence Conferences . . . . . . . . . . . . . . . . . 172
Guidelines for Sending Content to Legacy Endpoints in Telepresence Conferences . . . 173
Content Display on Legacy Endpoints in Telepresence Conferences . . . . . . . . . . . 173
Enabling Telepresence Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 174
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Monitoring Telepresence Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 176
Monitoring Ongoing Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 176
Monitoring Participant Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
Creating Multiple Cascade Links Between Telepresence Conferences . . . . . . . . . . . . . . . . 178
Guidelines for Creating Multiple Cascading Links between Conferences . . . . . . . . . . . . 178
Enabling and Using Multiple Cascade Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 179
Creating a Link Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181
Link Participant in the Dial Out RMX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181
Participant Link in the Dial In RMX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183
Monitoring Multiple Cascade Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
Disconnection Causes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
Additional Conferencing Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 186
Video Preview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 186
Video Preview Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 186
Workstation Requirements to Display Video Preview . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
Testing your Workstation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
Previewing the Participant Video . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 188
Auto Scan and Customized Polling in Video Layout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
Guidelines for Using Auto Scan and Customized Polling . . . . . . . . . . . . . . . . . . . . . . . . 190
Enabling the Auto Scan and Customized Polling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
Enabling the Auto Scan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
Customized Polling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
Packet Loss Compensation - LPR and DBA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Packet Loss . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Causes of Packet Loss . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Effects of Packet Loss on Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Lost Packet Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Lost Packet Recovery Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Enabling Lost Packet Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 194
Monitoring Lost Packet Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
Layout Indications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 197
Network Quality Indication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 197
Network Quality Levels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 197
Indication Threshold Values . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
Displaying the Network Quality icons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
Customizing the Network Quality Icon Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
Recording Indications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
Audio and Video Participants Indication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
Setting Layout Indication via User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 201
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Layout Indications Icon Display Positions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 202
Layout Indications Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203
1x1 Layout Scenario Adaptation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 204
Lecture Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 204
Enabling Lecture Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 205
Selecting the Conference Lecturer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 205
Enabling the Automatic Switching . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 207
Lecture Mode Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 208
Restricting Content Broadcast to Lecturer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 210
Muting Participants Except the Lecturer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 211
Guidelines for Muting all the Participants Except the Lecturer . . . . . . . . . . . . . . . . . 211
Enabling the Mute Participants Except Lecturer Option . . . . . . . . . . . . . . . . . . . . . . 212
Audio Algorithm Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
Audio Algorithm Support Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
SIP Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 214
Mono . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 214
Stereo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 215
Audio algorithms supported for ISDN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 216
Monitoring Participant Audio Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 217
Automatic Muting of Noisy Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 219
Guidelines for Automatically Muting Noisy Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . 219
Automatic Muting of Noisy Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 220
Automatic Muting of Noisy Endpoints at the Conference Level . . . . . . . . . . . . . . . . 220
Automatic Muting of Noisy Endpoints at the MCU Level . . . . . . . . . . . . . . . . . . . . . 221
Permanent Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 222
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 222
Enabling a Permanent Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 223
Closed Captions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
Closed Captions Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
Enabling Closed Captions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
Defining Cascading Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
Cascading Link Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
Setting the Video Layout in Cascading conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
Guidelines for Setting the Video Layout in Cascading Conferences . . . . . . . . . . . . 227
Flags Controlling Cascading Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 228
DTMF Forwarding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 228
Play Tone Upon Cascading Link Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 228
Possible Cascading Topologies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
Basic Cascading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
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Basic Cascading Using IP Cascaded Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 230
Dialing Directly to a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 230
Dialing to an Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 230
Automatic Identification of the Cascading Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 231
Basic Cascading Using ISDN Cascaded Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 231
Network Topologies Enabling Content Sharing Over ISDN Cascaded Links . . . . . . 231
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 232
Gateway to Gateway Calls via ISDN Cascading Link . . . . . . . . . . . . . . . . . . . . . . . . 232
Gateway to MCU Calls via ISDN Cascading Link . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
MCU to MCU Calls via ISDN Cascading Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 234
RealPresence Collaboration Server Configuration Enabling ISDN Cascading Links 234
Suppression of DTMF Forwarding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 239
Star Cascading Topology . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 239
Master-Slave Cascading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 239
Creating a Cascade Enabled Dial-out/Dial-in Participant Link . . . . . . . . . . . . . . . . . 241
Cascading via Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
Enabling Cascading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
Creating the Cascade-enabled Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247
Creating the Dial-out Cascaded Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 248
Enabling Cascaded Conferences without Password . . . . . . . . . . . . . . . . . . . . . . . . 250
Monitoring Star Cascaded Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 250
Creating the Dial-out Link from a Conference Running on the MGC to the Conference
Running on the RealPresence Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . . 251
Cascading Conferences - H.239-enabled MIH Topology . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
MIH Cascading Levels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
Cascading Topologies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
MIH Cascading Guidelines in CP Licensing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 254
MGC to RealPresence Collaboration Server Cascading . . . . . . . . . . . . . . . . . . . . . 257
Meeting Rooms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 268
Use Time Out as DTMF Delimiter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
Meeting Rooms List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
Use Time Out as DTMF Delimiter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 271
Meeting Room Toolbar & Right-click Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 272
Creating a New Meeting Room . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 273
Entry Queues, Ad Hoc Conferences and SIP Factories . . . . . . . . . . . . . . . . . . 274
Entry Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 274
Defining a New Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 275
Listing Entry Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
Modifying the EQ Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
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Transit Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
Setting a Transit Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 280
IVR Provider Entry Queue (Shared Number Dialing) . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
Guidelines for setting the Entry Queue as IVR Provider . . . . . . . . . . . . . . . . . . . . . . 281
Configuring the Collaboration Server as IVR Provider . . . . . . . . . . . . . . . . . . . . . . . 281
Using External IVR Services via the MCCF-IVR Package . . . . . . . . . . . . . . . . . . . . . . . 282
Call Flows . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
Guidelines for Using External IVR Services via the MCCF-IVR Package . . . . . . . . 284
Configuring the MCU to Support External IVR Services via the MCCF-IVR . . . . . . 285
Configuring the Entry Queue to Use External IVR Services . . . . . . . . . . . . . . . . . . . 285
SIP Factories . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
Creating SIP Factories . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
SIP Registration & Presence for Entry Queues and SIP Factories with SIP Servers . . . . . . 288
Guidelines for registering Entry Queues and SIP Factories with SIP Servers . . . . . 288
Monitoring Registration Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
Ad Hoc Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
Gateway to Polycom® Distributed Media Application™ (DMA™) 7000 . . . . . . . . . . . . . 290
Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
Viewing the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 292
Displaying and Hiding the Group Members in the Navigation Pane . . . . . . . . . . . . . . . . 293
Participants List Pane Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 293
Displaying and Hiding the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 294
Adding Participants from the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 294
Adding Individual Participants from the Address Book to Conferences . . . . . . . . . . . . . 294
Adding a Group from the Address Book to Conferences . . . . . . . . . . . . . . . . . . . . . . . . 295
Participant Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 295
Managing Groups in the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 295
Managing the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 297
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 297
Adding a Participant to the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 297
Adding a New participant to the Address Book Directly . . . . . . . . . . . . . . . . . . . . . . . . . 297
Substituting E.164 Number in Dial String . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 303
Adding a Participant from an Ongoing Conference to the Address Book . . . . . . . . . . . . 303
Modifying Participants in the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 304
Deleting Participants from the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
Copying or Moving a Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 306
Searching the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
Filtering the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
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Filtering Address Book Data Using a Predefined Pattern . . . . . . . . . . . . . . . . . . . . . 308
Filtering Address Book Data Using a Custom Pattern . . . . . . . . . . . . . . . . . . . . . . . 309
Clearing the Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
Obtaining the Display Name from the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . 312
Guidelines for Obtaining the Display Name from the Address Book . . . . . . . . . . . . 312
Enabling and Disabling the Obtain Display Name from Address Book Feature . . . . 312
Importing and Exporting Address Books . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
Exporting an Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
Importing an Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
Upgrading and Downgrading Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 314
Integrating the Global Address Book (GAB) with the Collaboration Server . . . . . . . . . . . . . . 314
Integration with Resource Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 314
Scheduling Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 316
Using the Reservation Calendar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 318
Toolbar Buttons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 318
Reservations Views . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
Week View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
Day View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
Today View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
List View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
Changing the Calendar View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 321
Scheduling Conferences Using the Reservation Calendar . . . . . . . . . . . . . . . . . . . . . . . . . . 323
Creating a New Reservation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 323
Managing Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 329
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 329
Viewing and Modifying Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 329
Using the Week and Day views of the Reservations Calendar . . . . . . . . . . . . . . . . 329
Adjusting the Start Times of all Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 331
Deleting Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 333
Searching for Reservations using Quick Search . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 333
Operator Assistance & Participant Move . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
Operator Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
Operator Conference Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 336
Defining the Components Enabling Operator Assistance . . . . . . . . . . . . . . . . . . . . . . . . 336
To define a Conference IVR Service with Operator Assistance Options . . . . . . . . . 337
To define an Entry Queue IVR Service with Operator Assistance Options . . . . . . . 339
To define a Conference Profile for an Operator Conference . . . . . . . . . . . . . . . . . . 340
Starting an Ongoing Operator Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 342
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Saving an Operator Conference to a Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 345
Starting an Operator Conference from a Template . . . . . . . . . . . . . . . . . . . . . . . . . 346
Monitoring Operator Conferences and Participants Requiring Assistance . . . . . . . . . . . 347
Requesting Help . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 347
Participant Alerts List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 348
Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Using Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 350
Using Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 351
Toolbar Buttons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 351
Creating a New Conference Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 352
Creating a new Conference Template from Scratch . . . . . . . . . . . . . . . . . . . . . . . . . . . . 352
Saving an Ongoing or AVC-CP Operator Conference as a Template . . . . . . . . . . . . . . 358
Starting an Ongoing Conference From a Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . 359
Starting an Operator Conference from a Template (AVC Conferencing) . . . . . . . . . 360
Scheduling a Reservation From a Conference Template . . . . . . . . . . . . . . . . . . . . . . . . 361
Deleting a Conference Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 363
Exporting and Importing Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 363
Exporting Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 364
Exporting All Conference Templates from an MCU . . . . . . . . . . . . . . . . . . . . . . . . . 364
Exporting Selected Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
Importing Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 366
Start a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 369
Start an AVC CP Conference from the Conferences Pane . . . . . . . . . . . . . . . . . . . . . . . 370
General Tab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 371
Participants Tab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 375
Information Tab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 377
Media Sources Tab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 378
Starting a Mixed CP and SVC or SVC Only Conference from the Conferences Pane . . 379
Starting a New SVC Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 379
Scheduling an AVC-based Reservation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 380
Starting an Ongoing Conference From a Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . 381
Starting a Meeting from Microsoft Outlook Using Polycom Add-in . . . . . . . . . . . . . . . . . 382
Conference and Participant Level Operations . . . . . . . . . . . . . . . . . . . . . . . . . . 383
Conference Operations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 383
Copy Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 383
Paste Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 383
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Paste Conference As . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 384
Participant Operations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 385
Operational Buttons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 385
Copy, Cut and Paste Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 387
Copy Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 387
Cut Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 387
Paste Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 387
Paste Participant As . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 388
Moving Participants Between Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 389
Move Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 389
Moving Participants Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
Moving a Participant Interactively . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
Personal Conference Manager (PCM) (AVC CP Conferences) . . . . . . . . . . . . . . . . . . . . . . 391
PCM Usage Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 391
Modifying the PCM_FECC System Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 392
PCM Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 393
Initiating PCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 393
Remote Control Device Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 393
Menu Navigation - Arrow and Zoom Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 394
DTMF Codes - Numeric Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
PCM Main Menu - Level 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Click & View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Invite Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 396
Participants Mute/Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 397
Far End Camera Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 398
Video Forcing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 400
Recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 400
PCM Main Menu - Level 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 402
Drop Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 402
Terminating a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 403
Polycom Conferencing for Microsoft Outlook® . . . . . . . . . . . . . . . . . . . . . . . . . 404
Setting up the Calendaring Solution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 405
Calendaring Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 410
Creating and Connecting to a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 412
Creating a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 412
Connecting to a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 413
Collaboration Server Standalone Deployment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 414
Collaboration Server and Polycom RealPresence DMA System Deployment . . . . . 414
Polycom Solution Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 414
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Conference and Participant Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 415
General Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 415
Multi-Selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 416
Using the Chairperson Password for Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 416
Conference Level Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 417
Monitoring the Conference in the Conference List Pane . . . . . . . . . . . . . . . . . . . . . . . . 418
Viewing the Properties of CP and Mixed CP and SVC Conferences . . . . . . . . . . . . . . . 420
Viewing the Properties of Ongoing SVC Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . 432
Monitoring of Operator Conferences and Participants Requiring Assistance . . . . . . . . . 437
Requesting Help . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 438
Request to Speak . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 439
Participant Alerts List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 439
Participant Level Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 440
Participant Connection Monitoring (AVC and SVC-based Connections) . . . . . . . . . . . . 441
Viewing the Properties of Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 445
Monitoring IP Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 446
Monitoring SIP BFCP Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 457
Detecting Endpoint Disconnection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 458
Monitoring ISDN/PSTN Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 459
Monitoring Telepresence Participant Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 466
Recording and Streaming Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 467
Creating Multiple Virtual Recording Rooms on the Polycom® RealPresence® Media Suite
468
Configuring the Collaboration Server to Enable Recording . . . . . . . . . . . . . . . . . . . . . . . . . . 468
Defining the Dial Out Recording Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 469
Enabling the Recording Features in a Conference IVR Service . . . . . . . . . . . . . . . . . . . 470
Enabling the Recording in the Conference Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 471
Dial Out Recording Link Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 473
Dial Out Recording Link Encryption Flag Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . 473
Dial Out Recording Link Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 474
Managing the Recording Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 474
Recording Layout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 475
Using the Collaboration Server Web Client to Manage the Recording Process . . . . . . . 476
Using DTMF Codes to Manage the Recording Process . . . . . . . . . . . . . . . . . . . . . . . . . 478
Conference Recording with Codian IP VCR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 478
Users Connections and Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 479
Collaboration Server Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 479
User Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 479
Administrator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 479
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Administrator Read-only . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 480
Operator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 480
Chairperson . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 480
Auditor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 480
Machine Account . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 480
Listing Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 480
Adding a New User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 481
Deleting a User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 482
Changing a User’s Password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 483
Disabling a User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 483
Enabling a User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 483
Renaming a User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 484
Machine Account . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 484
Guidelines for defining a machine account . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 485
Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 485
Active Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 485
Connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 486
Viewing the Connections List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 486
Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 486
Using Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 486
IP Network Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 488
Collaboration Server IP Network Services Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 488
Management Network (Primary) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 489
Default IP Service (Conferencing Service) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 489
Using IPv6 Networking Addresses for Collaboration Server Internal and External Entities . .
490
IPv6 Addressing Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 490
Modifying the Management Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 491
Modifying the Default IP Network Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 499
Ethernet Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 515
IP Network Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 519
LAN Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
Media Redundancy - Collaboration Server (RMX) 2000/4000 . . . . . . . . . . . . . . . . . . . . 525
Media and Signaling Redundancy - Collaboration Server (RMX) 1800 . . . . . . . . . . . . . 525
Signaling and Management Redundancy - Collaboration Server (RMX) 4000 . . . . . . . . 527
Management Redundancy - Collaboration Server (RMX) 1800 . . . . . . . . . . . . . . . . . . . 527
Configuration Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 528
RealPresence Collaboration Server 1800 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 528
RealPresence Collaboration Server (RMX) 2000 . . . . . . . . . . . . . . . . . . . . . . . . . . . 529
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RealPresence Collaboration Server (RMX) 4000 . . . . . . . . . . . . . . . . . . . . . . . . . . . 529
On all systems: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 529
Hardware Monitor Indications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 529
Network Traffic Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 530
SIP Proxy Failover With Polycom® Distributed Media Application™ (DMA™) 7000 . . . . . . 530
RealPresence Collaboration Server (RMX) Network Port Usage . . . . . . . . . . . . . . . . . . . . . 531
Defining ISDN/PSTN Network Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 534
ISDN/PSTN Network Services Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 534
Adding/Modifying ISDN/PSTN Network Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 536
Obtaining ISDN/PSTN required information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 536
Modifying an ISDN/PSTN Network Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 542
Network Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Collaboration Server (RMX) 4000 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Collaboration Server 1800 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Collaboration Server (RMX) 2000 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Multiple Network Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 546
Resource Allocation and Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 548
First Time Installation and Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 548
Upgrading to Multiple Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 548
Gather Network Equipment and Address Information - IP Network Services Required Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 549
RealPresence Collaboration Server (RMX) Hardware Installation . . . . . . . . . . . . . . . . . 550
Collaboration Server (RMX) 4000 Multiple Services Configuration . . . . . . . . . . . . . 550
RealPresence Collaboration Server (RMX) 2000 Multiple Services Configuration . 551
Collaboration Server (RMX) 1800 Multiple Services Configuration . . . . . . . . . . . . . 553
Collaboration Server Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 554
System Flags and License Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 554
IP Network Service Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 554
Setting a Network Service as Default . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 562
Ethernet Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 562
Signaling Host IP Address and MCU Prefix in GK Indications . . . . . . . . . . . . . . . . . 563
Conference Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 563
Gateway Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 563
Hardware Monitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 565
Signaling Monitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 566
Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 566
Defining Dial Out Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 566
Reserving Video Resources for a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 567
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Monitoring Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 567
Resource Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 568
Port Usage Gauges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 568
NAT (Network Address Translation) Traversal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 569
Deployment Architectures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 570
Remote Connection Using the Internet . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 570
Business to Business Connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 571
FW (Firewall) NAT Keep Alive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 571
System Configuration in SBC environments . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 572
SIP TCP Keep-Alive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 572
IVR Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 573
IVR Services List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 573
IVR Services Toolbar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 574
Adding Languages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 575
Uploading a Message File to the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . 576
Defining a New Conference IVR Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 578
Defining a New Conference IVR Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 579
Change to Chairperson . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 596
Controlling the receipt of in-band and out-of-band DTMF Codes . . . . . . . . . . . . . . . 597
Entry Queue IVR Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 597
Defining a New Entry Queue IVR Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 597
Setting a Conference IVR Service or Entry Queue IVR Service as the Default Service . 603
Modifying the Conference or Entry Queue IVR Service Properties . . . . . . . . . . . . . . . . . . . . 605
Replacing the Music File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 606
Adding a Music File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 606
Creating Audio Prompts and Video Slides . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 607
Recording an Audio Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 607
Defining the format settings for audio messages . . . . . . . . . . . . . . . . . . . . . . . . . . . 607
Recording a new audio message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 609
Creating a Welcome Video Slide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 611
Inviting Participants using DTMF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 612
Invite Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 612
Entering Additional DTMF Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 612
Error Handling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 612
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 613
Enabling the Invite Participants using DTMF Option . . . . . . . . . . . . . . . . . . . . . . . . . . . . 613
Disabling the Invite Participant Option . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 616
External IVR Service Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 617
IVR Services Support with TIP Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 617
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Guidelines for TIP Support with IVR Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 617
Default IVR Prompts and Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 618
Volume Control of IVR Messages, Roll Call and Music . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 622
IVR Services in TIP-Enabled Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 623
IVR Services in TIP-Enabled Conferences Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 623
Entry Queue and Virtual Entry Queue Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 623
Configuring the Conference and Entry Queue IVR Services . . . . . . . . . . . . . . . . . . 623
Call Detail Record (CDR) Utility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 624
The CDR File Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 624
CDR File Formats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 624
Multi-Part CDR Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 625
Enabling the Multi-Part CDR Option . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 626
CDR File Contents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 626
Viewing, Retrieving and Archiving Conference Information . . . . . . . . . . . . . . . . . . . . . . . . . 627
Viewing the Conference Records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 627
Multi-part CDR File display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 628
Refreshing the CDR List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 629
Retrieving and Archiving Conference CDR Records . . . . . . . . . . . . . . . . . . . . . . . . . . . 629
Gateway Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 630
Gateway Functionality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 630
Call Flows . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 631
IP Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 631
Direct Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 631
Gateway IVR Dialing For IP Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 634
Direct Dialing Using IP Addresses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 635
Calling a SIP Endpoint in a Remote Domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 637
ISDN Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 638
Dialing via Gateway IVR for ISDN Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . 638
Direct Dial-in to Endpoints or DMA VMR using Automatically Generated Destination
Numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 639
Configuring the Gateway Components on the Collaboration Server . . . . . . . . . . . . . . . . . . . 641
Defining the IVR Service for Gateway Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 641
Defining the Conference Profile for Gateway Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 644
Defining the Gateway Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 645
System Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 646
Displaying the Connection Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 646
Enabling PSTN dial-in using GK prefix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 646
Gateway Calls Redialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 646
Gateway Redial Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 646
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Redial on Wrong Number . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 646
Disconnect on Busy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 648
Disconnect on No Answer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 648
Disconnect on Wrong Number . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 648
Monitoring Ongoing Gateway Sessions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 648
Connection Indications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 648
Gateway Session Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 649
Connected Participant Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 650
Direct Dialing from ISDN/PSTN Endpoint to IP Endpoint via a Meeting Room . . . . . . . . . . . 651
Dialing to Polycom® RealPresence DMA System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 653
Calling a RealPresence DMA Direct with Automatically Generated Destination
Dial Strings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 653
Calling the RealPresence DMA via Gateway IVR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 654
Manual Dial String Entry . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 654
Automatic Dial String Generation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 655
PSTN Dial-in Using GK Prefix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 656
Deploying a Polycom RMX™ Serial Gateway S4GW . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 657
RMX Manager Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 658
Installing the RMX Manager Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 658
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 659
Accessing or Downloading the RMX Manager Installer . . . . . . . . . . . . . . . . . . . . . . . . . 659
Accessing the RMX Manager Application Installer Directly from the MCU . . . . . . . . 659
Downloading the Installation files from Polycom Support Site . . . . . . . . . . . . . . . . . 660
Accessing the RMX Manager Installer from the Login screen . . . . . . . . . . . . . . . . . 661
Installing the RMX Manager on Your Workstation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 661
Installing the RMX Manager for Multi-User Capability . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 663
Starting the RMX Manager Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 665
Connecting to the MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 667
RMX Manager Main Screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 668
MCUs Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 668
Use Other RMX Manager UI Components . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 670
Adding MCUs to the MCUs List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 670
Grouping the Participants by MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 672
Start Monitoring/Stop Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 672
Modifying the MCU Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 673
Disconnecting an MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 673
Removing an MCU from the MCUs Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 674
Changing the RMX Manager Language . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 674
Import/Export RMX Manager Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 675
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Installing RMX Manager in Secure Communication Mode . . . . . . . . . . . . . . . . . . . . . . . . . . 676
Using an Internal Certificate Authority . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 681
Use the RMX Web Client . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 684
Starting the RMX Web Client . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 684
RMX Web Client Screen Components . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 685
Viewing and System Functionality Permissions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 686
Conferences List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 687
List Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 688
RMX Management Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 688
Status Bar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 688
System Alerts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 689
Participant Alerts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 689
Port Usage Gauges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 690
MCU State . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 691
Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 691
Displaying and Hiding the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 692
Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 692
Displaying and Hiding Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 693
Customizing the Main Screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 693
Increasing and Decreasing the Text Size . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 694
Customizing the RMX Management Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 694
Accessing the Management Console When Security Policy is Stringent . . . . . . . . . . . . . . . 696
Administration and Utilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 697
System and Participant Alerts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 697
System Alerts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 698
Participant Alerts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 700
RMX Time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 701
Altering the clock . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 701
Resource Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 703
Resource Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 703
Flexible Resource Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 704
Forcing Video Resource Allocation to CIF Resolution . . . . . . . . . . . . . . . . . . . . . . . 704
Resource Reports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 705
Displaying the Resource Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 706
Resource Reports for Collaboration Server 1800 and 2000/4000 with MPMRx . . . . 706
ISDN/PSTN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 708
MCU Resource Management by RealPresence Resource Manager, and Polycom
RealPresence DMA System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 708
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 709
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Port Usage Threshold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 709
Setting the Port Usage Threshold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 709
SIP Dial-in Busy Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 710
Port Usage Gauges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 711
System Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 712
SNMP (Simple Network Management Protocol) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 712
MIBs (Management Information Base) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 713
Traps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 713
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 713
MIB Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 713
Private MIBs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 713
Support for MIB-II Sections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 714
The Alarm-MIB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 714
H.341-MIB (H.341 – H.323) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 714
Standard MIBs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 714
Unified MIB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 715
Traps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 717
Status Trap . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 718
RMX MIB entities that do not generate traps. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 718
Defining the SNMP Parameters in the Collaboration Server . . . . . . . . . . . . . . . . . . . . . 720
Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 727
Using Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 727
Audible Alarm Permissions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 728
Stop Repeating Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 728
Configuring the Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 728
User Customization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 728
Replacing the Audible Alarm File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 729
Multilingual Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 730
Customizing the Multilingual Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 730
Banner Display and Customization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 731
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 732
Non-Modifiable Banner Text . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 732
Sample 1 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 732
Sample 2 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 733
Sample 3 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 733
Sample 4 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 733
Customizing Banners . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 734
Banner Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 735
Login Screen Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 735
Main Screen Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 736
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Software Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 737
Backup and Restore Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 737
Using Software Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 738
Ping the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 739
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 739
Using Ping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 739
Notification Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 740
Logger Diagnostic Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 741
Information Collector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 743
Standard Security Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 743
Ultra Secure Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 744
Network Intrusion Detection System (NIDS) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 744
Using the Information Collector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 745
Step 1: Creating the Information Collector Compressed File . . . . . . . . . . . . . . . . . . 745
Step 2: Saving the Compressed File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 746
Step 3: Viewing the Compressed File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 747
Auditor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 747
Auditor Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 747
Retrieving Auditor Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 748
Auditor File Viewer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 749
Audit Events . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 752
Alerts and Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 752
Transactions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 754
ActiveX Bypass . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 755
Installing ActiveX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 755
Resetting the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 756
System Configuration Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 758
Controlling System Flag Values . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 758
Modifying System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 758
Manually Adding a System Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 759
Deleting a System Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 759
The System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 760
The MCMS System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 760
The CS System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 804
Special System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 807
Auto Layout Configuration Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 807
Content Related Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 810
Automatic Password Generation Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 810
Flags Specific to Maximum Security Environments - Ultra Secure Mode . . . . . . . . . 812
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Ultra Secure Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 814
Enable Ultra Secure Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 816
ULTRA_SECURE_MODE System Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 817
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 817
Modifying the Ultra Secure Mode System Flag Value . . . . . . . . . . . . . . . . . . . . . . . 817
System Flags affected by Ultra Secure Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 818
Certificate Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 821
(PKI) Public Key Infrastructure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 821
Adding Certificates to the Certificate Repository . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 822
Trusted Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 823
Personal Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 826
Certificate Validation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 828
CRL (Certificate Revocation List) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 829
Adding a CRL . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 831
Removing a CRL . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 834
Certificate Revocation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 834
Revocation Method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 834
Self-signed Certificate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 835
Self-signed Certificate Creation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 835
Emergency CRL (Certificate Revocation List) Update . . . . . . . . . . . . . . . . . . . . . . . . . . 837
SIP TCP Keep-Alive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 841
Keep Alive Frequency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 843
User and Connection Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 843
Managing the System Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 844
Disabling/Enabling Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 844
Renaming Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 844
Disabling Inactive Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 844
Managing the User Login Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 844
Implementing Strong Passwords . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 844
Password Length . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 846
Implementing Password Re-Use / History Rules . . . . . . . . . . . . . . . . . . . . . . . . . . . 846
Defining Password Aging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 846
Maximum Repeating Characters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 847
Defining Password Change Frequency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 847
Forcing Password Change . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 847
Temporary User Lockout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 848
User Lockout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 848
User Login Record . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 848
Controlling User Sessions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 849
Management Sessions per System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 849
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Sessions per User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 849
Connection Timeout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 849
Session Timeout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 849
Erase Session History After Logout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 850
Banner Display and Customization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 850
Guidelines for Customizing the Login Page Banner . . . . . . . . . . . . . . . . . . . . . . . . . 850
Non-Modifiable Banner Text . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 851
Sample 1 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 851
Sample 2 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 851
Sample 3 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 851
Sample 4 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 852
Customizing Banners . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 852
Banner Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 854
Login Screen Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 854
Main Screen Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 855
Securing an External Database . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 855
MS Active Directory Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 856
Directory and Database Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 856
Ultra Secure Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 856
Standard Security Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 856
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 857
Enabling Active Directory Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 858
Restoring the Collaboration Server Using the USB Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . 859
MLPP (Multi Level Precedence and Preemption) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 860
Enabling Precedence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 861
SIP Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 861
Dial-in calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 861
Dial-out calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 864
Precedence Level Change . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 864
Configuring and Modifying Precedence Domains and DSCP Values . . . . . . . . . . . . . . . 865
System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 866
Monitoring Precedence Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 866
IEEE 802.1X Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 867
Certificate Repository . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 867
Enabling and Configuring 802.1X Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . 868
System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 869
Disabling 802.1X Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 870
Ethernet Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 870
White List Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 870
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 870
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Enabling, Disabling and Modifying the White List . . . . . . . . . . . . . . . . . . . . . . . . . . . 871
Alternative Network Address Types (ANAT) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 873
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 873
System Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 873
BFCP Over UDP – AS-SIP Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 874
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 874
Enabling AS-SIP Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 875
System Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 875
Internet Control Message Protocol (ICMP) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 876
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 876
System Flag: ENABLE_ACCEPTING_ICMP_REDIRECT . . . . . . . . . . . . . . . . . . . . 876
System Flag: ENABLE_SENDING_ICMP_DESTINATION_UNREACHABLE . . . . . 876
Password Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 877
Upgrade / Downgrade Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 877
Non-hashed Passwords . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 879
Self-signed Certificate Creation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 879
Media Encryption and Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 881
System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 881
Collaboration Server Hardware Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . 882
Viewing the Status of the Hardware Components . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 882
HW Monitor Pane Toolbar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 884
Viewing the Properties of Hardware Components . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 884
Temperature Thresholds . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 892
Diagnostic Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 893
Appendix A - Disconnection Causes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 896
IP Disconnection Causes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 896
ISDN Disconnection Causes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 901
Appendix B - Active Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 906
Appendix C - CDR Fields Unformatted File . . . . . . . . . . . . . . . . . . . . . . . . . . . . 918
The Conference Summary Record . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 918
Event Records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 920
Standard Event Record Fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 920
Event Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 920
Event Specific Fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 921
Disconnection Cause Values . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 952
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Appendix D - Ad Hoc Conferencing and External Database Authentication . 956
Ad Hoc Conferencing without Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 956
Ad Hoc Conferencing with Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 957
Entry Queue Level - Conference Initiation Validation with an External Database
Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 958
Conference Access with External Database Authentication . . . . . . . . . . . . . . . . . . . . . . . . . 959
Conference Access Validation - All Participants (Always) . . . . . . . . . . . . . . . . . . . . . . . 960
Conference Access Validation - Chairperson Only (Upon Request) . . . . . . . . . . . . . . . . 961
System Settings for Ad Hoc Conferencing and External Database Authentication . . . . . . . . 962
Ad Hoc Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 962
Authentication Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 963
MCU Configuration to Communicate with an External Database Application . . . . . 963
Enabling External Database Validation for Starting New Ongoing Conferences . . . 964
Enabling External Database Validation for Conferences Access . . . . . . . . . . . . . . . 965
Appendix E - Participant Properties Advanced Channel Information . . . . . . . 967
Appendix F - Secure Communication Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . 969
Certificate Configuration and Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 969
Certificate Template Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 969
Certificate Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 970
Configure Certificate Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 970
Switching to Secure Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 970
Purchasing and Installing a Certificate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 970
Creating/Modifying System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 971
Enabling Secure Communication Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 971
Alternate Management Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 973
Restoring Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 973
Appendix G - Configuring Direct Connections to the Collaboration Server . . 974
Management Network (Primary) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 974
Alternate Management Network (2000/4000) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 974
Configure the Workstation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 974
Connect to the Management Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 978
Connect to the Alternate Management Network (2000/4000) . . . . . . . . . . . . . . . . . . . . . . . . 980
Connect to the Collaboration Server using a Modem (2000/4000) . . . . . . . . . . . . . . . . . . . . 982
Procedure 1: Install the RMX Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 982
Procedure 2: Configure the Modem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 982
Procedure 3: Create a Dial-up Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 982
Procedure 4: Connect to the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 987
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Appendix H - Deployment Into Microsoft Environments . . . . . . . . . . . . . . . . . 988
Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 988
Lync 2013 SVC Connectivity to RealPresence Collaboration Servers . . . . . . . . . . . . . . . . . 989
Deployment Architectures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 989
Single Premise Deployment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 989
RealPresence Hosted Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 991
RealConnect Cascaded Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 992
Remote Premise Deployment - RealConnect via Service Provider . . . . . . . . . . . . . . . . 994
Organization Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 995
Process Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 995
Supported Deployment Topology . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 996
Backward Compatibility to Lync 2010 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 997
Video Resource Requirements and Implications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 997
Support for HD1080p Resolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 998
Limit Maximum Resolution for MS SVC Using a System flag . . . . . . . . . . . . . . . . . . 998
FEC (Forward Error Correction) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 998
IPv6 Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 998
DHCPv6 Support for Auto IPv6 Address Assignment . . . . . . . . . . . . . . . . . . . . . . . . . . . 999
System Flags for Cropping Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 999
Sharing Content During a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1000
RealPresence Polycom® ContentConnect™ Gateway . . . . . . . . . . . . . . . . . . . . . 1000
RealPresence Polycom® ContentConnect™ Plug in . . . . . . . . . . . . . . . . . . . . . . . 1001
Cisco TIP Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1004
Lync 2013 Participant Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1004
Monitoring Participant Properties - Channel Status Tab . . . . . . . . . . . . . . . . . . . . . 1005
Monitoring Participant Properties - Channel Status - Advanced Tab . . . . . . . . . . . 1006
DNS Load Balancing on Lync Front End Pool . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1006
Video Resource Requirement Selection in Lync 2013 AVMCU Cascade . . . . . . . . . . . . . . 1009
Video Forcing and Changing Layout in Lync 2013 AVMCU Cascade . . . . . . . . . . 1010
Handle Low Bit Rate Calls From the AVMCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1010
Remove Empty Cells From the Video Layout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1010
Initiate and Connect to a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1011
Meet Now Lync Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1011
RealConnect for Lync 2013 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1012
Cascading Conference Reestablishment Process . . . . . . . . . . . . . . . . . . . . . . . . . 1012
Reestablished Cascading Conference Termination . . . . . . . . . . . . . . . . . . . . . . . . 1012
Call Admission Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1013
Conferencing Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1013
Monitoring Participant Connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1014
Collaboration Server Chairperson vs. AVMCU Meeting Organizer . . . . . . . . . . . . 1015
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Active Alarms and Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1016
Active Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1016
ICE Active Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1017
Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1019
Known Issues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1019
Appendix I - Polycom Open Collaboration Network (POCN) . . . . . . . . . . . . . 1020
Collaboration With Cisco’s Telepresence Interoperability Protocol (TIP) . . . . . . . . . . . . . . 1020
Deployment Architectures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1021
Single Company Model - Polycom and Cisco Infrastructure . . . . . . . . . . . . . . . . . . . . . 1021
Call Flows . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1025
Multipoint call with DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1025
Multipoint call without DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1026
Company to Company Models Using a Service Provider . . . . . . . . . . . . . . . . . . . . . . . 1027
Model 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1027
Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1029
Multipoint call via Service Provider - Model 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1029
Multipoint call via Service Provider - Model 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1030
Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1032
Multipoint call via Service Provider - Model 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1032
Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1033
Gatekeepers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1033
Standalone Polycom DMA System as a Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . 1033
Standalone Cisco IOS Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1033
Neighbored Cisco IOS and Polycom DMA Gatekeeper . . . . . . . . . . . . . . . . . . . . . 1033
CUCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1033
Configuring the Cisco and Polycom Equipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1033
Cisco Equipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1034
CUCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1034
IOS Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1034
IOS and DMA Gatekeepers (Neighbored) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1034
Polycom Equipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1034
Configuring the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1035
Configuring DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1035
Configuring Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1035
Configuring Entry Queues and IVR Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1036
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1036
Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1036
Procedure 1: Set the MIN_TIP_COMPATIBILITY_LINE_RATE System Flag . . . . 1037
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Procedure 2: Configuring Collaboration Server to statically route outbound SIP
calls to DMA or CUCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1038
Procedure 3: Configuring the Collaboration Server’s H.323 Network Service
to register with DMA gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1039
Procedure 4: Configuring a TIP Enabled Profile on the Collaboration Server . . . . 1040
Content Sharing Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1045
Procedure 5: Configuring an Ad Hoc Entry Queue on the Collaboration Server
if DMA is not used . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1046
Procedure 6: Configuring a Meeting Room on the Collaboration Server . . . . . . . . 1046
Procedure 7: Configuring Participant Properties for dial out calls . . . . . . . . . . . . . 1046
Collaboration with Microsoft and Cisco . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1047
Deployment Architecture: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1048
Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1051
Multipoint Calls using DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1051
Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1052
DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1052
Microsoft Lync Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1052
CUCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1052
Solution Interoperability Table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1052
TIP Layout Support & Resource Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1054
Supported TIP Resolutions and Resource Allocation . . . . . . . . . . . . . . . . . . . . . . . . . . 1054
Supported Resolutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1054
Resource Allocation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1054
Configuring the Microsoft, Cisco and Polycom Components . . . . . . . . . . . . . . . . . . . . . . . 1055
Content Sharing Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1060
Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1061
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1061
Resolution Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1064
Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1065
Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1065
Operations During Ongoing Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1066
Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1066
CTS Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1066
Lync Participants (RTV) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1068
Known Limitations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1069
Appendix J - Restore Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1071
Perform a Standard Restore . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1071
Perform a Comprehensive Restore . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1073
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Appendix K - SIP RFC Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1077
Appendix L - Media Traffic Shaping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1079
Traffic Shaping Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1079
System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1079
Capacity Reduction During Traffic Shaping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1080
System Limitation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1080
Appendix M - Homologation for Brazil . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1081
H.323 & SIP Protocol Flag Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1081
H.323 & SIP Flag Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1081
Flag name: SIP_TIMERS_SET_INDEX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1081
Flag name: H323_TIMERS_SET_INDEX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1082
Flag name: DISABLE_DUMMY_REGISTRATION . . . . . . . . . . . . . . . . . . . . . . . . . 1082
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Overview
About RealPresence Collaboration Server (RMX)
Administrator Guide
RealPresence Collaboration Server (RMX) Administrator Guide provides instructions for configuring,
deploying, and administering Polycom Multipoint Control Units (MCUs) for video conferencing. This guide
will help you understand the Polycom video conferencing components, and provides descriptions of all
available conferencing features.
This guide will help you perform the following tasks:
● Customize the Collaboration Server conferencing entities such as conference Profiles, IVR Services,
Meeting Rooms, Entry Queues, etc., to your organization’s needs (optional).
● Define Collaboration Server Users. Further customize the Collaboration Server IP Network Settings
for IPv6 environments.
● Further customize the Collaboration Server Network Settings for ISDN networks and IP networks for
Ultra Secure Mode environments.
● Define video protocols and resolution configuration for CP conferencing.
● Configure Templates, the Address Book and schedule Reservations (optional).
● Record conferences.
● Configure the Collaboration Server to support special call flows and conferencing requirements, such
as Cascading Conferences.
● Configure the Collaboration Server to act as a gateway and manage gateway calls.
● Configure the Collaboration Server to support Polycom third-party and partner environments, such
as Microsoft, IBM, Cisco, Avaya, Broadsoft and Siemens.
● Configure the Collaboration Server for special needs by setting various system flags.
● Manage and troubleshoot the Collaboration Server’s performance.
The RealPresence Collaboration Server (RMX) 1800/2000/4000 Getting Started Guide provides a
description of basic conferencing operations. It will help you perform the following tasks:
● First time installation and configuration, including unpacking, installing, power up and initial setup.
● Selecting a language.
●
●
●
Starting a conference using default profiles.
Troubleshooting.
Using the RMX Web Client
For information about Resource Capacities and other operational specifications, see the RealPresence®
Collaboration Server (RMX) Hardware Guide specific to your system.
The RealPresence Collaboration Server (RMX) 1800/2000/4000 Deployment Guide for Maximum Security
Environments provides a deployment methodology for system administrators implementing Maximum Security
Environments.
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Overview
Note: Not supported on Collaboration Server (RMX) 1800 with no DSP cards
• Integration into partner and third-party environments ( Microsoft, IBM, Cisco, Avaya, Broadsoft,
Siemens).
• Gateway functionality.
• Cascading Conferences.
Who Should Read This Guide?
System administrators and network engineers should read this guide to learn how to properly set up
Polycom Collaboration Server systems. This guide describes administration-level tasks.
For a detailed description of first time installation and configuration, description of the Collaboration Server
(RMX) Web Client, and basic operation of your Collaboration Server system, see RealPresence
Collaboration Server (RMX) 1800/2000/4000 Getting Started Guide.
Prerequisites
This guide assumes the user has the following knowledge:
● Familiarity with Windows® XP or Windows 7 operating systems and interface.
● Familiarity with 32-bit Microsoft® Internet Explorer® Version 7, 8, 9 or 10.
● Basic knowledge of video conferencing concepts and terminology.
How This Guide is Organized
The following typographic conventions are used in this guide to distinguish types of in-text information.
Typographic Conventions
Convention
Description
Bold
Highlights interface items such as menus, soft keys, flag names, and directories. Also used
to represent menu selections and text entry to the phone.
Italics
Used to emphasize text, to show example values or inputs, file names and to show titles of
reference documents available from the Polycom Support Web site and other reference
sites.
Underlined Blue
Used for URL links to external Web pages or documents. If you click on text in this style,
you will be linked to an external document or Web page.
Blue Text
Used for cross referenced page numbers in the same or other chapters or documents. If
you click on blue text, you will be taken to the referenced section.
Also used for cross references. If you click the italic cross reference text, you will be taken
to the referenced section.
<variable name>
Indicates a variable for which you must enter information specific to your installation,
endpoint, or network. For example, when you see <IP address>, enter the IP address of
the described device.
>
Indicates that you need to select an item from a menu. For example, Administration >
System Information indicates that you need to select System Information from the
Administration menu.
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Overview
About the RealPresence Collaboration Server System
The RealPresence Collaboration Server system is a high performance, scalable, IP-network (H.323 and
SIP) ISDN/PSTN MCU that provides feature-rich and easy-to-use multipoint voice and video conferencing.
The Collaboration Server 2000/4000 meets International Telecommunication Union - Telecommunication
Standardization Sector, (ITU-T, formerly CCITT) standards for multipoint multimedia bridging devices, and
meets ETSI standards for telecommunication products. In addition, it has been designed in compliance with
IETF (Internet Engineering Task Force).
The MCU can be used as a standalone device to run voice and video conferences or it can be used as part
of a solution provided by Polycom. This solution may include the following components:
● Polycom® Distributed Media Application™ (DMA™) system - Provides call control and MCU
virtualization with carrier-grade redundancy, resiliency and scalability.
Connection to RealPresence DMA MCU Pools and DMA Cascading are not supported on
Collaboration Server (RMX) 1800 with no DSP cards.
● Polycom RealPresence Resource Manager - Centrally manages, monitors and delivers Cloud based
Video as a Service (VaaS) and enterprise video collaboration.
● Polycom® RealPresence® Access Director™ - Removes communication barriers and enables
internal and external teams to collaborate more easily and effectively over video.
● Polycom® RealPresence® Media Suite - Provides one-touch recording and secure playback on
telepresence and video conferencing systems, tablets and smart phones, or from your Web browser.
The following diagram describes the multipoint video conferencing configuration with the Collaboration
Server as a standalone system.
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Overview
Multipoint Video Conferencing using a RealPresence Collaboration Server (RMX) 1800/2000/4000
The RealPresence Collaboration Server system can be controlled via the LAN by the Collaboration Server
Web Client application using Internet Explorer installed on the user’s workstation or the RMX Manager
application. The RMX Manager can control several MCU units. For more information about the RMX
Manager, see RMX Manager Application.
Network Services Guidelines
IP Networks
In Collaboration Server 2000 system management and IP conferencing are performed via a single LAN port.
In Collaboration Server 1800/4000 system management and IP conferencing are performed on separate
LAN ports.
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Overview
In Collaboration Server 2000/4000 the networks can be separated in Maximum Security Environments.
ISDN Networks
RealPresence Collaboration Server (RMX) 1800 with 3 DSP cards with built-in ISDN module, supports one
ISDN card with up to 4 E1/T1 PRI lines.
RealPresence Collaboration Server (RMX) 2000/4000 support a maximum of two RTM ISDN cards, each
providing connection for up to either 7 E1 or 9 T1 PRI lines.
On all Collaboration Servers, E1 and T1 connections cannot be used simultaneously.
RealPresence Collaboration Server (RMX) 1800 with no DSP cards does not support ISDN/PSTN
connections.
For more detailed information about Collaboration Server abilities, see Hardware Description in the
RealPresence Collaboration Server (RMX) Hardware Guides.
Card Configuration Modes
Note: Media Cards
This section is applicable only for MCUs with MPM media card(s), meaning Collaboration Servers
2000/4000 only.
The media card installed in the system determines the Card Configuration Mode. The Card Configuration
Mode represents different generations of the media card. Each new generation provides additional
functionality, higher video resolutions and higher resource capacity.
Only one Media Card type can be installed in any Collaboration Server, which sets the Card Configuration
Mode for that Collaboration Server.
Viewing the Card Configuration Mode
The Card Configuration Mode is determined according to the installed media card.
The Licensing Mode and the Card Configuration Mode for your MCU can be viewed in the System
Information dialog box (Administration > System Information).
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Overview
In the example shown here, the Card Configuration Mode is MPMRx.
RealPresence Collaboration Server Supported Features
Features Supported in RealPresence Collaboration Server
Feature Name
Description
Scalable Video Coding (SVC)
Scalable Video Coding (SVC) Conferencing, based on the SVC video protocol
and SAC audio protocol.
SVC Conferencing offers high resolution video conferencing with low
end-to-end latency, improved Error Resiliency and higher system capacities.
w448 Resolution support
Improves interoperability with Tandberg MXP 990/3000 endpoints providing
these endpoints the resolution of W448p (768x448 pixels) at 25fps.
HD H.264 Content and H.264
Content for Cascading links
Enables conference participants to receive higher quality Content in both
standard conferences and cascaded conferences.
Site Names
Additional controls over the display of site names in the conference Profile.
Interactive Video Forcing
Participants in ongoing conferences can be interactively forced to a Video
Window in the conference layout by using Drag and Drop.
Video Preview
H.264 High Profile is supported with Video Preview.
Recording indication
A Recording Indication can be displayed to all conference participants
informing them that the conference is being recorded.
Note: Only in non-TIP conferences.
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Overview
Features Supported in RealPresence Collaboration Server
Feature Name
Description
Network Quality Indication
A Network Quality Indicator is displayed for each participant in the CP layout
indicating the quality of the participants’ video channels.
Note: Only in non-TIP conferences.
Auto scan and Customized
Polling
A single cell in the conference layout is used to cycle the display of participants
that are not in the conference layout. The order of the cyclic display can be
predefined.
SirenLPR
Prevents audio degradation and maintains high audio (CD) quality if packet loss
occurs.
Speaker Change Threshold
The option to configure the amount of time a participant must speak
continuously until becoming the speaker.
Integration with Cisco
Telepresence Systems (CTS)
The MCU natively inter-operate with Cisco TelePresence Systems and
Polycom TelePresence and vide conferencing endpoints, ensuring optimum
quality multi-screen, multipoint calls.
Telepresence Mode is not supported by Collaboration Server (RMX) 1800 with
no DSP cards.
POCN - Collaboration with
Microsoft and Cisco in the same
environment
The POCN solution, enables Polycom, Microsoft and Cisco users, each within
their own environment, to participate in the same conference running on a
Collaboration Server.
Additional Chinese Font Types
Additional Chinese fonts may be selected for several features when using the
Collaboration Server in Chinese.
Support for Microsoft Protocols, algorithms and workflows
RTV Video Protocol
Microsoft RTV Video protocol is supported.
RTV B-Frame Support
B-frame encoding and decoding are supported to enhance the viewing
experience of Microsoft Lync clients. It provides higher video quality at the
same bit-rate with better scalability on the decoder side.
Conferencing Entities Presence
in Microsoft Office
Communications Server Client or
Lync Server Client
Registration & Presence enables the OCS or LYNC client users to see the
availability status (Available, Busy or Offline) of Meeting Rooms, Entry Queues
and SIP Factories and connect to them directly from the buddy list.
Cascading between
Collaboration Server Meeting
room / Microsoft A/V MCU
Microsoft Lync users can connect a Collaboration Server Meeting Room to a
conference running on the Microsoft A/V MCU.
Microsoft Lync is not supported by Collaboration Server (RMX) 1800 with no
DSP cards.
FEC Support
Support of Microsoft RTV FEC (Forward Error Correction) that controls and
correct packet loss when receiving and sending video streams using the
Microsoft Lync Server 2010.
ICE Over TCP
Enables the automatic usage of the ICE connection through the TCP port
instead of UDP when the UDP port in the firewall is blocked.
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Overview
Features Supported in RealPresence Collaboration Server
Feature Name
Description
Media Over TCP
Media is automatically transmitted using TCP when UDP, the default transport
protocol, is not available.
Error Recovery
The Collaboration Server can automatically recover from short duration
network errors (5 seconds), enabling calls in Microsoft Lync to continue video
or audio conferences without disconnecting.
The RMX Web Client and RMX Manager applications can be installed in an environment that meets the
following requirements:
● Minimum Hardware – Intel® Pentium® III, 1 GHz or higher, 1024 MB RAM, 500 MB free disk space.
● Workstation Operating System – Microsoft® Windows® XP, Windows® 7, and Windows® 8.
● Network Card – 10/100/1000 Mbps.
● Web Browser - Microsoft® Internet Explorer® Version 7, 8, 9, and 10.
● Collaboration Server Web client and RMX Manager are optimized for display at a resolution of 1280
x 800 pixels and a magnification of 100%
The following table lists the environments (Web Browsers and Operating Systems) with which the
Collaboration Server Web Client and RMX Manager applications are supported.
Collaboration Server Web Client/RMX Manager Environment Interoperability Table
Web Browser
Operating System
Internet Explorer 7
Windows Vista™
Windows 7
Internet Explorer 8
Windows 7
Internet Explorer 9
Windows 7 and Windows 8
Internet Explorer 10*
Windows 8
Software Prerequisites and Guidelines
● .Net Framework 3.5 SP1 is required and installed automatically. Internet Explorer must be enabled
to allow running Signed ActiveX.
If ActiveX installation is blocked, see the ActiveX Bypass.
●
●
●
Collaboration Server Web Client does not support larger Windows text or font sizes. It is recommended to set
the text size to 100% (default) or Normal in the Display settings in Windows Control Panel on all workstations.
Otherwise, some dialog boxes might not appear properly aligned. To change the text size, select Control Panel
> Display. For Windows XP, click the Appearance tab, select Normal for the Font size, and click OK. For
Windows 7, click the Smaller - 100% option, and click OK.
When installing the Collaboration Server Web Client, Windows Explorer >Internet Options> Security Settings
must be set to Medium or less.
It is not recommended to run RMX Web Client and Polycom CMAD applications simultaneously on the same
workstation.
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Overview
● If you have problems getting the Collaboration Server Web Client to work with Windows 8, it is
recommended to run Internet Explorer as an administrator by holding the shift key and right-clicking
on the IE icon, and then select Run as Administrator.
For Windows 7™ Security Settings and Internet Explorer 8 configuration, refer to Windows 7™ Security
Settings and Internet Explorer 8 configuration sections in RealPresence Collaboration Server (RMX)
1800/2000/4000 Getting Started Guide.
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Conferencing Modes Overview
The MCU system offers the following types of conferences (Conferencing Modes), based on the video
protocol and the video display during the conference:
● AVC-based Conferencing - CP Only (Video Transcoding)
● AVC-based Conferencing - Video Switching
● SVC-based Conferencing (Media Relay) - SVC Only
● Mixed AVC and SVC Conferencing - CP and SVC
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
AVC Conferencing
AVC-based Conferences allow endpoints that support AVC video to connect to these conferences. AVC
(Advanced Video Coding) video refers to the H.264 video protocols used to send and receive video. On the
Collaboration Server system it also includes all the standard video protocols such as H.261, H.263, and
RTV.
All endpoints (including SVC-enabled endpoints) have AVC capabilities and can connect to AVC
conferences running on the MCU. AVC-based endpoints can connect using different signaling protocols and
different video protocols.
Based on the video processing required during the conference, the Collaboration Server offers the
Continuous Presence Conferencing Mode for AVC-based conferencing.
The MCU also offers the Video Switching conferencing Mode.
The Conferencing Mode determines the video display options (full screen or split screen with all participants
viewed simultaneously) and the method in which the video is processed by the MCU (with or without using
the MCU’s video resources).
Continuous Presence (CP) Conferencing
The dynamic Continuous Presence (CP) capability of the Collaboration Server system enables viewing
flexibility by offering multiple viewing options and window layouts for video conferencing. It enables several
participants to be viewed simultaneously and each connected endpoint uses its highest video, audio and
data capabilities up to the maximum line rate set for the conference.
AVC-based endpoints can connect to the conference using any:
● Signaling protocol: H.323, SIP, ISDN/PSTN and RTV line rate, up to a maximum line rate defined for
the conference
● Video Protocol: H.261, H.263, H.264 Base Profile and H.264 High Profile) and at any resolution and
frame rate, provided they meet the minimum requirements set for the conference:
 Video Resolutions: from QCIF, CIF and up to 1080p60
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Conferencing Modes Overview
 Frame rates up to 60fps
In Continuous Presence conferences, the MCU receives the video stream from each endpoint at the video
rate, video resolution and frame rate that it is capable of sending, and it superimposes all the received
streams into one video stream that includes the input from the other endpoints arranges in the selected
video layout.
Participants do not see themselves in the video layout. By Default, the speaker is shown in the top left layout
cell in symmetric layouts, in the larger cell in asymmetric layouts, or in full screen. The speaker sees the
previous speakers (their number depends on the number of cells on the speaker’s layout.
The Continuous Presence video session offers layouts to accommodate different numbers of participants
and conference settings including support of the VUI annex to the H.264 protocol for endpoints that transmit
wide video instead of 4CIF resolution. Each participant can select his/her layout for viewing during the
conference, as can be seen in AVC Continuous Presence (CP) video streams and built layouts.
For conferences with more participants than display squares, the RealPresnce Collaboration Server
dynamic video mix capability allows the viewed sites to be modified throughout the conference. The
displayed layout can be changed during an ongoing conference, allowing a participant to view different
screen layouts of the other conference participants. These layout options allow conferences to have greater
flexibility when displaying a large number of participants and maximizes the screen’s effectiveness.
AVC Continuous Presence (CP) video streams and built layouts
Video quality in Continuous Presence conferences is affected by the conference line rate (that determines
the maximum line rate to be used by the connecting endpoints), and the video capabilities of the endpoints
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Conferencing Modes Overview
such as the video protocol, video resolution and frame rate. Content sharing is available in all CP
conferences.
This requires extensive processing of the video sent to each participant in the conference. The higher the
video rate and resolution, the more processing power is required.
By default every conference, Entry Queue and Meeting Room has the ability to declare the maximum CP
resolution as defined for the system. This includes conferences launched by the Collaboration Server Web
Client and conferences started via the API.
CP conferencing is defined in the Conference profile by setting the following main
features:
● Setting the Conferencing Mode to CP only
● Conference Line Rate
● Video Quality – Motion or Sharpness
● Video Layout
Video Protocol Support in CP Conferences
The video protocol selected by the system determines the video compression standard used by the
endpoints. In Continuous Presence conferences, the system selects the best video protocol for each of the
endpoint according to he endpoint’s capabilities.
The following Video protocols are supported in CP conferences:
● H.261 - The legacy video compression algorithm mandatory to all endpoints. It is used by endpoints
that do not support other protocols.
● H.263 - A video compression algorithm that provides a better video quality than H.261. This standard
is not supported by all endpoints.
● H.264 Base Profile - A video compression standard that offers improved video quality, especially at
line rates lower than 384 Kbps.
H.264 High Profile - Allows higher quality video to be transmitted at lower line rates.
● RTV - A video protocol that provides high quality video conferencing capability to Microsoft OCS
(Office Communicator Server) endpoints at resolutions up to HD720p30. (SIP only).
Video Switching (VSW) Conferencing
In Video Switching mode all participants see the same video picture (full screen). The current speaker is
displayed in full screen on all the participants' endpoints, while the speaker sees the previous speaker.
Switching between participants is voice-activated; whenever a participant starts to speak, he or she
becomes the conference speaker and is viewed on all screens. All conference participants must use the
same line rate and video parameters such as video protocol, frame rate, annexes and interlaced video mode
as no video processing is performed. Endpoints that are unable to meet these requirements connect as
Secondary (audio only).
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Conferencing Modes Overview
AVC Video Switching (VSW) video streams and Full Screen Layout
Video Switching (VSW) Conferencing Guidelines
You can run VSW conferences according to the following the guidelines:
● Video Switching conferences can be set to one of the following resolutions, depending on the
capabilities of the endpoints connecting to the conference:
 H.264 1080p60 (Symmetrically, at bit rates of up to 6Mbps).
 HD1080p60
 H.264 1080p30
 H.264 720p30
 H.264 720p60
 H.264 SD 30
 H.264 CIF
 H.263 CIF
 H.261 CIF
● Both H.264 Base and High Profile video protocols are supported.
● Video Switching conferencing mode is unavailable to ISDN participants.
● Video Switching uses fewer system resources than CP: only one CIF video resource per participant
for any resolution (including HD).
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Conferencing Modes Overview
● The display aspect ratio is 4x3 or 16x9.
● Site (endpoint) names, skins, message overlay etc. are not supported in Video Switching.
● Video forcing is enabled at the conference and participant levels.
● To connect to a Video Switching conference via Entry Queue, the Profile assigned to the Entry Queue
must be set to Video Switching. It is recommended to use the same profile for both the destination
conference and Entry Queue.
● Telepresence Mode is unavailable in Video Switching conferences.
● In RealPresence Collaboration Server (RMX) 2000/4000 with MPMRx media cards, each participant
connecting to a VSW conference consumes one HD720 port.
● The HD_THRESHOLD_BITRATE flag must be set in the System Configuration. The value of this flag
is the system minimum threshold bit rate for HD resolutions. The line rate selected in the conference
Profile must be the same as or higher than that specified by the HD_THRESHOLD_BITRATE flag.
Note: Threshold bit rate
The HD_THRESHOLD_BIT RATE flag is responsible for negotiation only, It does not guarantee that
the endpoint will open an HD channel or transmit on an opened HD channel.
The HD_THRESHOLD_BITRATE flag line rate value ranges from 384kbps to 4Mbs, default is
768kbps. For more information, see Modifying System Flags.
AVC Basic Conferencing Parameters
The main parameters that define the quality of an AVC-based video conference and its display are:
● Line (Bit) Rate - The transfer rate of video and audio streams. The higher the line (bit) rate, the better
the video quality. The MCU supports the following line rates:
 Collaboration Server 2000/4000 with MPMRx card - 64kbps to 6144kpbs
 Collaboration Server 1800 - 64kbps to 6144kpbs
● Audio Algorithm - The audio compression algorithm determines the quality of the conference audio.
● Video protocol, video format, frame rate, annexes, and interlaced video mode - These
parameters define the quality of the video images. The RealPresnce Collaboration Server will send
video at the best possible resolution supported by endpoints regardless of the resolution received
from the endpoints.
 When Sharpness is selected as the Video Quality setting in the Conference Profile, the
RealPresnce Collaboration Server will send 4CIF (H.263) at 15fps instead of CIF (H.264) at 30fps.
 H.264 High Profile protocol provides better compression of video images in line rates lower than
384 Kbps and it will be automatically selected for the endpoint if it supports H.264 High Profile. If
the endpoint does not support H.264 High Profile, the RealPresnce Collaboration Server will try
H.264 Base Profile which provides good compression of video images in line rates lower than
384 Kbps (better than H.263 and not as good as H.264 High Profile).
 When working with RealPresnce Collaboration Servers at low bit rates (128, 256, or 384Kbps),
HDX endpoints will transmit SD15 resolution instead of 2CIF resolution.
When using a full screen (1x1) conference layout, the RealPresnce Collaboration Server transmits
the same resolution it receives from the endpoint.
● Video resolution:
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Conferencing Modes Overview
 H.261 CIF/QCIF – Supported in Continuous Presence (CP) conferences at resolutions of 288 x
352 pixels (CIF) and 144 x 176 pixels (QCIF). Both resolutions are supported at frame rates of up
to 30 frames per second.
 H.263 4CIF - A high video resolution available to H.263 endpoints that do not support H.264. It is
only supported for conferences in which the video quality is set to sharpness and for lines rates
of 384kbps to 1920kbps.
 Standard Definition (SD) - A high quality video protocol which uses the H.264 and H.264 High
Profile video algorithms. It enables compliant endpoints to connect to Continuous Presence
conferences at resolutions of 720 x 576 pixels for PAL systems and 720 x 480 pixels for NTSC
systems.
 High Definition (HD) – HD is an ultra-high quality video resolution that uses the H.264 and H.264
High Profile video algorithms. Depending on the RealPresnce Collaboration Server’s type,
compliant endpoints are able to connect to conferences at the following resolutions:
 720p - 1280 x 720 pixels
 1080p - 1920 x 1080 pixels
● Lost Packet Recovery (LPR) - LPR creates additional packets that contain recovery information
used to reconstruct packets that are lost during transmission.
Supplemental Conferencing Features
In addition to basic parameters that determine the quality of the video, additional features can be enabled,
adding capabilities to the conference, or enabling special conferencing modes:
● Content Sharing (H.239) – Allows compliant endpoints to transmit and receive two simultaneous
streams of conference data to enable Content sharing. H.239 is also supported in cascading
conferences. Both H.263 and H.264 Content sharing protocols are supported. If all endpoints
connected to the conference have H.264 capability, Content is shared using H.264, otherwise
Content is shared using H.263.
For more information, see Sharing Content During Conferences.
● Video Clarity (CP Conferences only) - Video Clarity applies video enhancing algorithms to
incoming video streams of resolutions up to and including SD.
Video Clarity is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
● Encryption – Used to enhance media security at conference and participant levels.
For more information, see Implementing Media Encryption for Secured Conferencing.
● Conference Recording and Webcasting - The Collaboration Server enables audio and video
recording, streaming, and webcasting of conferences using the Polycom® RealPresence® Media
Suite.
● Lecture Mode (CP Conferences only) – The lecturer is seen by all participants in full screen while
the lecturer views all conference participants in the selected video layout.
For more information, see Lecture Mode.
● Presentation Mode (CP Conferences only) – When the current speaker’s speech exceeds a
predefined time (30 seconds), the conference layout automatically changes to full screen, displaying
the current speaker as the conference lecturer on all the participants’ endpoints. During this time the
speaker’s endpoint displays the previous conference layout. When another participant starts talking,
the Presentation Mode is cancelled and the conference returns to its predefined video layout.
Presentation mode is available with Auto Layout and Same Layout.
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Conferencing Modes Overview
 If the speaker in a video conference is an Audio Only participant, the Presentation Mode is
disabled for that participant.
 Video forcing works in the same way as in Lecture Mode when Presentation Mode is activated,
that is, forcing is only enabled at the conference level, and it only applies to the video layout
viewed by the lecturer.
● Telepresence Mode (CP Conferences only) - Enables the connection of numerous high definition
telepresence rooms and of different models (such as TPX and RPX) into one conference maintaining
the telepresence experience. This mode is enabled by a special license.
Telepresence Mode is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
● TIP Support (CP Conferences only) - TIP is a proprietary protocol created by Cisco for deployment
in Cisco TelePresence systems (CTS). Polycom’s solution is to allow the Collaboration Server to
natively inter-operate with Cisco TelePresence Systems, ensuring optimum quality multi-screen,
multipoint calls.
SVC-based Conferencing
The SVC-Based conferencing mode provides video without transcoding by the MCU, hence requiring less
video resources while providing better error resiliency and lower latency.
Using the SVC video protocol, SVC conferences provide video bit streams at different resolutions, frame
rates and line rates to SVC-enabled endpoints with various display capabilities and layout configurations.
In the SVC-based conference, each SVC-enabled endpoint transmits multiple bit streams, called
simulcasting, to the RealPresence® Collaboration Server. Simulcasting enables each endpoint to transmit
at different resolutions and frame rates such as 720p at 30fps, 15fps, and 7.5fps, 360p at 15fps and 7.5fps,
and 180p at 7.5fps.
The Polycom SVC-enabled endpoints (such as Polycom® RealPresence® Desktop and Polycom®
RealPresence® Mobile) compose the layout according to their layout settings and video capabilities. This
enables the MCU to send or relay the selected video streams to each endpoint without processing the video
streams and sending the composite video layout to the endpoints.
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Conferencing Modes Overview
SVC video streams and Layouts
-
The video streams displayed in the conference layout on each endpoint is obtained from the different
streams received from each of the endpoints displayed in the layout. Depending on the size of the video cell
in the configured layout, the endpoint requests the video stream in the required resolution from the
RealPresence Collaboration Server. The higher the display quality and size, the higher the requested
resolution will be sent to the endpoint. The endpoint creates the displayed layout from the different video
streams it receives.
For instance, an SVC endpoint might want to receive three video streams at different frame rates and
resolutions, and create a conference layout with the received video streams. Each SVC-enabled endpoint
sends encoded SVC bit streams to the MCU to relay to the other SVC-enabled endpoints in the conference.
The endpoints encode the video in multiple resolutions and decodes the multiple video input streams.
For example:
RealPresence mobile client (2) will transmit two resolutions; one that is suited for RealPresence Desktop
client (3) and a second that is suited for two other endpoints: RealPresence Desktop client (4) and (1).
RealPresence Desktop client (1) transmits two resolutions; one that is suited for RealPresence Mobile client
(2) and a second that is suited for RealPresence Desktop client (4).
The MCU determines which of the incoming resolutions to send to each endpoint. It does not perform any
SVC encoding and decoding, or any transcoding of the video streams. The RealPresence Collaboration
Server functions as the multipoint media relay to the endpoints. For voice activated selection of the video
streams, the RealPresence Collaboration Server determines which of the incoming bit streams to send to
each endpoint.
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Conferencing Modes Overview
Advantages of SVC Conferencing
SVC increases the scalability of video networks and enables mass desktop video deployments. Some of
the advantages of SVC conferencing are:
● Offers high-resolution video conferencing with low end-to-end latency, improved error resiliency and
higher system capacities.
● Allows the SVC-enabled video endpoints to manage display layouts, supporting multiple line rates,
resolutions and frame rates.
● The RealPresence Collaboration Server functions as a media relay server providing low cost
production benefits. The RealPresence Collaboration Server reduces bandwidth usage by only
selecting the necessary video stream to be sent to the endpoints.
SVC Conferencing Guidelines
You can run SVC-based conferences when following the guidelines listed below.
● SVC conferences are supported only with the following:
 SVC Licensing
 SIP over UDP signaling
 SIP over TLS Signaling
 Polycom SVC-enabled endpoints (Polycom® RealPresence® Desktop, Polycom®
RealPresence® Mobile)
 Ad Hoc conferencing via Meeting Rooms and ongoing conferences
● SVC Only conferences can run on the same MCU as AVC Only conferences.
● On RealPresence Collaboration Server (RMX) 2000/4000, all the endpoints participating in a single
SVC Only conference must be connected to the same media card and cannot be handled by different
media cards as the SVC media streams cannot be shared between them.
● End-to-end latency on a local network (same site), is around 200msec to ensure AV sync (also known
as Lip-sync).
● Dial-out is not available in SVC Only conference.
● Dial-in is available as follows:
 AVC endpoints (participants) can only connect to an AVC conference or Mixed CP and SVC
conference. When dialing into SVC Only conferences they will be disconnected and the calls fail.
 SVC endpoints support both AVC and SVC video protocols:
 When dialing into SVC Only conferences, they connect as SVC endpoints.
 When dialing into AVC Only conferences, they connect as AVC endpoints. They cannot
connect to an AVC conference using the SVC capabilities.
● SVC endpoints can connect to conferences via Entry Queues, however:
 The Entry Queue and Conference Modes must match - both SVC Only or both Mixed.
 Both the Entry Queue and the Conference must have the same line rate.
● SVC endpoints cannot be moved between conferences.
● Content is supported in H.264 (AVC).
 Only the H.264 Cascade and SVC Optimized option is supported.
 LPR and DBA are not supported for SVC content sharing.
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Conferencing Modes Overview
● In SVC Only conferences and Mixed CP and SVC conferences, Auto Layout is the default and the
layout display for SVC endpoints is controlled from the endpoint application.
● Site names display on SVC endpoints is controlled from the SVC endpoints.
● When Hot backup is enabled, all the conferences are created on the Slave MCU.
Hot Backup is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
● When Hot Backup is activated and the Slave MCU becomes the Master MCU:
 All AVC endpoints will be reconnected to the AVC (CP and VSW) conferences. SVC endpoints
connected to AVC conferences using their AVC capabilities will be reconnected to their AVC
conferences.
 SVC endpoints cannot be reconnected to their SVC Only conferences as dial-out is not supported
for SVC endpoints. These endpoints will have to manually reconnect to their SVC conferences.
● Cascading between SVC Only conferences or between AVC and SVC Only conferences is not
supported.
● Gateway sessions are not supported for SVC calls.
● Reservations cannot be scheduled for SVC Only conferences.
● The following functionality and features are not supported during SVC Only conferences:
 FECC
 Skins. The video cells are displayed on the endpoint’s default background.
 IVR functionality
 Conference Gathering phase
 All DTMF enabled features during the conference
 Manual selection of video layout
 Chairperson functionality
 Media Encryption
 Recording of SVC Only conferences
 Text messaging using Message Overlay
MCU Supported Resolutions for SVC Conferencing
The MCU automatically selects the resolution and frame rate according to the conference line rate. The
table below details the maximum resolution and frame rates supported by the MCU for each conference line
rate. The actual video rate, resolution and frame rates displayed on each endpoints is determined by the
endpoint’s capabilities.
SVC Conferencing - Maximum Supported Resolutions per Simulcast Stream
Conference Line Rate
(kbps)
Profile
Maximum
Resolution
Max. Frame
Rate (fps)
Audio Rate
(kbps)
2048 - 4096
High Profile
1080p
30fps
48
1472 - 2048
High Profile
720p
30fps
48
1024 - 1472
High Profile
720p
15fps
48
768 - 1024
High Profile
720p
15fps
48
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Conferencing Modes Overview
SVC Conferencing - Maximum Supported Resolutions per Simulcast Stream
Conference Line Rate
(kbps)
Profile
Maximum
Resolution
Max. Frame
Rate (fps)
Audio Rate
(kbps)
512 - 768
High Profile
360p
30fps
48
256 - 512
Base Profile
180p
15fps
48
192 - 256
Base Profile
180p
30fps
48
128 - 192
Base Profile
180p
15fps
48
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Conferencing Modes Overview
Mixed CP and SVC Conferencing
In a mixed CP (AVC) and SVC conference, AVC-based endpoints and SVC-enabled endpoints can be
supported in the same conference.
In a mixed CP (AVC) and SVC conference, SVC endpoints transmit multiple resolutions and temporal layers
to the RealPresence Collaboration Server like the SVC-based conferences, while AVC endpoints, for
example, send only one AVC video stream to the Collaboration Server. AVC endpoints can send different
video protocols, such as H.263, and H.264. The Collaboration Server relays SVC-encoded video bit streams
to the SVC-enabled endpoints in the conference according to their request. This enables the video
conference layouts to be automatically assembled by the endpoint. AVC endpoints connected to the
conference send a single AVC video bit stream to the Collaboration Server, which is then transcoded to SVC
video streams. SVC-enabled endpoints receive the AVC converted video bit streams through the
Collaboration Server from the AVC endpoints as a single SVC video bit stream. Alternatively, AVC endpoints
receive a single video bit stream with the defined video conference layout from the Collaboration Server.
The following diagram illustrates an example of a mixed CP and SVC conferencing mode:
In this example, an SVC endpoint (1) receives three video streams at different frame rates and resolutions,
and creates the conference layout with the received video streams. The video bit stream that the SVC
endpoint receives from the AVC endpoint (3) is decoded in the Collaboration Server and then encoded into
an SVC bit stream in the required resolution.
Alternatively, an AVC endpoint (4) sends a single resolution video stream to the Collaboration Server. The
Collaboration Server first decodes the SVC bit streams and AVC bit streams, then the Collaboration Server
composes the video layout for the AVC endpoint and sends a single resolution video stream with the video
layout to the participant. In the displayed example, the Collaboration Server creates different video layouts
for each AVC endpoint.
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Conferencing Modes Overview
Note: DMA configuration
When a DMA system is part of the solution, both dial-in and dial-out, should originate from the DMA.
SVC endpoints receive a participant count from the DMA in order to decide on layout and how many
media streams they wish to receive. If participants are dialed out from the conference, the call flow is
not routed through the DMA resulting in the participant count and layout information not being
received.
MCU Resource Capacities for Mixed CP and SVC Conferences
In a mixed CP and SVC conference, video resources are allocated according to the MCU type and the
translation pools (AVC to SVC and SVC to AVC) used to convert video streams. Translation pools are
dynamically allocated, when the conference becomes a mixed CP and SVC conference; resources are not
released when the conference stops being a mixed CP and SVC conference. The translation pools send
one SVC to AVC stream with a resolution of 360p, two AVC to SVC streams with a resolution of 360p and
180p for AVC HD endpoints, and one video stream with a resolution of 180p for AVC SD endpoints. When
a video stream with a resolution of 360p is not available, a video stream with a resolution of 180p is sent
instead.
Translations between different endpoints can be done without using the highest resolution, thus saving
translation resources. CP video layouts in mixed CP and SVC conferences support the standard resolutions
as in normal CP conferences.
Taking these factors into consideration and the type of MCU deployed in the environment, the resource
capacities for a mixed CP and SVC conference can vary.
Note: Resource Units
System Resources are now reported in terms of HD720p30 CP ports. One HD video port equals 3
CIF video ports.
For information on resource capacity allocations for the RealPresence Collaboration Server see Resource
Capacities for Full Capacity Collaboration Server per Resolution in Mixed CP and SVC Conferencing.
In a mixed CP and SVC conference, video resources are used according to the amount of both AVC and
SVC participants in the conference and according to the actual type of the conference - mixed CP and SVC
conferences or CP only conferences. The ratio of resources in a mixed conference is one AVC HD (720p30)
video resource to three SVC video resources, meaning for each AVC HD video resource, three SVC video
resources can be allocated.
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Using Conference Profiles
Conference Profiles include conference parameters such as Conferencing Mode, conference line rate,
video and content sharing resolutions and settings, video layout, encryption, Lost Packet Recovery (LPR),
etc. Profiles eliminate the need to define these parameters for each new conference created on the MCU.
They are stored on the Collaboration Server and they enable you to define all types of conferences.
Conference Profiles are assigned to Conferences, Meeting Rooms, Reservations and Entry Queues. The
same Profile can be assigned to different conferencing entities. When modifying the Profile parameters, the
changes will be applied to all the conferencing entities to which the profile is assigned.
Conference Profile options differ according to the selected Conferencing Mode. Profiles can be defined for
AVC (Advanced Video Codec) CP and VSW, SVC (Scalable Video Codec), or Mixed CP and SVC
conferencing modes.
Conference Profiles can be saved to Conference Templates along with all participant parameters, including
their Personal Layout and Video Forcing settings. It enables administrators and operators to create, save,
schedule and activate identical conferences quickly and easily.
Note: RealPresence® Distributed Media Application™ (DMA®)
In the Polycom® RealPresence® Web Suite solution (Collaboration Servers 1800/2000/4000) and
CloudAxis® Suite (Collaboration Server Virtual Edition), conference Profiles are defined in the
RealPresence Distributed Media Application (DMA) system component and should not be defined
directly in the MCU component.
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
Note: Using Profiles with RealPresence DMA
In the Polycom® RealPresence® Web Suite solution, the Conference Profiles are defined in the
RealPresence DMA system component and should not be defined directly in the RealPresence
Collaboration Server (RMX) 1800/2000/4000 component.
Conferencing Parameters Defined in a Profile
When defining a new video Profile, you select the parameters that determine the video display on the
participant’s endpoint and the quality of the video, according to the selected Conferencing Mode. When
defining a new conference Profile, the system uses default values for the selected conferencing Mode.
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Using Conference Profiles
Conferencing Capabilities in the Various Conferencing Modes
The following table summarizes the conferencing capabilities and options available in the different
Conferencing Modes.
Conferencing Capabilities in the Different Conferencing Modes
Feature
CP Only
Mixed CP & SVC
SVC Only
VSW
Reservations




Operator Conferences

*

*

*

*












Conference Type
Entry Queues
Permanent Conference
Cascading ‡
Conferencing Feature
IVR
Reduced IVR set
for SVC endpoints
Dial Out




Auto Redial

LPR


***

***


Content




Presentation Mode




Lecture Mode




Same Layout



Layout Selection


AVC endpoints only
Layout set to Auto
Layout and defined
on the endpoint







Skins

AVC endpoints only
Encryption
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Using Conference Profiles
Conferencing Capabilities in the Different Conferencing Modes
Feature
CP Only
Recording

Mixed CP & SVC
SVC Only
VSW



AVC recording only
Site Names
Message Overlay



AVC endpoints only
Managed by the
endpoint (not via
MCU)




AVC endpoints only
* Entry Queue & Destination Conference must have the same profile (i.e. SVC only to SVC only, Mixed CP and SVC
to Mixed CP and SVC)
*** For AVC, the LPR error resiliency is used, however for SVC endpoints, new error resiliency methods are used.
‡ Cascading is not supported on Collaboration Server (RMX) 1800 with no DSP cards.
Default Profile Settings in CP Conferencing Mode
The Collaboration Server is shipped with a default Conference Profile for CP conferences which allows
users to immediately start standard ongoing CP conferences. These are also the default settings when
creating a new Profile. The default settings are as follows:
Default CP Only Conference Profile Settings
Setting
Value
Profile Name
Factory_Video_Profile
Line Rate
384Kbps
Video Switching
Disabled
Operator Conference
Disabled
Encryption
Disabled
Packet Loss Compensation (LPR
and DBA)
Enabled for CP Conferences
Auto Terminate
•
•
Auto Redialing
Disabled
Exclusive Content Mode
Disabled
TIP Compatibility
Disabled
Enable FECC
Enabled
Enabled Gathering Phase
Enabled
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When last participant remains - Disabled
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Using Conference Profiles
Default CP Only Conference Profile Settings
Setting
Value
Display Language
English
Video Quality
Sharpness
Maximum Resolution
Auto
Video Clarity *
Enabled
Auto Brightness *
Enabled
Content Settings
HiResGraphics (High Res Graphics)
Content Protocol
H.264 HD
Send Content to legacy endpoints
Enabled
Presentation Mode
Disabled
Same Layout
Disabled
Lecturer View Switching
Disabled
Telepresence Mode *
Auto
Telepresence Layout Mode
Continuous Presence
Auto Scan Interval
Disabled (10)
Auto Layout
Enabled
Echo Suppression
Enabled
Keyboard Noise Suppression
Disabled
Audio Clarity
Enabled
Mute participants except the lecturer
Disabled
Skin
Polycom
IVR Name
Conference IVR Service
Recording
Disabled
Site Names display
Disabled
Message Overlay
Disabled
Network Services - SIP Registration
Disabled
Network Services - Accept Calls
Enabled
* Video Clarity, Auto Brightness, and Telepresence Mode are not support by Collaboration Server (RMX)
1800 with no DSP cards.
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Using Conference Profiles
This Profile is automatically assigned to the following conferencing entities:
Name
ID
Meeting Rooms
Maple_Room
1001
Oak_Room
1002
Juniper_Room
1003
Fig_Room
1004
Entry Queue
Default EQ
1000
Default Profile Settings in SVC Only Conferencing Mode
The Collaboration Server is shipped with a default Conference Profile for SVC Only conferences which
allows users to immediately start standard ongoing SVC Only conferences. These are also the default
settings when creating a new Profile. The default settings are as follows:
Default SVC Only Conference Profile Settings
Setting
Value
Profile Name
Factory_SVC_Video_Profile
Line Rate
1920Kbps
Video Switching
Disabled
Operator Conference
Not supported
Encryption
Disabled
Packet Loss Compensation (LPR and
DBA)
Not supported
Auto Terminate
•
•
Auto Redialing
Not supported
Exclusive Content Mode
Disabled
TIP Compatibility
Disabled
Enable FECC
Disabled
Enabled Gathering Phase
Enabled
Display Language
English
Video Quality
Sharpness
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When last participant remains - Disabled
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Using Conference Profiles
Default SVC Only Conference Profile Settings
Setting
Value
Maximum Resolution
Auto
Video Clarity *
Enabled
Auto Brightness *
Enabled
Content Settings
Graphics
Content Protocol
H.264 Cascading and SVC Optimized
Presentation Mode
Not applicable
Send Content to legacy endpoints
Disabled
Same Layout
Not applicable
Lecturer View Switching
Not applicable
Telepresence Mode *
Auto
Telepresence Layout Mode
Continuous Presence
Auto Scan Interval
Not applicable
Auto Layout
Enabled (Only available option)
Echo Suppression
Enabled
Keyboard Noise Suppression
Disabled
Audio Clarity
Enabled
Mute participants except the lecturer
Not applicable
IVR Name
Conference IVR Service
Message Overlay
Disabled
Network Services - SIP Registration
Disabled
Network Services - Accept Calls
Enabled
* Video Clarity, Auto Brightness, and Telepresence Mode are not support by Collaboration Server (RMX)
1800 with no DSP cards.
Default Profile Settings in a Mixed CP and SVC Conferencing Mode
The Collaboration Server is shipped with a default Conference Profile (CP and SVC) for mixed CP and SVC
conferences which enables users to immediately start a standard ongoing mixed CP and SVC conference.
These are also the default settings when creating a new Profile. (During mixed SVC & CP conferences,
PSTN (Audio Only) calls are supported.) Dial-out is not available in Mixed CP and SVC conferences.
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Using Conference Profiles
The default settings are as follows:
Default Mixed CP and SVC Conference Profile Settings
Setting
Value
Profile Name
Factory_Mix_SVC_CP_Video_Profile
Line Rate
1920Kbps
Video Switching
Disabled
Operator Conference
Disabled
Encryption
Enabled
Packet Loss Compensation (LPR
and DBA)
Enabled for AVC participants only
Auto Terminate
•
•
Auto Redialing
Disabled
Font for text over video
Enabled for AVC participants only
Exclusive Content Mode
Disabled
TIP Compatibility
Disabled
Enable FECC
Enabled
Enabled Gathering Phase
Enabled
Display Language
English
Video Quality
Sharpness
Maximum Resolution
Auto
Video Clarity *
Disabled
Auto Brightness *
Disabled
Content Settings
Graphics
Content Protocol
H.264 Cascade and SVC Optimized (only)
Presentation Mode
Disabled
Send Content to legacy endpoints
Disabled
Same Layout
Enabled
Lecturer View Switching
Disabled
Telepresence Mode *
Off
Telepresence Layout Mode
Continuous Presence
Auto Scan Interval
Disabled
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After last participant quits - Enabled
When last participant remains - Disabled
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Using Conference Profiles
Default Mixed CP and SVC Conference Profile Settings
Setting
Value
Auto Layout
Enabled
Echo Suppression
Enabled for AVC participants only
Keyboard Noise Suppression
Enabled for AVC participants only
Audio Clarity
Enabled for AVC participants only
Mute participants except the lecturer
Disabled
Skin
Classic (for AVC participants)
IVR Name
Conference IVR Service
Recording
Enabled
Site Names display
Enabled for AVC participants only
Message Overlay
Disabled
Network Services - SIP Registration
Disabled
Network Services - Accept Calls
Enabled
Network quality indication
Enabled for AVC participants only
* Video Clarity, Auto Brightness, and Telepresence Mode are not support by Collaboration Server (RMX)
1800 with no DSP cards.
This Profile is automatically assigned to the following conferencing entities:
Name
ID
Meeting Rooms
Maple_Room
1001
Oak_Room
1002
Juniper_Room
1003
Fig_Room
1004
Entry Queue
Default EQ
1000
Viewing the List of Conference Profiles
Existing Conference Profiles are listed in the Conference Profiles list pane.
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Using Conference Profiles
To list Conference Profiles:
1 In the RMX Management pane, expand the Rarely Used list.
2 In the RMX Management pane, click the Conference Profiles button.
The Conference Profiles are displayed in the Conference Profiles list pane.
The number of the currently defined conference profiles appears in the title of the list pane.
The following Conference Profile properties are displayed in the List pane:
Conference Profiles Pane Columns
Field
Description
Name
The name of the Conference Profile.
Layout
Displays either Auto Layout or an icon of the layout selected for the profile.
Line Rate
The maximum bit rate in kbps at which endpoints can connect to the conference.
Routing Name
Displays the Routing Name defined by the user or automatically generated by the
system.
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Using Conference Profiles
Conference Profiles Pane Columns
Field
Description
Encryption
Displays if media encryption is enabled for the Profile. For more information see
Packet Loss Compensation - LPR and DBA.
Profiles Toolbar
The Profile toolbar provides quick access to the Profile functions:
Profile Toolbar buttons
Button
Button Name
Description
New Profile
To create a new Profile.
Delete Profile
To delete a Profile, click the Profile name and then click this
button.
Import Profile
To import Conference Profiles from another MCU in your
environment.
Export Profile
To export Conference Profiles to a single XML file that can be
used to import the Conference Profiles on multiple MCUs.
Modifying an Existing Profile
You can modify any of the Profile’s parameters but you cannot rename the Profile.
To modify the Profile properties:
1 In the Conference Profiles List, double-click the Profile icon, or right-click the Profile icon, and
select Profile Properties.
The Profile Properties - General dialog box opens.
2 Modify the required Profile parameter(s).
3 Click OK.
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Using Conference Profiles
Deleting a Conference Profile
You can delete profiles from the Profiles list.
Note: Deleting Profiles
A Conference Profile cannot be deleted if it is being used by Meeting Rooms, Reservations, Entry
Queues, and SIP Factories. A Profile that is assigned to only one ongoing conference and no other
conferencing entity can be deleted.
To delete a Conference Profile:
1 List the profiles that are currently defined in the system. For details, see Viewing the List of
Conference Profiles.
2 In the Conference Profiles list, select the Conference Profile you want to delete.
3 Click the Delete Profile (
) button.
or
Right-click the Conference Profile you wish to delete, and select Delete Profile from the menu.
4 In the confirmation dialog box, click OK.
The Conference Profile is deleted.
Defining New Profiles
Note: RealPresence Distributed Media Application (DMA)
In the Polycom® RealPresence® Web Suite solution, the Conference Profiles are defined in the
RealPresence DMA system component and should not be defined directly in the RealPresence
Collaboration Server (RMX) 1800/2000/4000 component.
Profiles are the basis for the definition of all ongoing conferences, Reservations, Meeting Rooms, Entry
Queues, and Conference Templates and they contain only conference properties.
Profiles can be defined for the following Conferencing Modes: AVC (Advanced Video Codec) CP and VSW,
SVC (Scalable Video Codec) or Mixed CP and SVC. The Profile tabs and options change according to the
selected Conferencing Mode and only supported options are available for selection. Unsupported options
are disabled (grayed out).
CP Conferencing Mode also offers a special functional conference - Operator Conference.
To facilitate the definition process of a new Profile, the system displays default values for each parameter
so you need only to modify the required settings.
To define a new Profile:
1 In the RMX Management pane, expand the Rarely Used list.
2 In the RMX Management pane, click Conference Profiles.
3 In the Conference Profiles pane, click the New Profile button.
The New Profile – General dialog box opens.
4 In the Display Name field, enter the profile name.
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Using Conference Profiles
5 Select the appropriate Conferencing Mode: CP, VSW, SVC Only or CP and SVC.
The New Profile tabs and options change according to the selected Conferencing Mode and only
supported options are available for selection.
6 Define the profile parameters as described in:
 Defining AVC CP Conferencing Profiles
 Defining an AVC Video Switching Conference Profile
 Defining SVC Conference Profiles
 Defining Mixed CP and SVC Conferencing Profiles
Exporting and Importing Conference Profiles
Conference Profiles can be exported from one MCU and imported to multiple MCUs in your environment,
enabling you to copy the Conference Profiles definitions to other systems. This can save configuration time
and ensures that identical settings are used for conferences running on different MCUs. This is especially
important in environments using cascading conferences that are running on different MCUs.
Guidelines for Exporting and Importing Conference Profiles
● Only Collaboration Server system administrators can export and import Conference Profiles.
Operators are only allowed to export Conference Profiles.
● You can select a single, multiple, or all Conference Profiles to be exported.
● Conference Templates and their related Conference Profiles can be exported and imported
simultaneously using the Conference Templates export and import function. For more information,
see Exporting and Importing Conference Templates.
Exporting Conference Profiles
Conference Profiles are exported to a single XML file that can be used to import the Conference Profiles on
multiple MCUs. Using the Export Conference Profile feature, you can export all or selected Conference
Profiles from an MCU.
Exporting All Conference Profiles from an MCU
To export all Conference Profiles from an MCU:
1 List the profiles that are currently defined in the system. For details, see Viewing the List of
Conference Profiles.
2 In the Conference Profiles List toolbar, click the Export Conference Profiles
button, or
right-click anywhere in the Conference Profiles pane, and then click Export Conference Profiles.
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Using Conference Profiles
The Conference Profile - Export dialog box is displayed.
3 In the Export Path field, click Browse to navigate to the location of the desired path where you want
to save the exported file.
4 In the Profiles file name field, type the file name prefix. The file name suffix (_confProfiles.xml) is
predefined by the system. For example, if you type Profiles01, the exported file name is defined
as Profiles01_confProfiles.xml.
5 Click OK to export the Conference Profiles to a file.
If the export file with the same file name already exists, a prompt is displayed.
6 Click Yes to replace the exported file, or click No to cancel the export operation and return to the
Conference Profiles list. You can modify the export file name and restart the export operation.
Exporting Selected Conference Profiles
You can select a single Conference Profile or multiple Conference Profiles and export them to a file to be
imported to other MCUs in your environment.
To export selected Conference Profiles:
1 List the Profile that are currently defined in the system. For details, see Viewing the List of
Conference Profiles.
2 In the Conference Profiles pane, select the profiles you want to export.
3 In the Conference Profiles List toolbar, click the Export Conference Profiles
button, or
right-click the selected Conference Profiles, and then click Export Selected Conference Profiles.
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Using Conference Profiles
The Conference Profile - Export dialog box is displayed.
4 In the Export Path field, click Browse to navigate to the location of the desired path where you want
to save the exported file.
5 In the Profiles file name field, type the file name prefix. The file name suffix (_confProfiles.xml) is
predefined by the system. For example, if you type Profiles01, the exported file name is defined
as Profiles01_confProfiles.xml.
6 Click OK to export the Conference Profiles to a file.
If the export file with the same file name already exists, a prompt is displayed.
7 Click Yes to replace the exported file, or click No to cancel the export operation and return to the
Conference Profiles list. You can modify the export file name, and restart the export operation.
Importing Conference Profiles
If your environment includes two or more MCUs, import previously exported Conference Profiles to your
MCU to save configuration time and ensure that all MCUs use the same conferencing parameters.
Note: Imported Profile
Conference Profiles are not imported when a Conference Profile with that name already exists or if
an IVR Service which is assigned to any of the imported Profile does not exist in the MCU.
Conference Profiles are not imported when a Conference Profile with that name already exists or if an IVR
Service which is assigned to any of the imported profiles does not exist in the MCU.
To import Conference Profiles:
1 Display the Conference Profiles List. For details, see Viewing the List of Conference Profiles.
2 In the Conference Profiles List toolbar, click the Import Conference Profiles
button, or
right-click the Conference Profiles pane, and then click Import Conference Profiles.
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Using Conference Profiles
The Conference Profile - Import dialog box is displayed.
3 In the Import Path field, click Browse to navigate to the path and file name of the exported
Conference Profiles you want to import.
4 Click OK to import the Conference Profiles.
When Conference Profiles cannot be imported, a Message Alert window is displayed with the
profiles that were not imported.
Conference Profiles that are not problematic are imported.
5 Click Cancel to exit the Message Alerts window.
The imported Conference Profiles appear in the Conference Profiles list.
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Defining AVC-Based Conference Profiles
AVC-based Conference Profile options differ according to the selected Conferencing Mode CP and VSW.
To facilitate the definition process of a new Profile, the system displays default values for each parameter
so you need only to modify the required settings.
Note: Conference Profiles Definition Location
In RealPresence® Platform, Conference Profiles should be defined in Polycom® RealPresence®
DMA® instead of in RealPresence Collaboration Server (RMX) 1500/1800/2000/4000.
Defining AVC CP Conferencing Profiles
When defining a new Profile, you select the parameters that determine the video display on the participant’s
endpoint, the quality of the video, content sharing parameters, whether the conference will be recorded,
encryption, Telepresence mode and other conferencing parameters.
The following parameters are defined:
● New AVC CP Profile - General Parameters
● New AVC CP Profile - Advanced Parameters
● New AVC CP Profile - Gathering Settings Parameters
● New AVC CP Profile - Video Quality Parameters
● New AVC CP Profile - Video Settings Parameters
● New AVC CP Profile - Audio Settings Parameters
● New AVC CP Profile - IVR Parameters
● New AVC CP Profile - Recording Parameters
● New AVC CP Profile - Site Names Parameters
● New AVC CP Profile - Message Overlay Parameters
● New AVC CP Profile - Network Services Parameters
● New AVC CP Profile - Layout Indications Parameters
To define a new CP Profile:
1 In the RMX Management pane, click Conference Profiles.
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Defining AVC-Based Conference Profiles
2 In the Conference Profiles pane, click the New Profile button.
The New Profile – General dialog box opens.
3 Define the Profile name and, if required, the Profile - General parameters:
New AVC CP Profile - General Parameters
Field/Option
Description
Display Name
Enter a unique Profile name, as follows:
• English text uses ASCII encoding and can contain the most characters (length
varies according to the field).
• European and Latin text length is approximately half the length of the maximum.
• Asian text length is approximately one third of the length of the maximum.
It is recommended to use a name that indicates the Profile type, such as CP or
Operator conference.
Notes:
• This is the only parameter that must be defined when creating a new profile.
• This field is displayed in all tabs.
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Defining AVC-Based Conference Profiles
New AVC CP Profile - General Parameters
Field/Option
Description
Line Rate
Select the conference bit rate. The line rate represents the combined video, audio
and Content rate.
The default setting is 384 Kbps.
Notes:
• This field is displayed in all tabs.
• The maximum line rate at which ISDN endpoints can connect to a conference is
768 kbps.
Downspeeding is not supported. As a result, ISDN calls will consume
bandwidth resources according to the Line Rate specified in the conference
Profile. For example, if the conference Line Rate is 512kbps, ISDN calls
connecting at lower line rates (256kbps) will still consume the bandwidth
resources of 512kbps calls. As a result, if bandwidth resources are fully
consumed, ISDN calls may be rejected before media card resources are
exhausted.
• The Collaboration Server supports AVC-CP conferences at line rates of up to
6Mbps:
 For Dial-in calls only.
 For use with HD1080p30/60 and HD720p60 resolutions.
To prevent excessive bandwidth usage and resource consumption, the system
will ignore a Line Rate of 6Mbps, substituting line rates appropriate to the
Maximum Resolution defined.
 Up to 1Mbps for SD30, CIF60 and below
 Up to 4Mbps for SD60, HD720p30
 Up to 6Mbps for HD720p60, HD1080p30, HD1080p60
(RMX 1800 with no DSP cards maximum AVC-CP conferencing line rate:
4Mbps.)
• If ‘Maximum Resolution’ is specified in ‘Video Quality Parameters’, the line rate
might be modified automatically according to table ‘Maximum negotiated line
rate for resolutions’.
• If ‘Maximum Resolution’ is specified in ‘Video Quality Parameters’, the line rate
might be modified automatically according to the Maximum Negotiated Line Rate
for Resolutions table.
Conferencing Mode
Select the required Conferencing Mode. The selection affects the available tabs
and their fields.
Notes:
• This field is displayed in all tabs.
• If the selected Line Rate is higher than 4Mbps, and Conferencing Mode is
modified to SVC Only or CP and SVC, the line rate is dropped to match
SVC-enabled conferencing rates.
Routing Name
Enter the Profile name using ASCII characters set.
The Routing Name can be defined by the user or automatically generated by the
system if no Routing Name is entered as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered in
Display Name, the ID (such as Conference ID) is used as the Routing Name.
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New AVC CP Profile - General Parameters
Field/Option
Description
Video Switching
Disabled. Applicable for VSW conferencing. For more information, see Defining an
AVC Video Switching Conference Profile.
H.264 High Profile
Disabled. Applicable for VSW conferencing. For more information, see H.264 High
Profile Support in Video Switching Conferences.
Operator Conference
(CP Only)
Select this option to define the profile of an Operator conference.
An Operator conference can only be a Continuous Presence conference, therefore
when selected, the Video Switching option is disabled and cleared.
When defining an Operator Conference, the Send Content to Legacy Endpoints
option in the Video Quality tab is cleared and disabled.
For more information, see Operator Assistance & Participant Move.
4 Click the Advanced tab.
The New Profile – Advanced dialog box opens.
5 Define the following parameters:
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New AVC CP Profile - Advanced Parameters
Field/Option
Description
Encryption
Select the Encryption option for the conference:
• Encrypt All - Encryption is enabled for the conference and all conference
participants must be encrypted.
• No Encryption - Encryption is disabled for the conference.
• Encrypt when Possible - Enables the negotiation between the MCU and the
endpoints and let the MCU connect the participants according to their
capabilities, where encryption is the preferred setting. For connection
guidelines see Mixing Encrypted and Non-encrypted Endpoints in one
Conference.
For more information, see Packet Loss Compensation - LPR and DBA.
LPR
When selected (default for CP conferences), Lost Packet Recovery creates
additional packets that contain recovery information used to reconstruct packets
that are lost during transmission.
The LPR check box is automatically cleared if Video Switching is selected as
the Conferencing Mode, but can be selected if required.
For more information, see Packet Loss Compensation - LPR and DBA.
Auto Terminate
When selected (default), the conference automatically ends when the
termination conditions are met:
• Before First Joins — No participant has connected to a conference during
the n minutes after it started. Default idle time is 10 minutes.
• At the End - After Last Quits — All the participants disconnected from the
conference, and the conference is idle (empty) for the predefined time period.
Default idle time is 1 minute.
• At the End - When Last Participant Remains — Only one participant is
connected to the conference for the predefined time period (excluding the
recording link which is not considered a participant when this option is
selected). This option should be selected when defining a Profile that will be
used for Gateway Calls, and wishing to ensure the call is automatically
terminated when only one participant is connected. Default idle time is 1
minute.
Note: The selection of this option is automatically cleared and disabled when
the Operator Conference option is selected. The Operator conference cannot
automatically end unless it is terminated by the Collaboration Server User.
Auto Redialing
The Auto Redialing option instructs the Collaboration Server to automatically
redial H.323 and SIP participants that have been abnormally disconnected from
the conference.
• Auto Redialing is disabled by default.
• Auto Redialing can be enabled or disabled during an ongoing conference
using the Conference Properties – Advanced dialog box.
• The Collaboration Server will not redial an endpoint that has been
disconnected from the conference by the participant.
• The Collaboration Server will not redial an endpoint that has been
disconnected or deleted from the conference by an operator or administrator.
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New AVC CP Profile - Advanced Parameters
Field/Option
Description
Exclusive Content Mode
Select this option to limit the Content broadcasting to one participant, preventing
other participants from interrupting the Content broadcasting while it is active.
TIP Compatibility
Select the TIP Compatibility mode when implementing a Collaboration Server
and Cisco Telepresence Systems (CTS) Integration solution. The TIP
Compatibility mode affects the user video and content experience.
The following TIP Compatibility modes are available:
• None
• Prefer TIP
From Version 8.1.1, Polycom endpoints can also connect to Entry Queues,
Meeting Rooms and conferences using the TIP protocol.
The connection of the Polycom endpoints with TIP protocol to a TIP Compatible
Entry Queues, Meeting Rooms and conferences using the TIP protocol is
enabled when the Polycom endpoints are registered to the CUCM and the
Prefer TIP option is selected in the conference Profile.
When the Prefer TIP option is selected, conferencing entities can include
endpoints connected using all protocols, including TIP and SIP protocols.
When Prefer TIP is selected, Gathering Settings, content settings, Message
Overlay, Site Names and Network Indication(s) cannot be enabled.
Note: If an option other than None is selected in this field, the Gathering
Settings options are disabled.
For more information, see .
Enable FECC
This option is enabled by default, allowing participants in the conference to
control the zoom and PAN of other endpoints in the conference via the FECC
channel. Clear this check box to disable this option for all conference
participants.
FW NAT Keep Alive
The MCU can be configured to send a FW NAT Keep Alive message at specific
Intervals for the RTP, UDP and BFCP channels.
For more information see FW (Firewall) NAT Keep Alive.
Interval
If needed, modify the NAT Keep Alive Interval field within the range of 1 - 86400
seconds. For more information see FW (Firewall) NAT Keep Alive.
MS AV MCU Cascade
Mode
This enables you to set Cascade Mode as either Resource Optimized or Video
Optimized.
• Resource Optimized
System resource usage is optimized by allowing high resolution connections
only at high line rates and may result in lower video resolutions for some line
rates. This option allows you to save MCU resources and increase the
number of participant connections.
• Video Quality Optimized
Video is optimized through higher resolution connections at lower line rates
increasing the resource usage at lower line rates. This may decrease the
number of participant connections.
For more information, see H.264 Base Profile and High Profile Comparison.
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6 For CP Conferences only: Click the Gathering Settings tab.
7 Optional. Define the following fields if the conference is not launched by the Polycom Conferencing
Add-in for Microsoft Outlook:
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Note: Polycom Conferencing Add-in for Microsoft Outlook
If the conference is launched by the Polycom Conferencing Add-in for Microsoft Outlook, the field
information is received from the meeting invitation and existing field value are overridden. For more
information see Polycom Conferencing for Microsoft Outlook®.
• Gathering is disabled in gateway calls.
• Gathering is not supported in Cascading Conferences.
• Cascading is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
New AVC CP Profile - Gathering Settings Parameters
Field
Description
Display Name
This field is defined when the Profile is created. For more information see the
Defining New Profiles.
Enable Gathering
Select this check box to enable the Gathering Phase feature.
Default: Selected.
Displayed Language
Select the Gathering Phase slide language:
Gathering Phase slide field headings are displayed in the language selected.
The Gathering Phase slide can be in a different language to the Collaboration
Server Web Client.
Default: English
Note: When working with the Polycom Conferencing Add-in for Microsoft Outlook,
the language selected should match the language selected for the conference in the
Polycom Conferencing Add-in for Microsoft Outlook to ensure that the Gathering
Phase slide displays correctly.
Dial-in Number 1
Enter the ISDN or PSTN number(s) to call to connect to the conference.
Note: The numbers entered must be verified as the actual Access Numbers.
Dial-in Number 2
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New AVC CP Profile - Gathering Settings Parameters
Field
Description
Info 1
Optionally, enter any additional information to be displayed during the Gathering
Phase.
These fields are not limited in the Collaboration Server Web Client but only 96
characters can be displayed in the Gathering Slide on a 16:9 monitor.
If the Gathering slide is displayed on a 4:3 endpoint: the slide is cropped on both
sides:
• The left most characters of the information fields are not displayed.
• The live video is cropped on the right side of the display.
Info 2
Info 3
For more information see Auto Scan and Customized Polling in Video Layout.
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8 Click the Video Quality tab.
The New Profile – Video Quality dialog box opens.
9 Define the following parameters:
New AVC CP Profile - Video Quality Parameters
Field/Option
Description
People Video Definition
Video Quality
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Sharpness is the only supported content format that supports higher video
resolutions.
Depending on the amount of movement contained in the conference video, select
either:
• Motion – For a higher frame rate without increased resolution. When selected,
Video Clarity is disabled.
• Sharpness – For higher video resolution and requires more system resources.
Note: When Sharpness is selected as the Video Quality setting in the conference
Profile, the Collaboration Server will send 4CIF (H.263) at 15fps instead of CIF
(H.264) at 30fps.
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New AVC CP Profile - Video Quality Parameters
Field/Option
Description
Maximum Resolution
This setting overrides the Maximum Resolution setting of the Resolution
Configuration dialog box.
The administrator can select one of the following Maximum Resolution options:
• Auto (default) - The Maximum Resolution remains as selected in the
Resolution Configuration dialog box.
• CIF
• SD
• HD720
• HD1080
Maximum Resolution settings can be monitored in the Profile Properties - Video
Quality and Participant Properties - Advanced dialog boxes.
Notes:
• The Resolution field in the New Participant - Advanced dialog box allows
Maximum Resolution to be further limited per participant endpoint.
• The Maximum Resolution settings for conferences and participants cannot be
changed during an ongoing conference.
Video Clarity™
When enabled (default), video enhancing algorithms is applied to incoming video
streams of resolutions up to and including SD. Clearer images with sharper edges
and higher contrast are sent back to all endpoints at the highest possible resolution
supported by each endpoint.
All layouts, including 1x1, are supported.
Notes:
Video Clarity is enabled only when Video Quality is set to Sharpness (default
setting) and is disabled when Video Quality is set to Motion.
Video Clarity is not supported by Collaboration Server (RMX) 1800 with no DSP
cards.
Auto Brightness
Auto Brightness detects and automatically adjusts the brightness of video windows
that are dimmer than other video windows in the conference layout.
• Auto Brightness only increases brightness and does not darken video windows.
• Auto Brightness is selected by default.
• Auto Brightness cannot be selected and deselected during an ongoing
conference.
• When Auto Brightness is enabled, color changes may be observed in
computer-based VGA Content sent by HDX endpoints through the People video
channel.
Default: On
Auto Brightness is not supported by Collaboration Server (RMX) 1800 with no
DSP cards.
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New AVC CP Profile - Video Quality Parameters
Field/Option
Description
Content Video Definition
Content Settings
Select the transmission mode for the Content channel:
• Graphics — Basic mode, intended for normal graphics
• Hi-res Graphics — Higher bit rate intended for high resolution graphic display
• Live Video — Content channel displays live video
• Customized Content Rate — Manual definition of the Conference Content
Rate, mainly for cascading conferences.
Selection of a higher bit rate for the Content results in a lower bit rate for the people
channel.
For a detailed description of each of these options, see Content Sharing
Parameters in Content Highest Common (Content Video Switching) Mode.
AS SIP Content
AS-SIP is an implementation of SIP that utilizes SIP’s built in security features.
When selected, content is shared using the Multiple Resolutions mode and is not
supported in any other Content sharing mode.
For more information, see Enabling AS-SIP Content.
Multiple Content
Resolutions
Click this check box to enable the Multiple Content Resolutions mode, in which
content is shared in multiple streams, one for each video protocol: H.263 and
H.264. This allows endpoints with different protocols to connect and disconnect
without having to restart Content sharing in the middle of a conference. For more
information, see Multiple Content Resolutions.
When enabled, the H.264 is always selected and can not be deselected.
Note: If Multiple Content Resolutions is selected in a Cascading environment, the
Content Protocol must be set to H.264 Cascade Optimized, and the Transcode
to field must be set (checked) to H.264 Cascade.
Optional. Select additional protocols:
• H.263 - if the conference will include H.263-capable endpoints that do not support
H.264 protocol for content sharing.
• H.264 Cascade - if the conference will include cascading links that should use a
fixed video format for content sharing.
Optional. If H.264 Cascade is selected, select the desired Content Resolution.
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New AVC CP Profile - Video Quality Parameters
Field/Option
Description
Content Protocol
Select the Content Protocol to be used for content sharing in Highest Common
Content Sharing Mode.
• H.263
Content is shared using the H.263 protocol.
Use this option when most of the endpoints support H.263 and some endpoints
support H.264.
• H.263 & H.264 Auto Selection
When selected, content is shared using H.263 if a mix of H.263-supporting and
H.264-supporting endpoints are connected, or H.264 if all connected endpoints
have H.264 capability.
• H.264 Cascade Optimized
All Content is shared using the H.264 content protocol and is optimized for use
in cascaded conferences.
• H.264 HD (default)
Ensures high quality Content when most endpoints support H.264 and HD
resolutions.
Note: When Multiple Content Resolutions is selected, the Content Protocol field
is hidden.
For more information, see Content Protocols and Defining Content Sharing
Parameters for a Conference
H.264 High Profile
The H.264 High Profile check box is un-checked by default and is displayed next to
the Content Protocol drop-down menu if all the following conditions are met:
• The MCU is a RealPresence Collaboration Server 1800, or 2000/4000 containing
MPMRx cards.
• Conferencing mode is VSW (Video Switching).
• Multiple Resolutions (Content Transcoding) is not selected.
• The selected Content Protocol is H.264 Cascade Optimized.
If H.264 HD, H.264 Cascade Optimized is selected, the Content Resolution is
set according to the line rate.
• TIP Compatibility (in the Profile - Advanced dialog box) is selected as None.
Content Resolution
Select the Content Resolution and frame rate according to the selected Content
Sharing Mode (Highest common Content or Multiple Resolution Contents) and the
video protocol. For more information, see Defining Content Sharing Parameters for
a Conference.
Content Rate
The Content Rate drop-down menu is displayed next to the Content Resolution
drop-down menu when:
• H.264 Cascade Optimized is the selected Content Protocol
and
• CustomizedContentRate is the selected Content Setting.
When used with the Collaboration Server 1800 the Content Rate is limited to 4096
kbps.
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New AVC CP Profile - Video Quality Parameters
Field/Option
Description
Send Content to
Legacy Endpoints
(CP / mixed CP and
SVC)
When enabled (default) Content can be sent to H.323/SIP/ISDN endpoints that do
not support H.239 Content (legacy endpoints) over the video (people) channel. For
more information see Sending Content to Legacy Endpoints.
Notes:
• When enabled, an additional HD video resource is allocated to the conference.
• This option is valid when sending Content as a separate stream is enabled by
setting the System Flag ENABLE_H239 to YES.
• Select this option when Avaya IP Softphone will be connecting to the conference.
• In Video Switching Conferencing Mode the Send Content to Legacy Endpoints
option is disabled.
• If the Same Layout option is selected, the Send Content to Legacy Endpoints
selection is cleared and disabled.
• Once an endpoint is categorized as Legacy, it will not be able to restore its
content to the Content channel and will receive content only in the video channel.
H.264 High Profile
Displayed only when the Conferencing Mode is VSW (Video Switching) and H.263
& H.264 Auto Selection or H.264 HD is the selected in Content Protocol, Base
Profile or High Profile will be auto selected based on the endpoint’s capability (up to
HD1080p60).
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10 Click the Video Settings tab.
The New Profile - Video Settings dialog box opens.
In AVC-based CP and Mixed CP and SVC conferences, while the conference is running, you can
change the video layout and select one of the video layouts supported by the MCU.
Video Layout selection can be done in two levels:
 Conference Level – Applies to all conference participants. All participants have the same video
layout.
 Participant Level – The participant’s video layout is changed. The video layout of all other
conference participants is unaffected.
Participant level video layout selection overrides conference level video layout settings. Participants
cannot change their Personal Layouts when Lecturer Mode is enabled.
11 Define the video display mode and layout using the following parameters:
New AVC CP Profile - Video Settings Parameters
Field/Option
Description
Presentation Mode
(CP only)
Select this option to activate the Presentation Mode.
In this mode, when the current speaker speaks for a predefined time (30 seconds),
the conference changes to Lecture Mode. When another participant starts talking,
the Presentation Mode is cancelled and the conference returns to the previous
video layout.
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New AVC CP Profile - Video Settings Parameters
Field/Option
Description
Same Layout
(CP only)
Select this option to force the selected layout on all participants in a conference.
Displays the same video stream to all participants and personal selection of the
video layout is disabled. In addition, if participants are forced to a video layout
window, they can see themselves.
Lecture View
Switching
Select this option to enable automatic switching of participants on the Lecturer’s
screen when Lecture Mode is enabled for the conference.
The automatic switching is enabled when the number of participants exceeds the
number of video windows displayed on the Lecturer’s screen.
Note: Lecture Mode is enabled in the Conference Properties – Participants tab.
For more information, see Lecture Mode.
Telepresence Mode
(CP only)
Select the Telepresence Mode from the drop-down menu:
• Off - Normal conference video is sent by the Collaboration Server.
• Auto (Default) - If any ITP (Immersive Telepresence) endpoints are detected,
ITP features are applied to the conference video for all participants.
When Auto is selected, the ITP features are dynamic. If all ITP endpoints
disconnect from the conference, normal conference video is resumed for all
participants. ITP features are resumed for all participants should an ITP
endpoint re-connects to the conference.
• On - ITP features are applied to the conference video for all participants
regardless of whether there are ITP endpoints connected or not.
Notes:
• This field is enabled only if the Collaboration Server system is licensed for
Telepresence Mode.
• Telepresence Mode is unavailable in Video Switching conferences.
• Telepresence Mode is not supported by Collaboration Server (RMX) 1800 with
no DSP cards.
Telepresence Layout
Mode
(CP only)
The Telepresence Layout Mode drop-down menu enables VNOC operators and
Polycom Multi Layout Applications to retrieve Telepresence Layout Mode
information from the Collaboration Server.
The following modes can be selected:
• Manual
• Continuous Presence (MLA) - Room Continuous Presence (Default)
• Speaker Priority For more information see Speaker Priority in CP Video
Layouts.
• Room Switch - Voice Activated Room Switching. For more information see
Room Switch Telepresence Layouts.
Note: This field is enabled only if the Collaboration Server system is licensed for
Telepresence Mode.
Auto Scan Interval(s)
(CP only)
Select the time interval, 5 - 300 seconds, that Auto Scan uses to cycle the display
of participants that are not in the conference layout in the selected cell.
Auto Scan is often used in conjunction with Customized Polling which allows the
cyclic display to be set to a predefined order for a predefined time period.
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New AVC CP Profile - Video Settings Parameters
Field/Option
Description
Auto Layout
(CP only)
When selected (default), the system automatically selects the conference layout
based on the number of participants currently connected to the conference. When a
new video participant connects or disconnects, the conference layout automatically
changes to reflect the new number of video participants.
For more information, see Auto Layout – Default Layouts in CP Conferences.
Clear this selection to manually select a layout for the conference.
The default Auto Layout settings can be customized by modifying default Auto
Layout system flags in the System Configuration file. For more information see
Auto Layout Configuration Flags.
Note: In some cases, the default layout automatically selected for the conference
contains more cells than the number of connected participants, resulting in an
empty cell. For example, if the number of connected participants is 4, the default
layout is 2x2, but as only 3 participants are displayed in the layout (the participants
do not see themselves), one cell is empty.
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Auto Layout – Default Layouts in CP Conferences
Number of Video Participants
Auto Layout Default Settings
0–2
3
4–5
6–7
8-10
11
12+
In layout 2+8, the two central windows display the last two speakers in the conference: the current
speaker and the “previous” speaker. To minimize the changes in the layout, when a new speaker is
identified the “previous” speaker is replaced by the new speaker while the current speaker remains
in his/her window.
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Note: Support for VUI addition to the H.264 protocol
• The Collaboration Server supports the VUI addition to the H.264 protocol for endpoints that
transmit wide video (16:9) in standard 4SIF resolution.
• When there is a change of speaker in a Continuous Presence conference, the transition is set by
default to fade in the current speaker while fading out the previous speaker.
To make this transition visually pleasant, fading in the current speaker while fading out the
previous speaker is done over a period of 500 milliseconds.
The Fade In/Out feature can be disabled by adding FADE_IN_FADE_OUT as a new flag to the
System Configuration, and setting its value to NO.
For more information about System Flags, see Modifying System Flags.
12 Optional. To select the Video Layout for the conference, deselect the Auto Layout, and click the
required number of windows from the layouts bar and then select the windows array. The selected
layout is displayed in the Video Layout pane.
Video Layout Options
Number of Video Windows
Available Video Layouts
1
2
3
4
5+
9
10+
Overlay
For more information see Overlay Layouts.
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13 Optional. To video force a participant in a window, select the participant’s name from the drop-down
list of conference participants.
When a participant is forced to a layout window, switching between participants is suspended for that
window and only the assigned participant is viewed. Video Forcing works on Conference Level or
Participant Level:
 Conference Level – When forcing a participant to a window, all conference participants will see
that participant in the selected window.
 Participant Level – When forcing a participant to a window, only the participant’s video layout
display is affected. All other participants see the conference layout.
Video Forcing Guidelines:
 A participant cannot appear in two or more windows at the same time.
 Participant level video forcing overrides conference level video forcing.
 In Same Layout mode, participant can view him/herself in a layout window.
 When different size video windows are used in video layouts such as 1+2, 1+3, 1+4, etc., a
participant can only be forced, in Personal Layout, to a video window of the same size as that
selected for him/her in Conference Layout.
 When changing the Video Layout at the conference level, the video forcing settings are not
applied to a new layout, and switching between participants is audio-activated. The video forcing
setting is saved and applied the next time that layout is selected.
 Windows that are not assigned any participant display the current speaker and last speakers, but
appear at random.
 Video Forcing doesn’t change the conference profile.
Video Forcing can be implemented through Media Sources Tab when adding a new Conference or a
new Meeting Room.
For information about preventing the MCU from reassigning cells in a forced layout that were
assigned to endpoints that have disconnected through a system flag named
PRESERVE_PARTY_CELL_ON_FORCE_LAYOUT, see System Configuration Flags.
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14 Click the Audio Settings tab.
The New Profile - Audio Settings dialog box opens.
15 Define the following parameters:
New AVC CP Profile - Audio Settings Parameters
Field/Option
Description
Audio Clarity
When selected, improves received audio from participants connected via low audio
bandwidth connections, by stretching the fidelity of the narrowband telephone
connection to improve call clarity.
• The enhancement is applied to the following low bandwidth (8kHz) audio
algorithms: G.729a and G.711
Note that when ENABLE_G729 value is NO, G.729 audio codec is
disabled in favor of G.711.
•
•
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Audio Clarity is selected by default.
Audio Clarity cannot be selected and deselected during an ongoing conference.
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New AVC CP Profile - Audio Settings Parameters
Field/Option
Description
Mute participant
except lecturer
When the Mute Participants Except Lecturer option is enabled, the audio of all
participants in the conference except for the lecturer can be automatically muted
upon connection to the conference. This prevents other conference participants
from accidentally interrupting the lecture, or from a noisy participant affecting the
audio quality of the entire conference. Muted participants cannot unmute
themselves unless they are unmuted from the Collaboration Server Web
Client/RMX Manager.
You can enable or disable this option during the ongoing conference.
Notes:
• When enabled, the mute indicator on the participant endpoints are not visible
because the mute participants was initiated by the MCU. Therefore, it is
recommended to inform the participants that their audio is muted by using the
Closed Caption or Message Overlay functions.
In the Collaboration Server Web Client/RMX Manager the mute by MCU
indicator is listed for each muted participant in the Audio column in the
Participants pane.
• This option can be disabled during an ongoing conference, thereby unmuting all
the participants in the conference.
• If the endpoint of the designated lecturer is muted when the lecturer connects to
the conference, the lecturer remains muted until the endpoint has been unmuted.
• When you replace a lecturer, the MCU automatically mutes the previous lecturer
and unmutes the new lecturer.
• When you disconnect a lecturer from the conference or the lecturer leaves the
conference, all participants remain muted but are able to view participants in
regular video layout until the you disable the Mute Participants Except Lecturer
option.
• A participant can override the Mute Participants Except Lecturer option by
activating the Mute All Except Me option using the appropriate DTMF code,
provided the participant has authorization for this operation in the IVR Services.
The lecturer audio is muted and the participant audio is unmuted. You can
reactivate the Mute Participants Except Lecturer option after a participant has
previously activated the Mute All Except Me option. The participant is muted and
the lecturer, if designated, is unmuted.
• In cascaded conferences, all participants (including the link participant) are
muted. Only the lecturer is not muted.
Speaker Change
Threshold
Indicates the amount of time a participant must speak continuously before
becoming the speaker.
Select the desired threshold:
• Auto (Default, 3 seconds)
• 1.5 seconds
• 3 seconds
• 5 seconds
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New AVC CP Profile - Audio Settings Parameters
Field/Option
Description
Auto mute noisy
endpoints
This option is automatically enabled in new Profiles. When enabled, the RMX can
detect AVC endpoints with a noisy audio channel and automatically mute them,
reducing the noise heard by other conference participants. When the auto muted
endpoint becomes the “speaker” the endpoint is automatically un-muted by the
system. If the speaker halts his/her conversation and the line still emits noises, the
endpoint will be automatically muted again.
Clear this check box to disable the feature.
For more details, see Automatic Muting of Noisy Endpoints.
16 Click the Skins tab to modify the background and frames.
The New Profile - Skins dialog box opens.
17 Select one of the Skin options.
Note: Skins - Description and Limitation
• In Classic View (for the first two skin options) the frames fill the screen with their borders touching.
• When Telepresence Mode is enabled, the Skin options are disabled as the system uses a black
background and the frames and speaker indication are disabled.
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18 Click the IVR tab.
The New Profile - IVR dialog box opens.
19 If required, set the following parameters:
New AVC CP Profile - IVR Parameters
Field/Option
Description
Conference IVR
Service
The default conference IVR Service is selected. You can select another conference
IVR Service if required.
Conference Requires
Chairperson
Select this option to allow the conference to start only when the chairperson
connects to the conference and to automatically terminate the conference when the
chairperson exits. Participants who connect to the conference before the
chairperson are placed on Hold and hear background music (and see the Welcome
video slide). Once the conference is activated, the participants are automatically
connected to the conference.
When the check box is cleared, the conference starts when the first participant
connects to it and ends at the predefined time or according to the Auto Terminate
rules when enabled.
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New AVC CP Profile - IVR Parameters
Field/Option
Description
Terminate conference
after chairperson
leaves
Select this check box to automatically terminate the conference after the
chairperson leaves. When the chairperson leaves, the Chairperson Has Left
IVR message is played to (heard by) all participants, at which point the conference
terminates. This way an operator does not need to monitor a conference to know
when to terminate it manually.
If there is a single chairperson in the conference who is changed to a regular
participant the conference will be terminated as if the chairperson left. If there is
more than one chairperson, then changing one chairperson to a regular participant
will not terminate the conference. It is therefore recommended that before changing
a single chairperson to regular participant, another participant first be changed to
chairperson.
Note: Applicability to Cascading Conferences
This applies to cascaded conferences, so long as the Collaboration Server version is 8.6 and up, and
all MCUs in the topology are set in a require a chairperson for the conference duration.
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20 Optional. Click the Recording tab to enable conference recording with the Polycom®
RealPresence® Media Suite.
The New Profile - Recording dialog box opens.
21 Define the following parameters:
New AVC CP Profile - Recording Parameters
Parameter
Description
Enable Recording
Select this check box to enable the Recording of the conference. If no
Recording Links are found, an error message is displayed.
Dial Out Recording
Link
Select the Recording Link to be used for conference recording.
Recording Links defined on the Collaboration Server can be given a
descriptive name and can be associated with a Virtual Recording
Room (VRR) saved on the Polycom® RealPresence® Media Suite.
For more information see Recording and Streaming Conferences
Start Recording
Select when to start the recording:
• Immediately – Conference recording is automatically started upon
connection of the first participant.
• Upon Request – The operator or chairperson must initiate the
recording (manual).
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New AVC CP Profile - Recording Parameters
Parameter
Description
Audio Only
Select this option to record only the audio channel of the conference.
Display Recording
Icon
This option is automatically selected to display a Recording Indication
to all conference participants informing them that the conference is
being recorded.
Clear the selection to prevent the display of the recording icon.
Note: Recording Link
The Recording link (which is listed as a participant in the conference) does not support H.264 High
Profile. If recording a conference that is set to H.264 High Profile, the Recording participant connects
as Audio Only and records only the conference Audio.
22 Click the Site Names tab.
The New Profile - Site Names dialog box opens.
Using the Site Name dialog box, you can control the display of the site names by defining the font,
size, color, background color and transparency and position within the Video Window. For a detailed
description of the site names options see Site Names Definition.
23 Define the following parameters:
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New AVC CP Profile - Site Names Parameters
Field
Description
Display Mode
Select the display mode for the site names:
• Auto - Display the Site Names for 10 seconds whenever the Video Layout
changes.
• On - Display the Site Names for the duration of the conference.
• Off (default) - Do not display the Site Names and all other fields in this tab are
grayed and disabled
Font Size
Click the arrows to adjust the font size (in points) for the display of Site Names.
Choose a Font Size that is suitable for viewing at the conference’s video resolution.
For example, if the resolution is CIF, a larger Font Size should be selected for
easier viewing.
Range: 9 - 32 points
Default: 12 points
Background Color
Select the color of the Site Names display text.
The color and background for Site Names display text is dependent on whether a
Plain Skin or a Picture Skin was selected for the conference in the Profile - Skins
tab. The choices are:
Note: Choose a Background Color combination that is suitable for viewing at the
conference’s video resolution. At low resolutions, it is recommended to select
brighter colors as dark colors may not provide for optimal viewing.
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New AVC CP Profile - Site Names Parameters
Field
Description
Display Position
Select the pre-set position for the display of the Site (endpoint) Names.
Selection
Site Names Position
LeftTop (Default)
Top
RightTop
LeftMiddle
RightMiddle
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New AVC CP Profile - Site Names Parameters
Field
Description
Display Position
(cont.)
LeftBottom
Bottom
RightBottom
Custom
The current endpoint (site)
names display position becomes
the initial position for Site Names
position adjustments using the
Horizontal and Vertical
Position sliders.
Horizontal Position
Move the slider to the left to move the
horizontal position of the endpoint (site)
names to the left within the video windows.
Move the slider to the right to adjust the
horizontal position of the endpoint (site)
names to the right within the video windows.
Note: Use of these sliders will
set the Display Position
selection to Custom.
Vertical Position
Move the slider to the left to move the vertical
position of the Site names upward within the
Video Windows.
Move the slider to the right to move the
vertical position of the endpoint (site) names
downward within the video windows.
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New AVC CP Profile - Site Names Parameters
Field
Description
Background
Transparency
Move the slider to the left to decrease the transparency of the background of the
endpoint (site) names text. 0 = No transparency (solid background color).
Move the slider to the right to increase the transparency of the background of the
endpoint (site) names text. 100 = Full transparency (no background color).
Default: 50
Note: This slider is only displayed if a Picture Skin is selected.
24 Click the Message Overlay tab.
The New Profile - Message Overlay dialog box opens.
Message Overlay enables you to send text messages to all participants during ongoing Continuous
Presence conferences.
The text message is seen as part of the in the participant’s video layout on the endpoint screen or
desktop display.
For more details, see Sending Text Messages During a Conference Using Message Overlay.
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25 Define the following fields:
New AVC CP Profile - Message Overlay Parameters
Field
Description
Enable
This option is disabled by default. Select this check box to enable Message Overlay
or clear it to disable it.
Content
Enter the message text. The message text can be up to 50 Chinese characters.
Font Size
Click the arrows to adjust the font size (points) for the display of the message text.
Font size range: 9 - 32 points, default: 24 points
Note: In some languages, for example Russian, when a large font size is selected,
both rolling and static messages may be truncated if the message length exceeds
the resolution width.
Color
From the drop-down menu select the color and background of the displayed text.
The choices are:
Default: White text on red background.
Vertical Position
Move the slider to the right to move the vertical position of the displayed text
downward within the Video Layout.
Move the slider to the left to move the vertical position of the displayed text upward
within the Video Layout.
Default: Top Left (10)
Background
Transparency
Move the slider to the left to decrease the transparency of the background of the
message text. 0 = No transparency (solid background color).
Move the slider to the right to increase the transparency of the background of the
message text. 100 = Full transparency (no background color).
Default: 50
Display Repetition
Click the arrows to increase or decrease the number of times that the text message
display is to be repeated.
Default: 3
Display Speed
Select whether the text message display is static or moving across the screen, the
speed in which the text message moves: Static, Slow, Fast
Default: Slow
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As the fields are modified the Preview changes to show the effect of the changes.
For example:
26 Click the Network Services tab.
The New Profile - Network Services dialog box opens.
Registration of conferencing entities such as ongoing conferences, Meeting Rooms, Entry Queues,
SIP Factories, and Gateway Sessions with SIP servers is done per conferencing entity. This allows
better control on the number of entities that register with each SIP server.
Selective registration is enabled by assigning a conference Profile in which registration is configured
to the required conferencing entities. Assigning a conference Profile in which registration is not
configure to conferencing entities will prevent them from registering.
By default, Registration is disabled in the Conference Profile, and must be enabled in Profiles
assigned to conferencing entities that require registration.
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27 Define the following parameters:
New AVC CP Profile - Network Services Parameters
Parameter
Description
IP Network Services
Service Name
This column lists all the defined Network Services, one or several depending on the
system configuration.
SIP Registration
To register the conferencing entity to which this profile is assigned with the SIP
Server of the selected Network Service, click the check box of that Network Service
in this column.
When SIP registration is not enabled in the conference profile, the Collaboration
Server's registering to SIP Servers will each register with an URL derived from its
own signaling address. This unique URL replaces the non-unique URL,
dummy_tester, used in previous versions.
Accept Calls
To prevent dial in participants from connecting to a conferencing entity when
connecting via a Network Service, clear the check box of the Network Service from
which calls cannot connect to the conference.
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28 Click the Layout Indications tab.
The Layout Indications dialog is displayed.
The Layout Indications are displayed in the conference video layout as an icon group that includes:
 Recording
 Audio Participants
 Network Quality
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Note: Layout Indications tab
• The Layout Indications tab is only displayed in either CP or CP and SVC (mixed) conferencing
modes.
• Beginning with version 8.4, the configuration options of the Layout Indications tab have replaced the
following system flags:
 DISABLE_SELF_NETWORK_IND
 SELF_IND_LOCATION
For further information on layout indications see Layout Indications.
29 Modify the following parameters to configure the display, position, and duration of the indication
icons.
New AVC CP Profile - Layout Indications Parameters
Field
Description
Position
Use the Position drop-down menu to configure the display position of the indication
icons group. Icons can be displayed in the following positions:
• Top-left
• Bottom-left
• Top center (default)
• Bottom
• Top-right
• Bottom-right
Recording
Select the check box to display the Recording icon.
The Recording indication icon is a duplicate of the Display Recording Icon field in the
Recording tab of the Profile dialog.
For more information, see Recording and Streaming Conferences.
Audio Participants
Select the check box to display the Audio Participants icon.
• Permanent - Sets the Audio Participant Indication to display permanently. In this
setting it is only displayed when audio participants are connected.
• On audio participants change - Sets the Audio Participant Indication to display for
a short period only when the number of audio participants changes.
 Duration - Sets the amount of time the Audio Participant icon displays.
For more information see Audio and Video Participants Indication.
Network Quality
To display the Network Quality icon, select the check box.
For more information see Network Quality Indication.
30 Click OK to complete the Profile definition.
A new Profile is created and added to the Conference Profiles list.
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Additional Information for Setting CP Profiles
This section includes detailed explanation of various CP Profile settings:
● Gathering Phase
● Overlay Layouts
● Site Names Definition
● Sending Text Messages During a Conference Using Message Overlay
● Selecting the Chinese Font for Text Display
Gathering Phase
The Gathering Phase of an AVC (CP only) conference is the time period during which participants are
connecting to a conference. During the Gathering Phase, a mix of live video from connected endpoints is
combined with both static and variable textual information about the conference into a slide which is
displayed on all connected endpoints.
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During the Gathering Phase, the audio of all participants can be heard, and the video of active speakers is
displayed in the video windows as they begin talking.
All connected participants are kept informed about the current conference status including names of
connected participants, participant count, participant type (video/audio) etc.
Gathering Phase Guidelines
● Gathering Phase is only available in AVC only (CP only) conferences. It is not supported in Video
Switching conferences and SVC Only conferences.
● The Gathering Phase slide can be displayed at any time during the conference by entering the Show
Participants DTMF code, *88.
Note: When the display of the Gathering Phase slide is removed, the message overlay text is also
removed.
● The names of the first eight participants to connect are displayed. If eight or more participants
connect, the 8th row displays “…”.
● Static text in the Gathering Phase slide such as the field headings: Organizer, Duration, Video/Audio
Participants, Access Number, IP are always displayed in the language as configured in the Polycom
Virtual Meeting Rooms Add-in for Microsoft Outlook. The following languages are supported:
 English
 French
 German
 International Spanish
 Korean
 Japanese
 Simplified Chinese
● Dynamic text in the Gathering Phase slide such as the meeting name, participants’ names, access
numbers and the additional information entered in the Info1/2/3 fields of the Gathering Settings tab
of the conference Profile are displayed in the language of the meeting invitation.
● The language of a Gathering Phase slide of a conference configured to include a Gathering Phase
that is not launched by the Polycom Conferencing Add-in for Microsoft Outlook is configured by the
administrator. Using the Collaboration Server Web Client, the administrator selects the language for
the Gathering Phase slide. The language selected can be different to that of the Collaboration Server
Web Client used by the administrator to perform the configuration.
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● Content can be sent during the Gathering Phase. The content is displayed in the large video window
of the participant’s layout while the Gathering slide is displayed in a smaller video window in the
layout.
● Gathering is not supported in Cascading Conferences.
Gathering Phase Duration
The duration of the Gathering Phase can be customized by the administrator so that it is long enough to be
viewed by most connected participants yet short enough so as not to over extend into the scheduled
conferencing time.
The Gathering Phase duration is configured for the Collaboration Server, by the following System Flags in
system.cfg in the Setup > System Configuration:
● CONF_GATHERING_DURATION_SECONDS
Range: 0 - 3600 seconds
Default: 180 seconds
The Gathering Phase duration of the conference is measured from the scheduled start time of the
conference.
Example: If the value of the flag is set to 180, the Gathering slide is displayed for three minutes to all
participants starting at the conference Start Time, and ending three minutes after the conference Start
Time.
For participants who connect before Start Time, the Gathering slide is displayed from the time of
connection until the end of the Gathering duration period.
● PARTY_GATHERING_DURATION_SECONDS
Range: 0 - 3600 seconds
Default: 15 seconds
The value of this flag determines the duration of the display of the Gathering slide for participants that
connect to the conference after the conference Start Time.
Participants connecting to the conference very close to of the end of the Gathering Phase (when there
are fewer seconds left to the end of the Gathering Phase than specified by the value of the flag) have
the Gathering slide displayed for the time specified by the value of the flag.
Example: If the value of the flag is set to 15, the Gathering Phase slide is displayed to the participant
for 15 seconds.
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Enabling the Gathering Phase Display
The Gathering Phase is enabled for per conference in the Conference Profile. The profile also includes the
dial-in numbers and the optional additional information to display on the slide.
Conferences that are configured to include a Gathering Phase that are not launched by the Polycom
Conferencing Add-in for Microsoft Outlook need the following information to be entered via the New Profile
or Profile Properties — Gathering Settings dialog box:
● Display Name (Optional, the Meeting Name is used if left blank.)
● Displayed Language
● Access Number 1 / 2 (Optional.)
● Additional Information (Optional free text)
 Info 1
 Info 2
 Info 3
Conferences launched by the Polycom Conferencing Add-in for Microsoft Outlook receive this information
from the meeting invitation.
For more information see Defining New Profiles.
Overlay Layouts
In Overlay Layouts additional participant endpoints can be displayed over the full screen display of the
conference speaker.
The following Overlay Layouts are available for use in CP Conferences:
Although the following Overlay Layout is included in the Profiles - Video Settings dialog box, it is not
available for use in any Conferencing Mode and is only available when included in the Polycom® Multipoint
Layout (MLA) application:
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Note: Overlay Layouts Availability
These Overlay Layouts will only be available in ITP (Telepresence) conferences when support for
Overlay Layouts is included in the Multipoint Layout (MLA) application.
Guidelines for using the Overlay Layouts
● The Overlay Layouts are supported:
 In CP Conferencing Mode only.
 With ITP, non-ITP and CTS endpoints used only as standard endpoints.
 With both new and classic Skins in Collaboration Server CP mode.
● Overlay Layouts are not supported in ITP conferences as they are not supported by the MLA
application.
● The Overlay Layouts are 20% of the height of the endpoint display and are supported on endpoints
of both 16:9 and 4:3 aspect ratios.
● Overlay Layouts are recommended for use with high resolution endpoints.
● Overlay Layouts are not selected as defaults by the system and are not included in the Auto Layout
settings.
● Message Overlay is not affected by the use of Overlay Layouts and is displayed on top of the video
layouts.
● Site Names are displayed for all cells. Because the smaller cells are located at the bottom of the large
cell, when enabling Site Names it is advisable not to locate the Site Name at the bottom of the cells.
● Standalone Endpoint Cells are displayed each with a border. For all Overlay Layouts, border color is
dependent on the selected Skin.
● System behavior for Video Forcing and Personal Layout Control when using the Overlay Layouts
during an ongoing conference is the same as for other video layouts.
● Overlay Layouts are only available for selection for the Conference Layout and are not available for
selection for Personal Layout.
● During an ongoing conference you cannot select the Overlay Layouts via PCM or Click&View.
● PCM menus can be used when the Overlay Layouts are active, and they are displayed as the top
layer.
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Selecting the Overlay Layouts
The Overlay Layouts are selected in the New Profile - Video Settings dialog box, in the Overlay tab of the
Video Layout tree.
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Site Names Definition
Note: Site Names Definition
The Site Names feature is not supported on the Collaboration Server (RMX) 1800.
During conferences you can view the names of the endpoints that are connected to the conference in your
endpoint video layout windows. The MCU can display up to 33 characters of the endpoint name, depending
on the window layout (size).
The following is an example of endpoint name display in the endpoint screen:
The display of the site names is enabled or disabled in the conference Profile for CP Only or Mixed CP and
AVC conferences for AVC-based endpoints.
SVC-based endpoints handle the display of the site names themselves and are not dependant on the
conference Profile. The Site Names option is not supported in Video Switching conferences.
The displayed name is determined as follows:
● The system displays the name that is defined at the endpoint.
● If the endpoint does not send its name:
 For a defined H.323 or SIP participant:
 The system displays the name from the participant definition.
 For an undefined H.323 participant:
 Display the H.323 ID alias.
or
Display the E.164 alias.
or
Display nothing if all the fields are empty.
 For a SIP undefined participant:
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 Display the SIP DisplayName field.
or
Display the SIP Address (SIP application server).
or
Display the SIP ContactDisplay field.
or
Display nothing if all the fields are empty.
 For a defined H.320 participant (Collaboration Server 2000/4000):
 The system displays the name from the participant definition.
 For an undefined H.320 participant:
 Display the Terminal Command String (TCS-2) to identify the participant.
or
Display nothing if the string is not received or empty.
● If the endpoint Display Name is changed in the Collaboration Server Web Client, it overrides all the
above.
You can control the display of the site names by defining the font, size, color, background color and
transparency and position within the video window in the Profile - Site Name dialog box.
Guidelines
● Site Names display is Off by default in a new profile.
● Site Names can be enabled to function in one of two modes:
 Auto – Site names are displayed for 10 seconds whenever the conference layout changes.
 On – Site names are displayed for the duration of the conference.
● During the display of the site names, the video frame rate is slightly reduced
● Site Names display is not available for Video Switching (VSW) conferences.
● Site Names display characteristics (position, size, color) can by modified during an ongoing
conference using the Conference Properties - Site Names dialog box. Changes are immediately
visible to all participants.
● Site Names display text and background color is dependent on the Skin selected for the conference:
 Plain Skins - Site Names text is displayed without a background.
 Picture Skins - Site Names text is displayed with a background.
Shorten the Site Name Display
The SIP_OMIT_DOMAIN_FROM_PARTY_NAME System Flag can be used to remove Domain Names
from SIP dial-in participants’ Site Names. This prevents long domain names being appended to SIP
participant names, as frequently happens when the Collaboration Server is used with a DMA.
The flag must be manually added to the System Configuration and its value modified as follows:
● YES (Default) - The domain name is omitted from SIP dial-in participant names.
● NO - The domain name remains as part of SIP dial-in participant names.
For more information, see Modifying System Flags.
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Site Names Display Position
The position of the Site Names displayed during the conference is controlled in the Profile - Site Names tab.
The following options can be used to define the display position:
● Display Position drop-down menu - To select a preset position.
● Horizontal Position and Vertical Position sliders - To move them to customize the preset position.
Dragging the sliders sets the Display Position drop-down menu field value to Custom.
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● Selecting Custom in Display Position drop-down menu. When selected, the current position
becomes the initial position for position adjustments and then using the Horizontal and Vertical
Position sliders to define the exact position.
The adjusted position of the Site Names can be viewed in the Preview.
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Sending Text Messages During a Conference Using Message Overlay
The Message Overlay option in the Conference Profile allows the operator or administrator to send text
messages to a single participant, a number of selected participants or all participants during an ongoing
conference.
The number of characters that can be included in a message varies according to the language and can differ
due to the type of font used. For example, the available number of characters in Chinese is 50 (32 for
Collaboration Server 1800) for English and 50 (48 for Collaboration Server 1800) and for Russian.
Note: Language Dependency
In some languages, for example Russian, when a large font size is selected, both rolling and static
messages may be truncated if the message length exceeds the resolution width.
The message can be set to be displayed at various positions on the screen in different colors, static or
scrolling.
Guidelines
● Text messaging using Message Overlay is supported in:
 Continuous Presence (CP) conferences
 Same Layout mode
 Encrypted conferences
 Unicode or ASCII characters
● Text messages using Message Overlay cannot be displayed:
 In Video Switching (VSW) conferences.
 In Lecture Mode
 When the PCM menu is active
 On endpoints that have their video suspended
● Text messaging using Messages Overlay can be enabled, disabled or modified (content and display
parameters) during the ongoing conference.
● The number of characters for each language can vary due to the type of font used, for example, the
available number of characters for Chinese is 18, while for English and Russian it is 48.
 In some languages, for example Russian, when large font size is selected, both rolling and static
messages may be truncated if the message length exceeds the resolution width.
● Changes to the Message Overlay Content or display characteristics (position, size, color and speed)
are immediately visible to all participants. When there is a current Message Overlay:
 The current message is stopped immediately, even it has not completed all of its repetitions.
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 The Display Repetition count is reset to 1.
 The new message content is displayed <Display Repetition> times or until it is stopped and
replaced by another content change.
● If during the ongoing conference the Show Number of Participants DTMF option (default DTMF
*88) is used, when the displayed number of participants is removed, the message overlay text is also
removed.
● The text messages cannot be sent via the Content channel.
● Message Overlay text settings are not saved in the Conference Template when saving an ongoing
conference as a Conference Template.
For a detailed description of all the Message Overlay parameters, see New AVC CP Profile - Message
Overlay Parameters.
Sending Messages to All Conference Participants using Message Overlay
To send messages to all conference participants Using Message Overlay:
1 In the Conferences List pane, double click the name of the conference to have Message Overlay
enabled or right-click the conference name and then click Conference Properties.
2 Click the Message Overlay tab.
3 Select the Enable check box.
The Send Text Message to Participant dialog box is displayed.
4 In the Content field, enter the text to be displayed to all the conference participants.
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5 Modify the remaining fields in the dialog box to display the message as required.
6 Click OK
Note: Resuming Overlay Messaging
When sending text messages using Message Overlay and stopping, you must artificially change
(and cancel the change) one of the message overlay properties in the Conference Properties Message Overlay dialog box before you can resume sending the same message.
To cancel the Message Overlay display:
1 In the Conferences List pane, double click the name of the conference to have Message Overlay
enabled or right-click the conference name and then click Conference Properties.
2 Click the Message Overlay tab.
3 Clear the Enable check box.
4 Click OK.
Sending Messages to Selected Participants Using Message Overlay
To send text to selected participants:
1 In the Participant List pane, choose a participant or a number of participants.
2 In the Participant List pane, right-click a participant or a number of participants and then select Send
Text Message to Participant.
The rest of configuration steps are the same with what in sending messages to all conference
participants, see Sending Messages to All Conference Participants using Message Overlay.
Selecting the Chinese Font for Text Display
When using the RMX Web Client or the RMX Manager in Chinese (either Simplified Chinese or Traditional
Chinese is selected as an available language in the Setup > Customize Display Settings > Multilingual
Setting, you can select one of several Chinese fonts for use when sending text over video. The font is used
to display text for the following:
● Display of Site Names
● Test messages sent using Message Overlay
● Text displayed on the Gathering slide when Chinese is selected as the display language
Selecting the Chinese Font
The Chinese fonts can be selected in the CP Conference Profile - Advanced dialog box only.
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The following Chinese fonts are available for selection:
● Heiti (Default)
● Songti
● Kaiti
● Weibei
The Chinese font cannot be changed during an existing conference. It can only be modified in the
conference profile.
A participant moved to another conference will be shown the font used by the new conference.
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Defining an AVC Video Switching Conference Profile
Note: VSW Conferencing Mode Platform Limitation
• VSW conferencing mode in not supported in Collaboration Server Virtual Edition.
• Video Switching conferencing mode is unavailable to ISDN participants.
An AVC Video Switching-enabled Profile must be created prior to running Video Switching conferences.
This profile can be assigned to Meeting Rooms, conferences, reservations and Entry Queues
To connect to a Video Switching conference via an Entry Queue, the Entry Queue must be Video Switching
enabled and must be set to the same line rate as the target conference. It is recommended to use the same
Profile for both the target conference and Entry Queue.
Video Switching conferencing mode is unavailable to ISDN participants.
When the Conferencing Mode is set to Video Switching, only the tabs and fields relevant to Video Switching
conferences are displayed and enabled.
To Create a Video Switching Profile:
1 In the RMX Management pane, click Conference Profiles.
2 In the Conference Profiles pane, click the New Profile button.
The New Profile – General dialog box opens.
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3 Define the New Profile - General parameters:
New AVC VSW Profile - General Parameters
Field/Option
Description
Display Name
Enter a unique Profile name, as follows:
• English text uses ASCII encoding and can contain the most characters (length
varies according to the field).
• European and Latin text length is approximately half the length of the maximum.
• Asian text length is approximately one third of the length of the maximum.
It is recommended to use a name that includes the Profile type, such as Weekly
Video Switching conference.
Note: This field is displayed in all tabs.
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New AVC VSW Profile - General Parameters
Field/Option
Description
Line Rate
Select the conference bit rate. The line rate represents the combined video, audio
and Content rate.
When defining a VSW profile, select a line rate that all connecting participants can
use. Participants that their endpoint or network that do not support this line rate
cannot connect to the conference or will connect as Audio Only (if resources were
designated as Voice ports).
If a high definition resolution will be selected for the conference video, make sure
that the selected line rate is higher than the minimum line rate threshold defined in
the flag HD_THRESHOLD_BITRATE for Video Switching conferences.
The default setting is 384 Kbps.
Note: This field is displayed in all tabs.
Conferencing Mode
Select VSW (Video Switching) to define a VSW conference Profile.
Note: This field is displayed in all tabs.
Routing Name
Enter the Profile name using ASCII characters set.
The Routing Name can be defined by the user or automatically generated by the
system if no Routing Name is entered as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered in
Display Name, the ID (such as Conference ID) is used as the Routing Name.
Video Switching
This check box is automatically selected when the Conferencing Mode is set to
Video Switching.
Select the video protocol and resolution for the conference:
• H.264 1080p60
• H.264 1080p30
• H.264 720p60
• H.264 720p30
• H.264 SD 30
• H.264 CIF
• H.263 CIF
• H.261 CIF
All participants must connect at the same line rate and use the same video
resolution. Participants with endpoints that do not support the selected line rate and
resolution will connect as secondary (audio only).
For more information, see Video Switching (VSW) Conferencing.
H.264 High Profile
Select this check box to enable the use of H.264 High Profile in Video Switching
conferences.
By default the High Profile check box is not selected.
If H.264 is not the selected video protocol the check box is inactive (grayed out).
For more information, see H.264 High Profile Support in Video Switching
Conferences.
Operator Conference
This option is unavailable and disabled in VSW Conferencing Mode.
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Note: Conference Line Rate reduction Impact on Resolution
Reducing the conference line rate may result in the system reverting to the default resolution for that
line rate (for example, 720p instead of 1080p). You may need to select the required resolution again,
provided the selected line rate is higher than the minimum threshold line rate defined for that
resolution in the system configuration. For more details, see Minimum Threshold Line Rate System
Flags below.
4 Define the various Profile parameters for a Video Switching conference. As it is an AVC -bases
conferencing mode, many of the fields and options are identical to the CP Profile. For more
information, see Defining AVC CP Conferencing Profiles.
The following AVC-based conferencing options are not available in VSW conferences:
 Operator Conference
 Gathering Phase
 Video Quality - Send Content To Legacy Endpoints
 Video Settings:
 Presentation Mode
 Auto Layout/Same Layout (only full screen, 1x1 layout display is available)
 Auto Scan
 Skins
 Site Names
 Message Overlay
5 Click OK.
H.264 High Profile Support in Video Switching Conferences
Beginning with Version 7.6, the H.264 High Profile video protocol is supported in Video Switching (VSW)
conferences.
Guidelines
● H.264 High Profile is supported in VSW conferences in H.323 and SIP networking environments only
● For H.264 High Profile-enabled VSW conferences, all endpoints connecting to the conference must
support High Profile and connect to the VSW conference at the exact line rate and exact resolution
defined for the conference. Endpoints that do not meet these requirements are connected as
Secondary (audio only).
● For H.264 Base Profile VSW conferences, both High Profile and Base Profile supporting endpoints
connect using the H.264 Base Profile.
 Endpoints that do not support the exact conference line rate are disconnected.
 Endpoints that do not support the exact video settings such as protocol and resolution defined for
the conference will be connected as Secondary (audio only).
Minimum Threshold Line Rate System Flags
The following table lists the System Flags that control the minimum line rate threshold for the various
resolutions available for High Profile-enabled VSW conferences.
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These system flags must be manually added to the Setup menu > System Configuration before you can
update their values. For more information see the Modifying System Flags.
System Flags - Minimum Threshold Line Rates
Flag Name
Minimum Threshold Line Rate (Kbps)
VSW_CIF_HP_THRESHOLD_BITRATE
64
VSW_SD_HP_THRESHOLD_BITRATE
128
VSW_HD_720p30_HP_THRESHOLD_BITRATE
512
VSW_HD_720p50_60_HP_THRESHOLD_BITRATE
832
VSW_HD_1080p_HP_THRESHOLD_BITRATE
1024
VSW_HD_1080p60_HP_THRESHOLD_BITRATE
1024
VSW_HD_1080p60_BL_THRESHOLD_BITRATE
1728
Line rate and resolution combinations are checked for validity. If the selected line rate is below the minimum
line rate threshold required for the selected resolution, the line rate is automatically adjusted to the minimum
line rate threshold value for the selected resolution.
The value of the SUPPORT_HIGH_PROFILE system flag (used for CP conferences) has no effect on VSW
conferences.
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Conference Profiles
Note: Using Conference Profiles in Version 8.1.4.J.
Although SVC Conferencing Mode options are available in Conference Profiles, it is advised that
they not be used with Version 8.1.4.J.
Note: Recommendations and Limitations
When using Polycom® RealPresence® Platform, the Conference Profiles are defined in the
RealPresence DMA system component and not directly in RealPresence Collaboration Server
(RMX) 1500/1800/2000/4000.
Defining SVC Conference Profiles
The SVC conference Profile definition is started by selecting SVC as the Conferencing Mode. The dialog
boxes and their options change as the conference behavior and the MCU video processing change. For
example, site name display is performed and controlled by the SVC endpoint and not by the MCU as in CP
conferences.
To define SVC Only Profile:
1 In the RMX Management pane, click Conference Profiles.
2 In the Conference Profiles pane, click the New Profile button.
The New Profile – General dialog box opens.
By default, the Conferencing Mode is set to CP.
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3 Select SVC Only to define the SVC Profile.
The profile tabs and options change accordingly and only supported options are available for
selection. Unsupported options are disabled (grayed out).
4 Define the Profile name and, if required, the Profile - General parameters:
New SVC Profile - General Parameters
Field/Option
Description
Display Name
Enter a unique Profile name, as follows:
• English text uses ASCII encoding and can contain the most characters (length
varies according to the field).
• European and Latin text length is approximately half the length of the maximum.
• Asian text length is approximately one third of the length of the maximum.
This is the only parameter that must be defined when creating a new profile.
Note: This field is displayed in all tabs.
Line Rate
Select the conference bit rate. The line rate represents the combined video, audio
and Content rate.
The default setting for SVC Only conferences is 1920kbps.
Note: This field is displayed in all tabs.
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New SVC Profile - General Parameters
Field/Option
Description
Routing Name
Enter the Profile name using ASCII characters set.
You can define the Routing Name or it can be automatically generated by the
system if no Routing Name is entered as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered in
Display Name, the ID (such as Conference ID) is used as the Routing Name.
5 Click the Advanced tab.
The New Profile – Advanced dialog box opens.
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6 Define the following supported parameters:
New SVC Profile - Advanced Parameters
Field/Option
Description
Encryption
Select the Encryption option for the conference:
• Encrypt All - Encryption is enabled for the conference and all conference
participants must be encrypted.
• No Encryption - Encryption is disabled for the conference.
• Encrypt when Possible - enables the negotiation between the MCU and the
endpoints and let the MCU connect the participants according to their
capabilities, where encryption is the preferred setting. For connection guidelines
see Mixing Encrypted and Non-encrypted Endpoints in one Conference.
For more information, see Packet Loss Compensation - LPR and DBA.
Auto Terminate
When selected (default), the conference automatically ends when the termination
conditions are met:
• Before First Joins — No participant has connected to a conference during the
n minutes after it started. Default idle time is 10 minutes.
• At the End - After Last participant Quits — All the participants have
disconnected from the conference and the conference is idle (empty) for the
predefined time period. Default idle time is 1 minute.
• At the End - When Last Participant Remains — Only one participant is still
connected to the conference for the predefined time period (excluding the
recording link which is not considered a participant when this option is selected).
It is not recommended to select this option for SVC Conferences. Default idle
time is 1 minute.
Exclusive Content
Mode
When selected, Content broadcasting is limited to one participant preventing other
participants from interrupting the Content broadcasting while it is active. For more
details, see
FW NAT Keep Alive
When selected, a FW NAT Keep Alive message is sent at an interval defined in the
field below the check box.
Interval
The time in seconds between FW NAT Keep Alive messages.
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7 Click the Video Quality tab.
The New Profile – Video Quality dialog box opens.
8 In SVC Conferencing Mode, the video and Content sharing parameters cannot be modified and they
are set to the following parameters:
New SVC Profile - Video Quality Parameters
Field/Option
Description
People Video Definition
Video Quality
Only Sharpness is available in SVC Conferencing Mode. The MCU
sends the video stream in the resolution required by the endpoint.
Maximum Resolution
Only Auto is available in SVC Conferencing Mode. The MCU sends
the video stream in the resolution required by the endpoint.
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New SVC Profile - Video Quality Parameters
Field/Option
Description
Content Video Definition
Content Settings
Select the transmission mode for the Content channel:
• Graphics — basic mode, intended for normal graphics
• Hi-res Graphics — a higher bit rate intended for high resolution
graphic display
• Live Video — Content channel displays live video
• Customized Content Rate — manual definition of the Conference
Content Rate, mainly for cascading conferences.
Selection of a higher bit rate for the Content results in a lower bit rate
for the people channel.
Content Protocol
Select the Content Protocol to be used for content sharing in Highest
Common Content Sharing Mode.
• H.263
Content is shared using the H.263 protocol.
Use this option when most of the endpoints support H.263 and
some endpoints support H.264.
• H.263 & H.264 Auto Selection
When selected, content is shared using H.263 if a mix of
H.263-supporting and H.264-supporting endpoints are connected,
or H.264 if all connected endpoints have H.264 capability.
• H.264 Cascade Optimized
All Content is shared using the H.264 content protocol and is
optimized for use in cascaded conferences.
• H.264 HD (default)
Ensures high quality Content when most endpoints support H.264
and HD resolutions.
Note: When Multiple Content Resolutions is selected, the Content
Protocol field is hidden.
For more information, see Content Protocols and Defining Content
Sharing Parameters for a Conference
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9 Click the Video Settings tab.
In SVC conferences, each endpoint determines its own video layout and there is no conference level
layout selected. Therefore, all the Video Settings parameters are disabled.
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10 Click the Audio Settings tab.
11 If required, define the Speaker Change Threshold: Auto (Default, 3 seconds), 1.5, 3, or 5.
It indicates the amount of time a participant must speak continuously before becoming the speaker.
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12 Click the IVR tab.
13 If required, set the following parameters:
New SVC Profile - IVR Parameters
Field/Option
Description
Conference IVR
Service
The default conference IVR Service is selected. You can select
another conference IVR Service if required.
Conference Requires
Chairperson
Select this option to allow the conference to start only when the
chairperson connects to the conference and to automatically
terminate the conference when the chairperson exits. Participants
who connect to the conference before the chairperson are placed on
Hold and hear background music (and see the Welcome video slide).
Once the conference is activated, the participants are automatically
connected to the conference.
When the check box is cleared, the conference starts when the first
participant connects to it and ends at the predefined time or according
to the Auto Terminate rules when enabled.
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New SVC Profile - IVR Parameters
Field/Option
Description
Terminate conference
after chairperson
leaves
Select this check box to automatically terminate the conference after
the chairperson leaves. When the chairperson leaves, the
“Chairperson Has Left” IVR message is played to all participants, at
which point the conference terminates. This way an operator does not
need to monitor a conference to know when to terminate it manually.
If there is a single chairperson in the conference who is changed to a
regular participant the conference will be terminated as if the
chairperson left. If there is more than one chairperson, then changing
one chairperson to a regular participant will not terminate the
conference. It is therefore recommended that before changing a
single chairperson to regular participant, another participant first be
changed to chairperson.
Applicability to Cascading Conferences
This applies to cascaded conferences, so long as the Collaboration Server version is 8.6 and up, and
all MCUs in the topology are set in a require a chairperson for the conference duration.
The following IVR features are not supported during SVC conferences:
 Roll Call
 Invite Participants
 Entry and Exit tones
 Click & View
 PCM
Note: Features applicability in SVC conferences
On Collaboration Server1800, Roll Call, Entry and Exit tones are supported in SVC-only
conferences.
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14 Click the Network Services tab.
The New Profile - Network Services tab opens.
Registration of conferencing entities such as ongoing conferences, Meeting Rooms, and SIP
Factories with SIP servers is done per conferencing entity. This allows better control of the number
of entities that register with each SIP server. Selective registration is enabled by assigning a
conference Profile in which registration is configured for the required conferencing entities. Assigning
a conference Profile in which registration is not configure for conferencing entities will prevent them
from registering. By default, Registration is disabled in the Conference Profile, and must be enabled
in Profiles assigned to conferencing entities that require registration.
15 Define the following parameters:
New SVC Profile - Network Services Parameters
Parameter
Description
IP Network Services
Service Name
Polycom®, Inc.
This column lists all the defined Network Services, one or several
depending on the system configuration.
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New SVC Profile - Network Services Parameters
Parameter
Description
SIP Registration
To register the conferencing entity to which this profile is assigned
with the SIP Server of the selected Network Service, click the check
box of that Network Service in this column.
When SIP registration is not enabled in the conference profile, the
Collaboration Server's registering to SIP Servers will each register
with an URL derived from its own signaling address.
Accept Calls
To prevent dial in participants from connecting to a conferencing
entity when connecting via a Network Service, clear the check box of
the Network Service from which calls cannot connect to the
conference.
16 Click OK to complete the Profile definition.
A new Profile is created and added to the Conference Profiles list.
Note: Layout Indications inapplicability to SVC Only Conferences
The Layout Indications tab is relevant only to AVC only or Mixed CP and SVC conferencing modes,
therefore though visible in SVC only conferencing mode, it is completely disabled and irrelevant.
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Defining Mixed CP and SVC Conferencing Profiles
The mixed CP and SVC Profile is based on the CP Profile with a few of the CP options disabled for
compatibility between AVC and SVC protocols and to enable the media conversion between these two
modes. The Gathering Settings and the Message Overlay options are unavailable in this Conferencing
Mode.
In a mixed CP and SVC conference, the Chairperson can be either an AVC-enabled or SVC-enabled
endpoint.
To configure a mixed AVC and SVC conference:
1 In the RMX Management pane, click Conference Profiles.
2 In the Conference Profiles pane, click the New Profile button.
The New Profile - General dialog box is displayed.
3 In the Conferencing Mode list, select CP and SVC to define a mixed AVC and SVC conference.
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Using the various Profile tabs, you can define the following profile parameters:
● CP and SVC Profile - Advanced parameters - these parameters are the same as for CP
conferences. For details, see New AVC CP Profile - Advanced Parameters.
● CP and SVC Profile - Video Quality parameters - to enable the sharing of video between SVC and
AVC, the common denominator parameters (in this conference, the SVC parameters) are selected
for the conference. For more details, see New SVC Profile - Video Quality Parameters.
● CP and SVC Profile - Video Settings parameters - the video layout parameters apply only to the
AVC-enabled endpoints and do not apply to SVC-enabled endpoints as the SVC endpoints generate
their own layout. Options that are not supported in SVC conferencing are disabled in this dialog box,
for example, Telepresence Mode. For more details, see New AVC CP Profile - Video Settings
Parameters.
● CP and SVC Profile - Audio Settings parameters - options that are not supported in SVC
conferencing are disabled in this dialog box. For more details, see New AVC CP Profile - Audio
Settings Parameters.
● CP and SVC Profile - Skins parameters - the display of a video skin applies only to the AVC-enabled
endpoints and do not apply to SVC-enabled endpoints as the SVC endpoints generate their own
layout.
● CP and SVC Profile - IVR parameters - to enable the same IVR behavior and DTMF usage for SVC
and AVC, the common denominator parameters (in this conference, the SVC parameters) are
selected for the conference. For more details, see New SVC Profile - IVR Parameters.
● CP and SVC Profile - Recording parameters - these parameters are the same as for CP
conferences as the recording is done in AVC format. For details, see New AVC CP Profile - Recording
Parameters.
● CP and SVC Profile - Site Names parameters - these parameters are the same as for CP
conferences as they apply the AVC-enabled endpoints. SVC-enabled endpoints generate the site
name display independent of the MCU. For details, see New AVC CP Profile - Site Names
Parameters.
● CP and SVC Profile - Network Services parameters - these parameters are the same as for CP and
SVC conferences. For details, see New AVC CP Profile - Network Services Parameters.
● CP and SVC Profile - Layout Indications parameters - these parameters are the same as for CP
conferences. For details, see New AVC CP Profile - Layout Indications Parameters.
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Video Protocols and Resolution
Configuration for CP Conferencing
Video Resolutions in AVC CP Conferencing
Note: Chapter Relevancy
For a description of resolutions for SVC Conferencing Mode see Defining SVC and Mixed CP and
SVC Conference Profiles.
The RealPresence Collaboration Server always attempts to connect to endpoints at the highest line rate
defined for the conference. If the connection cannot be established using the conference line rate, the
Collaboration Server attempts to connect at the next highest line rate at its highest supported resolution.
Depending on the line rate, the Collaboration Server sends video at the best possible resolution supported
by the endpoint regardless of the resolution received from the endpoint.
The video resolution is also defined by the Video Quality settings in the Profile.
● Motion, when selected, results in lower video resolution at higher frame rates (30 fps to 60 fps).
● Sharpness, when selected, results in higher video resolution at lower frame rate (30 fps and lower).
However, it can also be sent in 1080p 60fps.
The combination of frame rate and resolution affects the number of video resources required on the MCU
to support the call.
The following resolutions are supported:
Supported Resolutions
Resolution Name
Resolution in Pixels
Frames per Second
Symmetric/Asymmetric
CIF
352 x 288
30, 60
Symmetric
SD
720 x 576
30, 60
Symmetric
HD 720p
1280 x 720
30, 60
Symmetric
HD 1080p
1920 x 1080
30
Symmetric
HD 1080p
1920 x 1080
60
Symmetric in Motion,
Asymmetric in Sharpness
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Video Display with CIF, SD and HD Video Connections
Although any combination of CIF, SD and HD connections is supported in all CP conferences, the following
rules apply:
● In a 1X1 Video Layout:
 SD: If the speaker transmits CIF, the MCU will send CIF to all participants, including the SD
participants. In any other layout the MCU will transmit to each participant at the participant’s
sending resolution.
 HD: The MCU transmits speaker resolution (including input from HD participants) at up to SD
resolution. If 1x1 is the requested layout for the entire duration of the conference, set the
conference to HD Video Switching mode.
● In asymmetrical Video Layouts:
 SD: A participant in the large frame that sends CIF is displayed in CIF.
 HD: Where participants’ video windows are different sizes, the Collaboration Server transmits HD
and receives SD or lower resolutions.
● In panoramic Video Layouts:
 SD: Participants that send CIF also receive CIF.
 HD: the Collaboration Server transmits HD and receives SD or lower resolutions, the
Collaboration Server scales images from SD to HD resolution.
H.264 High Profile Support in CP Conferences
The H.264 High Profile is a new addition to the H.264 video protocol suite. It uses the most efficient video
data compression algorithms to even further reduce bandwidth requirements for video data streams.
Video quality is maintained at bit rates that are up to 50% lower than previously required. For example, a
512Kbps call will have the video quality of a 1Mbps HD call while a 1Mbps HD call has higher video quality
at the same (1Mbps) bit rate.
Note: H.264 HD Recommendation
H.264 High-Profile should be used when all or most endpoints support it.
H.264 High Profile Guidelines
● H.264 High Profile is supported in H.323, and SIP and ISDN networking environments.
● H.264 High Profile is supported in all conference modes, all bit rates, video resolutions and layouts.
● H.264 High Profile is the first protocol declared by the Collaboration Server, to ensure that endpoints
that support the protocol will connect using it.
Setting minimum bit rate thresholds that are lower than the default may affect the video quality of
endpoints that do not support the H.264 High Profile.
● For monitoring purposes, the Collaboration Server and endpoint H.264 High Profile capability is listed
in the Participant Properties - H.245 and SDP tabs for H.323 participants and SIP participants
respectively.
For more information see Monitoring IP Participants.
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● H.264 High Profile is not supported with Video Preview
HD1080p60 Resolution Guidelines
HD1080p60 resolution is supported in Continuous Presence (CP) mode:
● Symmetrically: The Collaboration Server receives and sends HD1080p60.
● With Video Quality mode set to Motion.
● For H.323 and SIP participants.
● At line rates up to 4Mbps.
RMX 1800 with no DSP cards only supports HD1080p60 symmetric video when Video Quality is set
as Motion in the conference Profile.
HD1080p60 resolution is supported in Video Switching (VSW) mode:
● At bit rates of up to 6Mbps.
● HD1080p60 is supported symmetrically: The RMX receives and sends HD1080p60.
HD1080p60 resolution is not supported:
● For ISDN participants.
● With RTV video protocol.
HD1080p60 resolution is not supported symmetrically:
● In 1X1 layouts, including 1X1 layouts in Telepresence Mode. Instead, the Collaboration Server
transmits the current speaker endpoint’s video resolution and frames per second.
● In TIP environments.
In Telepresence environments the RMX receives HD720p60 and sends HD1080p60 to all endpoints except
for those with 1x1 Video Layouts, which receive the same resolution and frame rate from the RMX as they
send. TIP endpoints are not supported.
PAL endpoints are supported at a frame rate of 50 fps.
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Video Protocols and Resolution Configuration for CP Conferencing
CP Conferencing with H.263 4CIF
The video resolution of 4CIF in H.263 endpoints is only supported for conferences in which the video quality
is set to sharpness and for line rates of 384 Kbps to 1920 Kbps as shown in the following table.
Video Quality vs. Line Rate
Video Quality
Endpoint
Line Rate Kbps
Motion
Sharpness
Resolution
Frame Rate
Resolution
Frame Rate
128
QCIF
30
CIF
30
256
CIF
30
CIF
30
384 - 1920+
CIF
30
4CIF
15
128
CIF
30
256
CIF
30
384 - 1920+
4CIF
15
The Collaboration Server Web Client supports monitoring of H.263 4CIF information. The H.245 or SDP tab
includes the additional information.
The creation of a new H.263 4CIF slide is supported in the IVR Service in addition to the current H.263 IVR
slide. If users utilize the default Polycom slides that are delivered with the Collaboration Server, the slide’s
resolution will be as defined in the profile, i.e. SD, HD, CIF, etc.
For more information see Accessing the Resolution Configuration Dialog Box.
H.263 4CIF Guidelines
● H.263 4CIF is supported with H.323, and SIP and ISDN connection endpoints.
● H.263 4CIF is supported in CP mode only.
● Click&View is supported in H.263 4CIF.
● AES encryption is supported with H.263 4CIF.
● Recording of H.263 4CIF conferences is supported by the Polycom® RealPresence® Media Suite
and other devices.
● All video layouts are supported in H.263 4CIF, except 1x1 layout. In a 1x1 layout, the resolution will
be CIF.
● H.239 is supported in H.263 4CIF and is based on the same bandwidth decision matrix as for HD.
Note: Resource Reference
For information about Resource Usage see Resource Capacity
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The CP Resolution Decision Matrix
All the CP resolution options and settings are based on a decision matrix which matches video resolutions
to connection line rates, with the aim of providing the best balance between resource usage and video
quality at any given line rate.
The following factors affect the decision matrices:
● The Media card(s) installed in the system affect the number of video resources used for each video
resolution and frame rate, the supported video protocols and the maximum resolution that can be
used by the Collaboration Server.
● The used video protocol: H.264 base Profile or H.264 High Profile. The H.264 High Profile maintains
the Video quality at bit rates that are up to 50% lower than previously required. For example, a 512
kbps call will have the video quality of a 1Mbps HD call while a 1Mbps HD call has higher video quality
at the same (1Mbps) bit rate.
● A different decision matrix is used for Motion and Sharpness as the quality requirements are different.
The system is shipped with three pre-defined settings of the decision matrix for H.264 Base Profile and three
pre-defined settings of the decision matrix for H.264 High Profile with Motion and Sharpness video quality
for each of the following resource management schemes:
● Resource-Quality Balanced (default)
A balance between video quality and resource usage. This is the only available resolution
configuration in version 6.0.x and earlier.
● Resource Optimized
System resource usage is optimized by allowing high resolution connections only at high line rates
and may result in lower video resolutions (in comparison to other resolution configurations) for some
line rates. This option allows to save MCU resources and increase the number of participant
connections.
● Video Quality Optimized
Video is optimized through higher resolution connections at lower line rates increasing the resource
usage at lower line rates. This may decrease the number of participant connections.
Video Resource Usage
Video resource usage is dependent on the participant’s line rate, resolution and Video Quality settings.
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Default Minimum Threshold Line Rates and Resource
Usage Summary
The following Table summarizes the Default Minimum Threshold Line Rates and Video Resource usage for
each of the pre-defined optimization settings for each Resolution, H.264 Profile, Video Quality setting
(Sharpness and Motion).
Default Minimum Threshold Line Rates and Video Resource Usage
Optimization Mode
Resolution
HD1080p60
HD1080p30
HD720p60
HD720p30
SD 60
SD 30
CIF 60
CIF 30
Profile
Balanced
Resource
Video Quality
Sharpness
Motion
Sharpness
Motion
Sharpness
Motion
Default
kbps
High
2560
2560
3584
3584
2048
1728
Base
3584
3584
4096
4096
3584
3072
Default
kbps
High
1536
2560
1024
Base
2048
2560
1728
Default
kbps
High
1280
2560
832
Base
1920
1920
1280
Default
kbps
High
832
1280
512
Base
1024
1536
832
Default
kbps
High
768
1024
512
Base
1024
1024
768
Default
kbps
High
256
384
256
Base
256
384
256
Default
kbps
High
256
384
256
Base
384
384
256
Default
kbps
High
64
64
64
64
64
64
Base
64
64
64
64
64
64
Maximum Negotiated Line Rate for Resolutions
Line Rate
Resolution
Up to 1M
CIF60, SD30 and below
Up to 4M
SD60, HD720p30
Up to 6M
HD720p60, HD1080p30, HD1080p60
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Resolution Configuration for CP Conferences
The Resolution Configuration dialog box enables you to override the default video resolution decision
matrix, effectively creating your own decision matrix. The minimum threshold line rates at which endpoints
are connected at the various video resolutions can be optimized by adjusting the resolution sliders.
System resource usage is also affected by the Resolution Configuration settings.
For more information see Video Resource Usage and Default Minimum Threshold Line Rates and Resource
Usage Summary.
Guidelines
● Resolution Slider settings affect all Continuous Presence (CP) conferences running on the
Collaboration Server. Video Switched conferences are not affected.
● A system restart is not needed after changing the Resolution Slider settings.
● Resolution Slider settings cannot be changed if there are ongoing conferences running on the
Collaboration Server. The displayed sliders and the resolutions change according the MCU type.
Modifying the Resolution Configuration
The Resolution Configuration dialog box is accessed by clicking Setup > Resolution Configuration in
the Collaboration Server Setup menu.
Clicking the Detailed Configuration button toggles the display of the Detailed Configuration pane, which
displays sliders for modifying minimum connection threshold line rates for endpoints that support H.264
Base Profile or High Profile. The Detailed Configuration pane can also be opened by clicking the Manual
radio button in the Resolution Configuration pane.
Sharpness and Motion settings are accessed by clicking the Sharpness and Motion tabs when the
Detailed Configuration is open.
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Resolution Configuration
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Resolution Configuration - Basic
The Resolution Configuration basic dialog box contains the following panes:
● Max CP Resolution Pane
● Resolution Configuration Pane
Maximum CP Resolution Pane
The Collaboration Server can be set to one of the following Maximum CP Resolutions:
● HD 1080p60
● HD 1080p30
● HD 720p30
● SD 30
● CIF 30
Limiting Maximum Resolution
Before a selection is made in this pane, the Maximum CP Resolution of the system is determined by the
MAX_CP_RESOLUTION System Flag.
The MAX_CP_RESOLUTION flag value is applied to the system during First Time Power-on and after a
system upgrade.
The default flag value is HD1080 setting the Maximum CP Resolution value in the Resolution
Configuration dialog box to 1080p60.
All subsequent changes to the Maximum CP Resolution of the system are made by selections in this pane.
The Maximum Resolution can further be limited per conference or per participant endpoint.
The Maximum Conference Resolution, can be limited via the Profile - Video Quality dialog box. For more
information see New AVC CP Profile - Video Quality Parameters and New SVC Profile - Video Quality
Parameters.
The Maximum Resolution can further be limited per participant endpoint via the Participant - Properties
dialog box. For more information see New Participant - Advanced.
Resolution Configuration Pane
The user can select from 3 pre-defined Resolution Configurations or select a manual Resolution Slider
adjustment mode. The pre-defined settings can be accepted without modification or be used as the basis
for manual fine tuning of resolution settings by the administrator.
The Manual radio button is automatically selected if any changes are made to the Resolution Sliders.
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The Resolution Configurations are:
● Resource-Quality Balanced (default)
A balance between the optimized video quality and optimized resource usage.
Use this option:
• When the priority is to maintain a balance between resource usage and video quality.
• When it is necessary to maintain backward compatibility with previous versions.
• When working with Polycom DMA/RealPresence Resource Manager.
The Balanced settings are described in the section: Default Minimum Threshold Line Rates and
Resource Usage Summary.
● Resource Optimized
System resource usage is optimized by allowing high resolution connections only at high line rates
and may result in lower video resolutions (in comparison to other resolution configurations) for some
line rates.
Use this option when the priority is to save MCU resources and increase the number of participant
connections.
The Resource Optimized settings are described in the section: Default Minimum Threshold Line
Rates and Resource Usage Summary.
● Video Quality Optimized
Video is optimized through higher resolution connections at lower line rates increasing the resource
usage at lower line rates. This may decrease the number of participant connections.
Use this option when the priority is to use higher video resolutions while decreasing the number of
participant connections.
fr
The Video Quality Optimized settings are described in the section: Default Minimum Threshold Line
Rates and Resource Usage Summary.
● Manual
Manually adjusting the sliders to accommodate local conferencing requirements.
Resolution Configuration - Detailed
The Resolution Configuration -Detailed dialog box contains the following panes:
● Sharpness Resolution sliders
● Motion Resolution sliders
Sharpness and Motion
Sharpness and Motion are Video Quality settings that are selected per conference and are defined in the
conference Profile. A conference with Sharpness selected in its Profile uses the Sharpness settings of the
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Resolution Configuration and a conference with Motion selected in its Profile uses the Motion settings of the
Resolution Configuration dialog box.
The Sharpness and Motion tabs in the Resolution Configuration dialog box allow you to view and modify
Resolution Configuration settings for conferences with either Video Quality setting.
The Sharpness and Motion tabs include separate settings for Base Profile and High Profile as the
Collaboration Server uses two decision matrices (Base Profile, High Profile) to enable endpoints to connect
according to their capabilities.
H.264 High Profile allows higher quality video to be transmitted at lower bit rates. However, setting minimum
bit rate thresholds that are lower than the default may affect the video quality of endpoints that do not
support the H.264 High Profile.
Resolution Configuration Sliders
The Detailed Configuration dialog box allows you to configure minimum connection threshold bit rates for
endpoints that support H.264 High Profile and those that do not support H.264 High Profile by using the
following slider panes:
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● Base Profile - Endpoints that do not support H.264 High Profile connect at these minimum threshold
bit rates.
In Sharpness mode
In Motion mode
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● High Profile - Endpoints that support H.264 High Profile connect at these minimum threshold bit
rates.
In Sharpness mode
In Motion mode
Although the default minimum threshold bit rates provide acceptable video quality, the use of higher bit rates
usually results in better video quality but will require more resources.
The Base Profile and High Profile sliders operate in the same manner in Sharpness and in Motion.
Video Resource usage is affected by the Resolution Configuration settings. The lower the line rate threshold
set for a certain resolution the more resources will be used to connect that participant (as a higher resolution
will be used).
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Flag Settings
Setting the Maximum CP Resolution for Conferencing
The MAX_CP_RESOLUTION flag value is applied to the system during First-time Power-up and after a
system upgrade. The default value is HD1080.
All subsequent changes to the Maximum CP Resolution of the system are made by selections in the Max
CP Resolution pane of the Resolution Configuration dialog box.
The Maximum CP Resolution of the Collaboration Server can be set to one of the following resolutions:
● HD 1080p60
● HD 1080p30
● HD 720p30
● SD 30
● CIF 30
Minimum Frame Rate Threshold for SD Resolution
The MINIMUM_FRAME_RATE_THRESHOLD_FOR_SD System Flag can be added and set to prevent low
quality, low frame rate video from being sent to endpoints by ensuring that an SD channel is not opened at
frame rates below the specified value. For more information see Modifying System Flags.
Additional Video Resolutions
The following higher video quality resolutions are available:
MPMRx Supported Resolutions
Resolution Name
Resolution in Pixels
Frames per Second
CIF
352 x 288
50
WCIF
512 x 288
50
WSD
848 x 480
50
W4CIF
1024 x 576
30
HD 720p
1280 x 720
60
HD 1080p
1920 x 1080
30
HD 1080p
1920 x 1080
60
The video resolution transmitted to any endpoint is determined by the endpoint’s capabilities, the
conference line rate, the Conference Profile’s Motion and Sharpness setting.
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w448p Resolution
For improved interoperability with Tandberg MXP 990/3000 endpoints, the appropriate System Flag settings
will force the Collaboration Server to send w448p (768x448 pixels) at 25fps as a replacement resolution for
WSD15 (848x480) and SD15 (720x576 pixels).
Guidelines
● The w448p resolution is supported:
 In CP mode.
 At conference line rates of 384kbps and 512kbps.
 With H.323, SIP, and ISDN endpoints.
H.323 endpoints must identify themselves as Tandberg MXP during capabilities exchange.
 In all Video Layouts.
 In 1x1 Layout:
 When Video Clarity is Off, the Collaboration Server transmits the same resolution as it
receives.
 When Video Clarity is On, the Collaboration Server changes the transmitted resolution to
w448p.
For more information see Video Clarity™.
● Resource consumption for the w448p resolution is the same as for SD and WSD resolutions.
The following table lists the video outputs from the Collaboration Server to the Tandberg Endpoints for both
16:9 Aspect Ratio when the w448p resolution is enabled.
Video Output to Tandberg Endpoints- Aspect Ratio 16:9
Video Quality
Tandberg
Collaboration
Server
Line
Rate
Kbps
H.323
SIP
Motion
Sharpness
H.323
SIP
Sharpness
*
Network
Environment
Sharpness
Resolution
Frame
Rate
fps
Resolutio
n
Frame
Rate fps
Tandberg to
Collaboration Server
Collaboration Server
to Tandberg
384
512x288
30
768x448
25
512
768x448
30
768x448
25
384
1024x576
15
768x448
25
512
1024x576
15
768x448
25
* It is recommend to set the endpoint to Motion to ensure the transmission of the higher frame rates of 25fps/30fps
to the Collaboration Server.
The following table list the video outputs from the Collaboration Server to the Tandberg Endpoints for 4:3
Aspect Ratio when the w448p resolution is enabled.
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Video Output to Tandberg Endpoints - Aspect Ratio 4:3
Video Quality
Tandberg
Collaboration
Server
Line
Rate
Kbps
H.323
SIP
ISDN
Motion
Sharpness
H.323
SIP
ISDN
Sharpness*
Network
Environment
Sharpness
Resolution
Frame
Rate
fps
Resolution
Frame
Rate
fps
Tandberg to
Collaboration Server
Collaboration Server
to Tandberg
384
576x448 ‡
25
768x448
25
512
576x448 ‡
25
768x448
25
384
4CIF
15
768x448
25
512
4CIF
15
768x448
25
* It is recommend to set the endpoint to Motion to ensure the transmission of the higher frame rates of 25fps/30fps
to the Collaboration Server.
‡ MXP 990/3000 endpoints transmit 576x448 pixels. Other MXP endpoints may transmit other
resolutions eg. CIF.
Content
Sharing and receiving Content is supported.
Bandwidth allocated to the Content channel during Content sharing may cause the video resolution to be
decreased as from w448p to w288p.
When Content sharing stops and the full bandwidth becomes available, video resumes at the previous
w448p resolution.
For more information see the Additional Conferencing Information.
Packet Loss Compensation
If there is Packet Loss in the network and Dynamic Bandwidth Allocation (DBA) is activated, allocating
bandwidth for Lost Packet Recovery, video resolution decreases from w448p to w288p.
When Packet Loss ceases and DBA no longer needs to allocate bandwidth for Lost Packet Recovery, the
full bandwidth becomes available and video resumes at the previous w448p resolution.
For more information see the Packet Loss Compensation - LPR and DBA.
Enabling Support of the w448p Resolution
w448p resolution support for Tandberg endpoints requires setting of the following entities:
● Tandberg endpoint
● Collaboration Server flags
● Collaboration Server Conference Profile
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Collaboration Server System Flag Settings
On the Collaboration Server, the Video Quality field in the New Profile - Video Quality dialog box must be
set to Sharpness.
For more information see Defining New Profiles.
Additional Intermediate Video Resolutions
Two higher quality, intermediate video resolutions replace the transmission of CIF (352 x 288 pixels) or SIF
(352 x 240 pixels) resolutions to endpoints that have capabilities between:
● CIF (352 x 288 pixels) and 4CIF (704 x 576 pixels) – the resolution transmitted to these endpoints is
432 x 336 pixels.
● SIF (352 x 240 pixels) and 4SIF (704 x 480 pixels) – the resolution transmitted to these endpoints is
480 x 352 pixels.
The frame rates (depending on the endpoint’s capability) for both intermediate resolutions are 25 or 30 fps.
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Content such as graphics, presentations, documents, or live video can be shared with conference
participants.
Content sharing architecture is comprised of various aspects:
● Content Control Protocols - H.239 (for H.323 / ISDN), BFCP (for SIP), or People+Content
(Polycom’s protocol used for CP conferences prior to H.239 creation)
● Content Media Protocols - H.263 (AVC only), H.264 (all conferencing modes), or TIP (Cicso’s
proprietary protocol for TelePresence endpoints).
● Content Transmission Methods - Content Video Switching and Multiple Content Resolutions.
● Content Settings - Graphics, Hi-res Graphics, Live Video, or Customized Content Rate.
Note: Legacy Content
Endpoints without the content capabilities matching the conference content sharing
requirements, can connect to the conference but cannot share or view content in the
content channel. Depending on Legacy definitions, they can view content in the people
video layout (see Sending Content to Legacy Endpoints).
Content Control Protocols
Endpoints wishing to share content, request the Content Token from the MCU (in cascaded environments,
the Master MCU). The MCU uses the control protocol to grant the Content Token to the requesting endpoint
(unless in Exclusive Content Mode).
Guidelines for Controlling Content
● Only the Content Token owner may send content.
● Content Token ownership is valid until:
 A new endpoint requests token ownership (unless in Exclusive Content Mode), even when the
endpoint is connected via a cascaded MCU. For more information see Guidelines for Content
Snatching in Cascaded Conferences.
 The owner relinquishes it.
 The Content Token owner endpoint disconnects from the conference.
 It is canceled by the MCU user.
Supported Content Control Protocols
Polycom supports content sharing using one of the following content control protocols:
● H.239 - For H.323 or ISDN participants
● BFCP - For SIP participants (over TCP or UDP)
● TIP Auto-Collaboration (CISCO TIP participants) - TIP Auto-Collaboration for H.264 content sharing
(see TIP Compatibility - Prefer TIP)
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● People+Content - Polycom’s proprietary content control protocol; for H.323 and ISDN participants
Endpoints supporting the content control protocols above can share content within the same conference.
Note: Content Sharing with Endpoints Limited to H.263 in TIP Scenarios
Since TIP compatible conferences (see TIP Compatibility - Prefer TIP) use H.264 as content
media protocol, endpoints supporting only H.263 are considered Legacy content endpoints (see
Sending Content to Legacy Endpoints), regardless of the control protocol they are using.
Content Sharing Using H.239 Protocol
This protocol is used by H.323 and ISDN endpoints.
The H.239 protocol allows compliant endpoints to share content stream simultaneously with video.
Cascaded links declare H.239 capabilities, and are supported in Star and MIH cascading topologies. For
more details, see Cascading Conferences - H.239-enabled MIH Topology.
Content sharing is supported only during conferencing (not while in Entry Queue).
Content Sharing Using BFCP Protocol
This protocol is used by SIP endpoints.
The MCU supports BFCP over either TCP or UDP, which enables the MCU to share content with both SIP
client types.
Guidelines for Using SIP BFCP Content
For SIP clients supporting BFCP/TCP or BFCP/UDP:
● BFCP content is not supported over SIP links (in Gateway and cascading scenarios). Therefore, in
cascading environment, the cascaded link must be defined as H.323 for content to be shared.
● BFCP/UDP is supported in both IPv4 and IPv6 addressing modes. BFCP/TCP is supported only in
IPv4 addressing mode.
● Note that Lync endpoints, though considered as SIP endpoints, do not use BFCP to share content,
and use a Microsoft proprietary protocol for that purpose. There are two options to bypass that
limitation:
 Polycom® ContentConnect™ plug-in (see Content Sharing via Polycom® ContentConnect™
Plug-in for Microsoft Lync Clients). This is the preferred option, and is used whenever the
Polycom® ContentConnect™ plug-in is currently used.
 Treating Lync endpoints as Legacy, thus viewing content via the people video layout (see Sending
Content to Legacy Endpoints).
● BFCP utilizes an unsecured channel (port 60002/TCP) even when SIP TLS is enabled. If security is
of higher priority than SIP content sharing, SIP/BFCP can be disabled by manually adding the
ENABLE_SIP_PEOPLE_ PLUS_CONTENT system flag, and setting its value to NO.
● SIP and BFCP capabilities are by default declared to all endpoints.
Capabilities declaration is controlled by the ENABLE_SIP_PPC_FOR_ALL_USER_AGENT system
flag, whose default value is YES, meaning BFCP capability is declared to all vendors’ endpoints.
When set to NO, the MCU declares SIP over BFCP capabilities only to Polycom and Avaya
endpoints. Note that a SIP proxy might remove the agent information, thus preventing the capability
declaration to Polycom and Avaya endpoints as well.
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● Set ENABLE_FLOW_CONTROL_REINVITE system flag to NO when SIP BFCP is enabled.
● If the system flags mentioned above do not exist in the system, they must be manually added (see
Modifying System Flags).
Content Sharing via Polycom® ContentConnect™ Plug-in for Microsoft Lync Clients
From version 8.1, Polycom® ContentConnect™ (Content Sharing Suite) plug-in for Lync clients allows Lync
clients to receive and send content via SIP BFCP, without having to use the people video layout.
The Polycom® ContentConnect™ plug-in invokes a separate call for content with or without video. This call
may be invoked per a Lync client.
Microsoft Lync is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
Dial-in / Dial-out Connections
For dial-out and dial-in connections to and from SIP Clients, BFCP/UDP or BFCP/TCP protocol can be given
priority by adding the adding the SIP_BFCP_DIAL_OUT_MODE and SIP_BFCP_DIAL_IN_MODE System
Flag to system.cfg and setting their values to UDP, TCP or AUTO.
The Collaboration Server’s content sharing, as determined by the system flags settings and SIP client
capabilities, is summarized in the following table:
System Flags - SIP_BFCP_DIAL_OUT_MODE / SIP_BFCP_DIAL_IN_MODE
SIP Client: BFCP Support
Flag Value
UDP
AUTO (Default)
BFCP/UDP selected as
content sharing protocol
UDP
TCP
TCP
BFCP/TCP selected as
content sharing protocol
UDP and TCP
BFCP/UDP selected as
content sharing protocol
Cannot share content
Cannot share content
BFCP/TCP selected as content sharing protocol
BFCP support in dial-in Connections
● The MCU shares content with dial-in SIP clients according to their preferred BFCP protocol.
● SIP clients connected as audio-only cannot share content.
Content Sharing Using People+Content Protocol
People+Content utilizes a different content control protocol, and is Polycom’s proprietary protocol used prior
to H.239. This protocol is supported in CP conferences, and is applicable for H.323 and ISDN endpoints.
Guidelines for Content Sharing Using People+Content Protocol
● If an endpoint supports both H.239 and People+Content protocols, H.239 is selected as the preferred
communication protocol.
● H.263 and H.263 and H.264 Auto-selection are the only supported content media protocols usage
modes (see MCU Usage Modes of Content Protocols).
● People+Content is enabled by default. It can be disabled for all conferences and endpoints by
manually adding the ENABLE_EPC System Flag, and setting its value to NO (default value is YES).
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● Endpoints supporting People+Content (for example, FX endpoints) may require a different signaling
protocol. For these endpoints, manually add the System Flag CS_ENABLE_EPC, and set its value
to YES (default value is NO).
Content Media Protocols
The RealPresence Collaboration Server transmits content using the following content media protocols:
● H.263 (Annex T) - Base profile only.
H.263 is the only media protocol supported by the MCU for content sharing in ISDN endpoints.
● H.264 - Base and high profiles support is according to the table below:
High Profile is Dependence on Platform
High profile is not supported on Collaboration Server 1800 with no DSP cards.
H.264 AVC (non-TIP) conferences
Content Profile
1800
RMX 2000/4000
Base profile


High profile


For single MCU conferences, the MCU determines the media protocol and high/base profile by
applying the Highest Common principle (see Highest Common). In cascaded environments,
high/base profile requires user configuration (see Sharing Content in Cascaded Environments).
.
H.264 Supported Resolutions for AVC (non-TIP) Conferences
Conference
Resolution
1800
RMX 2000/4000
Multiple Content
Resolution
720p5



720p30



1080p15



1080p30 *



1080p60



● TIP content - The supported TIP content rate is XGA, 5 fps, 512 Kbps, base profile (see Appendix I
- Polycom Open Collaboration Network (POCN)).
Should the minimal TIP content allocation be less than 512 Kbps, content channel will not be opened.
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Content Transmission Methods
There are two methods used for content transmission:
● Content Video Switching
● Multiple Content Resolutions - Content Transcoding
Content Transcoding is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
Note: Content in RealConnect Environments
In conferences using Polycom® RealConnect™ technology between remotes premises, content is
shared using Polycom® ContentConnect for Microsoft environments.
In this scenario, content sharing requires one of the following:
• Content transmission method is set to Multiple Content Resolutions, with the H.263 check-box
selected.
• H.264 Cascade Optimized (see MCU Usage Modes of Content Protocols)
• H.264 HD (likewise)
Content Video Switching
When using this method, content sent by endpoints is transmitted as-is to all content-capable participants.
The content rate is determined using one of the following methods:
● Highest Common
● Fixed Rate
Note: LPR
LPR has no effect on content rate, though due to the increased number of frames, video rate is
decreased.
Highest Common
Note: Method for single MCU conferences only
This method is applied for single MCU conferences.
The Highest Common method is applied in a few levels:
● Within the same protocol - The highest common content rate
● Between protocols - H.263 and H.264
● In H.264 - Base and high profile
In this method, the content rate is negotiated to highest common capabilities supported by the endpoints
connected to the conference. Therefore, if the conference includes participants supporting both lower and
higher content capabilities, the content is sent to all endpoints using the lowest endpoint capabilities.
Note, that for H.263 content, the MCU negotiates the content rate, and the endpoint sharing the content
determines the resolution and frame rate. However, for H.264 content, the MCU negotiates profile type,
content rate, resolution and frame rate, and the endpoint sharing the content may use lower content
parameters.
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When a new endpoint with lower content capabilities joins while content is shared, content sharing
parameters are downgraded to this endpoint capabilities:
● If downgrade affects the protocol (from H.264 to H.263) or profile type (from high to base), content
sharing is terminated, protocol automatically downgrades to H.263, or in H.264, profile downgrades
to base profile, and content sharing should be manually resumed.
● If downgrade affects only the content rate, resolution or frame rate (in H.264), the relevant content
parameters automatically downgrade during content sharing.
Once this endpoint leaves the conference during content sharing, no change occurs in the content
parameters.
Highest Common Calculation Depending on Content Settings
For Graphics, Hi-res Graphics and Live Video (see Content Settings), the highest common content is
calculated for the conference each time an endpoint connects. Highest Common content bit rate is
calculated using:
● For H.263 - Line rate
● For H.264 - Line rate, resolution, and frame rate.
Therefore, when endpoints with lower capabilities connect to an ongoing conference, content parameters
are re-calculated and decreased accordingly.
During content sharing, the MCU does not permit endpoints to increase their content bit rate, only their
content resolution.
For more information, see H.264 Highest Common Content Rates Tables. For information on minimum
content rates as determined by system flags, see H.264 HD System Flags.
Fixed Rate
Note: Fixed Content Rates as Applied in Cascading Scenarios
This method must be used for cascaded conferences.
In this method the content rate is fixed, and endpoints not supporting this rate do not receive content over
the content channel. Endpoints whose capabilities are too low, can only view content over the people video
layout (see Sending Content to Legacy Endpoints).
For H.263 cascaded conferences, the MCU automatically uses a fixed rate (once a cascaded link is
detected) according to the content rates described in Highest Common and Fixed Content Bit Rate
Allocation for H.263.
For H.264 cascaded conferences, the MCU uses a fixed rate according to the content rates described in
H.264 Cascade Optimized (Fixed) Content Rates Tables, and the high/base profile configured by the user.
Multiple Content Resolutions
Note: Naming Convention
This method is often related to as content transcoding.
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In this method, the content is shared in multiple streams, one for each video protocol: H.263 (optional) and
H.264HD (mandatory). Separate video resources are used for processing the content for each of the
required content streams. The MCU then applies the Highest-Common principle for sharing content with
each group of endpoints.
In cascading conferences, an additional resource is dedicated for the cascading link(s), the H.264 Cascade
option must be selected, and content is shared over the cascaded link(s) using fixed rates.
As a consequence, using this method allows endpoints with different capabilities to connect/disconnect
without having to repeatedly restart content sharing process.
Guidelines for Sharing Contents using Multiple Content Resolutions
● Content is always provided to H.264 HD endpoints.
● The Send Content to Legacy Endpoints option is set, and cannot be modified.
● Additional resources are allocated to the conference (in addition to resources for the conference
participants) for content processing:
 Resources are allocated only upon first content sharing initiation, but once allocated, they remain
unchanged until conference ends.
 H.264 basic operation requires 1.5 HD video resources for HD720p30 (defined) resolution
conferences, and 2HD video resources for HD1080p15.
 An additional content stream can be sent to H.263 endpoints, in which case a separate resource
(1 HD) is allocated. H.263 supported resolutions are CIF, 4CIF, XGA.
 Content can be sent to cascaded links, in which case, the H.264 Cascade option must be
selected, and an additional video resource (1 HD) is allocated. The links in both MCUs must use
the same content parameters (meaning content rate, resolution, and frame rate).
● When resources are insufficient, Multiple Content Resolutions mode is disabled, and content sharing
reverts to Content Video Switching mode.
 If H.264 Cascade was selected for the conference, the conference uses H.264 Cascade
Optimized as the content protocol.
 If H.264 Cascade was not selected for the conference, H.264 HD protocol is used.
 Endpoints supporting only H.263, are considered as Legacy content endpoints (see Sending
Content to Legacy Endpoints) due to insufficient resources.
● TIP endpoints cannot share content, only view it over the people video layout.
● When AS SIP is enabled for the conference, Multiple Resolutions is selected and cannot be
modified, H.264 is enabled, and both H.263 and H.264 Cascade are disabled. Resources are
allocated on conference beginning. All this enables AS SIP performance in high-traffic environments.
● In the event of a media card malfunction (and reset), all Multiple Content Resolution conferences will
be terminated.
Note: Support Limitations in Content Transcoding Scenarios
When using the Multiple Content Resolutions method, the following are not supported:
• High profile
• 1080p30/60 content resolutions. Only resolutions up to 1080p15 are supported.
• If there is a resource deficiency when creating a conference with Multiple Content Resolutions, the
Collaboration Server will downgrade Content to HD H.264 Video Switched Content. If Multiple
Content Resolutions is configured to support cascaded conferences, the Content setting will
downgrade to H.264 Cascade Optimized, otherwise it will downgrade to HD H.264.
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Content Settings
The Content channel can transmit in one of the following modes:
● Graphics (default) – For standard graphics.
● Hi-res Graphics – Requiring a higher bit rate to increase display quality or highly detailed graphics.
● Live Video – Highest bit rate, for video clips or live video display.
● Customized Content Rate - Allowing manual definition of the Conference Content Rate.
Content sharing consumes a quota of the video rate, depending on the content required quality; the better
the content required quality (such as in Live Video) the less bit rate remains for the video.
For each of the settings, the MCU allocates an approximate percentage (determined by the table in Content
Sharing Reference Tables) of the conference video capability towards content sharing: for Graphics - 33%,
for Hi-res Graphics - 50%, and for Live Video - 66%. However, in actuality this percentage might be lower,
since the endpoint capabilities are also taken into account when making this calculation.
Customized Content Rate
Customized Content Rate functionality may be implemented (see Selecting a Customized Content Rate)
when the content rate automatically calculated by the MCU is unsuitable.
For example, in a cascaded environment, the conference rate must be identical for all links. Yet, capabilities
may vary widely between the cascading conferences (such as one conference supporting 2 Mbps, while the
other - 512kbps).
The rates listed by the MCU are up to 66% of the conference maximum line rate.
Customized Content rate list comprised of single value of 0, means the conference bit rate capability is too
low to share content.
Customized content rate is not applicable to TIP content (when TIP compatibility is defined as Prefer TIP).
MCU Usage Modes of Content Protocols
Depending on the endpoints capabilities, you can determine the content sharing experience by selecting the
appropriate protocol and system behavior from the Content Protocol list:
● H.263
● H.263 & H.264 Auto Selection
● H.264 Cascade Optimized
● H.264 HD
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H.263
Note: Content for ISDN Endpoints Protocol Limitation
Collaboration Servers 2000/4000/1800 support content sharing with ISDN endpoints only over H.263
content media protocol.
In this mode, all endpoints share content using H.263 protocol. Select this option when either most
endpoints support H.263, or to share content over a cascading conference, in which case, the cascading
link should be created prior to participants joining the conference.
The Highest Common principle (see Highest Common) is applied for determination of content parameters
in single MCU scenarios, and a fixed rate (see Fixed Rate) in cascading environments.
H.263 & H.264 Auto Selection
Select this option to share content using a mix of H.263 and H.264 capable endpoints. Until version 7.6
(including), this option is named Up to H.264.
For single MCU scenario, the Highest Common principle (see Highest Common) is applied to determine
content parameters. In cascading environments, an H.263 fixed rate is used (see Fixed Rate) regardless of
endpoint capabilities.
H.264 Cascade Optimized
This content sharing option applies for H.264 content media protocol, and provides fixed content rate (see
Fixed Rate) and resolution according to the conference line rate. It must be used for cascading conferences.
The H.264 Cascade Optimized option maintains content quality, and prevents content refreshes upon
participants connect/disconnect from the conference.
In cascading environments using this option, the cascade link signaling must be H.323.
The H.264 Cascade Optimized option must be used for cascaded conferences sharing H.264 content. The
selected content is defined by the conference parameters:
● Line rate
● Content settings (Graphics, Hi-res Graphics, or Live Video)
● Resolution configuration
● Base / High profile content
For more information see H.264 Cascade Optimized (Fixed) Content Rates Tables.
H.264 HD
Note: H.264 HD Usage
• This is the default mode for AVC CP conferences.
• H.264 HD may be used for content sharing only for a single MCU, but not in cascading conferences.
H.264 Content Sharing Properties
The H.264 HD option should be selected if most endpoints in the conference support H.264 to ensure high
quality content.
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When this protocol option is selected, content minimal bit rate is determined. For more information, see
Setting the Minimum Content Rate for Each Content Quality Setting for H.264 HD.
Content parameters are determined by the Collaboration Server applying the Highest Common principle
(see Highest Common), for H.264 media content protocol only.
Guidelines for Sharing Content Using H.264 HD
● Only endpoints supporting HD H.264 content (at least HD720p5) can share content.
● Base or high profiles are auto-selected based on endpoints and MCU capabilities (see table H.264
AVC (non-TIP) conferences).
Setting the Minimum Content Rate for Each Content Quality Setting for H.264 HD
System flags determine the minimum line rate required for endpoints to share H.264 high quality content for
each of the Content Settings: Graphics, Hi Resolution Graphics and Live Video.
H.264 HD System Flags
Content Settings
Flag Name
Range
Default
Graphics
H264_HD_GRAPHICS_MIN_CONTENT_RATE
0-1536
128
Hi Resolution Graphics
H264_HD_HIGHRES_MIN_CONTENT_RATE
0-1536
256
Live Video
H264_HD_LIVEVIDEO_MIN_CONTENT_RATE
0-1536
384
To change the system flag value, the flag must be first manually added to the System Configuration (see
Modifying System Flags.
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Content Sharing Related Issues
Sharing Content in Cascaded Environments
In cascaded environments, content must be shared using the same content rate, and in H.264 conferences,
the same resolution and frame rate as well.
Sharing content has different constraints and guidelines over each of the signaling protocols:
● Over H.323 cascaded links  Fixed content rate is used.
 In cascading environments with non-Polycom MCUs, the Polycom MCU must be defined as
Master
 For H.263 cascading links:
 The cascading link must be created before connecting the participants.
 The cascaded link’s Master/Slave role must be determined, in topologies including more than
two MCUs.
 The value of the content-related system flag H263_ANNEX_T must be identical in both Master
and Slave MCUs.
 For H.264 cascading links, the following should be determined in advance:
 The cascaded link’s Master/Slave role must be configured for the link’s participants.
 The H.264 Cascade Optimized content protocol must be selected.
 High/Base profile must be configured identically between Master and Slave. Endpoints with
lower capabilities cannot share content, and can only view content on the people video layout
(see Sending Content to Legacy Endpoints).
● Over ISDN - For information on sharing content over ISDN cascaded links, see Basic Cascading
Using ISDN Cascaded Link.
● Over SIP - No content sharing is supported over SIP cascaded links.
From version 8.6 and on, content can be snatched in cascaded environments, that is the content token
ownership can be taken/snatched by another participant wishing to share content where content is currently
shared by another participant.
This capability depends on the conference configuration (Exclusive Content Mode is off), and the
ENABLE_CONTENT_SNATCH_OVER_CASCADE system flag (see
ENABLE_CONTENT_SNATCH_OVER_CASCADE).
Guidelines for Content Snatching in Cascaded Conferences
● Content Snatching is available only over anH.323 cascaded link. It is not applicable to either SIP
(BFCP) or PSTN.
● All the MCUs within the (H.323) cascading topology should:
 Use RealPresence Collaboration Server version 8.6 and up.
 Have Exclusive Content Mode turned off (meaning, Exclusive Content Mode check-box in
conference profile Properties > Advanced tab should be cleared).
 Have the system flag ENABLE_CONTENT_SNATCH_OVER_CASCADE value set to YES.
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● Gateway calls are implemented as cascaded links, typically SIP, in which case, content cannot be
shared via the content channel. It may be viewed over the people video layout, depending on Legacy
configuration (see Sending Content to Legacy Endpoints).
Note: Snatching Content Token Limitations Inapplicability Scenario
When a Master MCU is not in Exclusive Content Mode, another participant in the same MCU may
snatch the content, even if the value of the
ENABLE_CONTENT_SNATCH_OVER_CASCADE system flag is NO.
Sending Content to Legacy Endpoints
Note: Legacy Content Limitation to AVC Endpoints
The Content to Legacy feature is applicable for CP conferences, and in mixed CP and SVC
conferences, for AVC endpoints only.
The Collaboration Server can be configured to send content to endpoints not supporting the conference
content parameters (legacy endpoints) over the people video layout, thus allowing the participants to view
content. However, these endpoints cannot share content.
Guidelines for Sending Content to Legacy Endpoints
● A separate HD video resource is allocated to the conference for content sending to legacy endpoints.
Allocation is performed only once a legacy content endpoint is connected to the conference, and a
content session is initiated and transmitted via the people video layout. Once the resource is
allocated, it remains allocated to the conference until its end.
If the system cannot allocate the resource required for sending the content, the conference status
changes to Content Resource Deficiency, and content cannot be sent to the legacy endpoints.
● Endpoints receiving content via the people video layout, use the same video protocol and resolution
they use for receiving video.
● Content cannot be sent to Legacy endpoints when Same Layout mode is selected for the conference.
● This option in not supported in Video Switching conferences.
● Content can be sent to Legacy endpoints in gateway calls, depending on content configuration on
Master/Slave MCUs (for example, if the gateway conference is using H.263 content media protocol,
and the conference MCU is using H.264 content media protocol).
● FX endpoints dialing into the MCU, share content via the content channel (People+Content), whereas
FX endpoints dialing out to the MCU, view content via the people video layout.
Content Display on Legacy Endpoints
When content is sent to content legacy endpoints, their video layout automatically changes to the content
layout defined by the system flag LEGACY_EP_CONTENT_DEFAULT_LAYOUT (default layout is
CP_LAYOUT_1P4VER), and the content is shown in the larger/top-left cell. The video layouts of the other
conference participants do not change. (To modify the default layout, see
LEGACY_EP_CONTENT_DEFAULT_LAYOUT Flag.)
The switch to the content layout occurs in Auto Layout, Presentation Mode, Lecture Mode, and when a
layout is selected for the conference.
In Lecture Mode, when content is sent to legacy endpoints, switching to the content layout, results in the
content shown in the lecturer/speaker window, while the lecturer is shown in the second window. If the layout
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contains more than two windows, all other windows are empty. The non-lecturer legacy content participants
see the lecturer in full screen.
When content is stopped, the layout of the legacy participants terminates as well.
Legacy participants can change their layout using Click&View. In such a case, the content is forced to the
larger/top-left cell.
The Collaboration Server user can also change the layout for the legacy content endpoints participants
(selecting personal layout).
When forcing a video participant to the Content window (instead of Content), the Content display can be
restored only by selecting any other video layout.
Sending Content to Legacy Endpoints in Telepresence Mode
To ignore personal layouts during Telepresence conferences (while working with MLA), set the value of the
flag FORCE_LEGACY_EP_CONTENT_LAYOUT_ON_TELEPRESENCE to YES.
If the layout for displaying content in Legacy endpoints include multiples cells, MCU may populate
Telepresence room streams sources in remote cells.
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Exclusive Content Mode
In this mode, the MCU prevents participants other than the current content sharer, from sharing content.
Exclusive content may be limited to the lecturer, as described below.
To modify exclusive content mode by granting (or canceling) of token ownership, see Giving and Canceling
Token Ownership (AVC Participants)).
Guidelines for Sharing Content in Exclusive Content Mode
● Exclusive Content Mode is available in all conferencing modes.
● Exclusive Content Mode is enabled or disabled (system default) by a check box in the Conference
Profile - Advanced tab, or during an ongoing conference using the Conference Properties Advanced tab.
● In Exclusive Content Mode, when the RESTRICT_CONTENT_BROADCAST_TO_LECTURER
system flag is set to:
 NO - The first participant to send content becomes the Content Token holder, and releasing the
Content Token allows other participants to acquire the token, and begin transmitting content.
 YES - Only the designated Lecturer can be the Content Token holder.
● The Exclusive Content Mode check box replaces the EXCLUSIVE_CONTENT_ MODE system flag
used in previous versions to control exclusive content mode for the system.
● In Exclusive Content Mode, an endpoint attempting to send content after another endpoint starts
sending content, is rejected with a momentary content interruption.
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Forcing Other Content Capabilities
The H239_FORCE_CAPABILITIES system flag allows additional control of content sharing:
● When set to NO (default) - The MCU merely verifies the endpoint supports the content protocols:
H.263 or H.264.
● When set to YES - The MCU verifies frame rate, bit rate, resolution, annexes, and all other
parameters of content as declared by an endpoint during the capabilities negotiation phase. If the
endpoint does not support the content capabilities of the MCU, the participant cannot share content
over a dedicated content channel.
Managing Noisy Content Connections
The system can identify participants sending frequent content display refresh requests (usually as a result
of a problematic network connection), which cause frequent refreshing of content display, and degrading of
viewing quality.
When the system identifies such “noisy” participants, it marks them (in the Conference Participants list,
Status column) as Noisy, and automatically suspends these requests to avoid affecting the quality of the
content viewed by the other conference participants.
This process is controlled by the following system flags:
● MAX_INTRA_REQUESTS_PER_INTERVAL_CONTENT The maximum number of refresh (intra) requests per 10-second intervals allowed for an endpoint.
Beyond that number, content sent by this participant is identified as “noisy”, and its refresh requests
are suspended.
Default setting: 3
● MAX_INTRA_SUPPRESSION_DURATION_IN_SECONDS_CONTENT The duration, in seconds, for ignoring the participant’s content display refresh requests.
Default setting: 10
● CONTENT_SPEAKER_INTRA_SUPPRESSION_IN_SECONDS The interval, in seconds, between content refresh (intra) requests sent from the MCU to the content
sender due to refresh requests initiated by other conference participants. Additional refresh requests
received within that interval are deferred to the next interval.
Default setting: 5
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Useful Procedures in Content Sharing
For all MCUs, content sharing parameters are defined in the conference profile Video Quality dialog box.
Defining Content Sharing Parameters for a Conference
The available content options change according to the selected conferencing mode.
To set the content sharing parameters:
» In the Content Video Definition section, set the values for the Content Settings and Protocol as
follows:
Content sharing Options
Field
Description
Content Settings
Select the transmission mode for the Content channel:
• Graphics — Basic mode, intended for normal graphics
• Hi-res Graphics — A higher bit rate intended for high resolution graphic display
• Live Video — Content channel displays live video
• Customized Content Rate - Manual definition of the Conference Content Rate,
mainly for cascading conferences.
For a description of each of these options, see Content Settings.
AS-SIP Content
AS-SIP is a SIP implementation utilizing SIP’s built-in security features.
When selected, content is shared using the Multiple Resolutions mode, and is not
supported in any other content sharing mode.
Multiple Resolutions
Select this check box to enable Multiple Content Resolutions mode, for both H.263
and H.264 content protocols.
When enabled, H.264 is always selected and cannot be modified.
Optional. Select additional protocols:
• H.263 - If the conference includes endpoints with H.263 capabilities.
• H.264 Cascade - If the conference includes cascading links and you want to
define the video settings for content sharing.
For more information, see Sharing Content Using Multiple Content Resolutions
Mode.
Content Protocol
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•
H.263 - Content is shared using the H.263 protocol.
H.263 & H.264 Auto Selection - Content is shared according to conference
participants capabilities.
• H.264 HD (default) - Content is shared using the H.264 HD protocol.
• H.264 Cascade Optimized - Content is shared using the H.264 content protocol
(fixed), and optimized for use in cascaded conferences.
For a detailed description of each of these settings, see Content Media Protocols.
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Content sharing Options
Field
Description
Content Resolution
Select a Content Resolution from the pull-down menu.
The Content Resolutions available for selection are dependent on the content
sharing mode (Highest Common or Multiple Content Resolutions), Line Rate and
Content Settings selected for the conference.
For a full list of Content Resolutions see H.264 Supported Resolutions for AVC
(non-TIP) Conferences.
Note: This field is displayed only when H.264 Cascade Optimized is selected.
Send Content to
Legacy Endpoints
When enabled (default), content is sent to H.323/SIP/ISDN endpoints not supporting
current MCU content control protocol (legacy endpoints) over the people video layout
(see Sending Content to Legacy Endpoints).
H.264 High Profile
This field is displayed only when conferencing mode is VSW (Video Switching), or
the selected Content Protocol is H.264 Cascade Optimized.
In scenarios where endpoints not supporting high profile (such as HDX) are
connected to the conference, it is recommended to clear this check-box to enable
them to share content.
(RMX 1800 /
MPMRx only)
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H.264 Cascade Optimized Content Sharing
When H.264 Cascade Optimized is selected in conferences as the Content Protocol, an additional field,
Content Resolution is displayed in the Content Video Definition pane.
The Content Resolution value determines the fixed resolution and frame rate to be used for content
sharing in cascaded conferences.
The Content Resolutions available for selection are dependent on the Line Rate and Content Settings
selected for the conference.
An additional check box H.264 High Profile (unchecked by default) is displayed in the Video Quality dialog
box provided:
● The MCU is a RealPresence Collaboration Server 1800, or 2000/4000 containing MPMRx cards.
● TIP Compatibility (in the profile’s Advanced tab) is set to None.
The Content Rate drop-down menu list depends on Customized Content Rate being selected as the
Content Setting, and cannot exceed 66% of the conference line rate capability.
For information on the interaction of these parameters for Base Profile and High Profile settings, see H.264
Cascade Optimized (Fixed) Content Rates Tables.
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Selecting a Customized Content Rate
To Select the Customized Content Rate:
1 In the Content Settings list, select Customized Content Rate.
When selected, a drop-down menu of the available conference content rates is displayed. These
content rates vary according to the selected conference Line Rate.
2 Select the required content rate.
Note: Modifying Conference Rate in Discord from Current Customized Content Rate
If Customized Content Rate is already selected along with the content rate, and you attempt to modify
the conference line rate to a value that does not support the selected customized content rate, an
error message is displayed.
Modify the Content Line Rate or Conference Line Rate, or modify the Content Setting.
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Sharing Content in Multiple Content Resolutions Mode
For information on this content sharing mode see Multiple Content Resolutions.
The Multiple Content Resolutions mode can be modified in the conference Profile, in the Video Quality Tab.
To enable Multiple Content Resolutions:
1 Select the Multiple Resolutions check box.
By default, H.264 is always selected, and cannot be modified.
2 Select additional protocols:
 H.263 - If the conference includes H.263 (only) capable endpoints.
 H.264 Cascade - If the conference includes cascading links.
If H.264 Cascade is selected, select the Content Resolution.
3 Click OK.
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Giving and Canceling Token Ownership (AVC Participants)
For information on exclusive content ownership, see Exclusive Content Mode.
To give token ownership:
1 In the Participants list, right-click the AVC-enabled endpoint you wish to define as the exclusive
Content Token owner.
2 Select Change To Content Token Owner in the drop-down menu.
The endpoint receives exclusive ownership of the Content Token, and an indication icon is displayed
in the Role column of the participant’s entry in the Participants list.
Note: Designating Content Sharer during Ongoing Content Sharing
If another participant is currently sharing content, he is requested to release the token, and the
participant selected as the token owner is marked as exclusive. Once that participant shares content,
it cannot be snatched by other participants.
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To cancel token ownership:
1 In the Participants list, right-click the endpoint that currently has Content Token ownership.
2 Select Cancel Content Token Owner in the drop-down menu.
Content Token ownership is cancelled for the endpoint.
Stopping a Content Session
The Collaboration Server user can withdraw the content token from the current holder.
To end the current Content session:
» In the Conferences list pane, right-click the conference icon, and select Abort H.239 Session.
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Content Sharing Reference Tables
Resolutions and Content Rate Reference Tables
Note: Content Rates Clarification
The values in the tables below indicate the maximum negotiated content rate (for both H.263 and
H.264) and resolution/frame rate (only for H.264).
When using Highest Common, endpoints may lower the content parameters, whereas when using
fixed rate, endpoints must comply with the content parameters as determined by the MCU.
The actual content parameters used by the content sharing endpoint are determined by that endpoint,
and may be lower.
H.263 Content Rate Table
The table below describes the content rates for both Highest Common and Fixed content bit rates.
Highest Common and Fixed Content Bit Rate Allocation for H.263
1152
Content
Settings
/ MCU
64
128
256
768
1280
1536
96
192
320
384
512
832
64
64
128
128
64
128
192
64
128
256
3072
1024
1472
1728
1920
2048
2560
3584
4096
6144
*
256
256
384
512
512
512
768
768
1280
1280
256
384
512
512
768
1024
1024
1280
1536
384
512
512
768
1024
1280
1280
1536
1536
Graphics 33%
All
Hi-res Graphics 50%
All
Live Video 66%
All
* This line rate is applicable only for RealPresence Collaboration Server 1800 with 1 or 3 DSP cards, or with MPMRx card(s). In addition, it is
applicable only for non-SVC-enabled conferencing
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H.264 Resolution per Content Rate Tables
The tables below describe the resolution as negotiated by the MCU according to the content rate, for both
base and high profiles.
Maximum Negotiated Resolution and Frame Rate per Content Rate for H.264 Base Profile
Bit Rate Allocated to Content
Channel (Kbps)
Maximum Negotiated Content
Resolution
Frames/Second
64-512
H.264 HD720
5
512-768
H.264 HD720
30
768-1536
H.264 HD1080
15
1536-3072
H.264 HD1080
30 *
3072-4096
H.264 HD1080
60
Maximum Negotiated Resolution and Frame Rate per Content Rate for H.264 High Profile
Bit Rate Allocated to Content
Channel (Kbps)
Maximum Negotiated Content
Resolution
Frames/Second
64-384
H.264 HD720
5
384-512
H.264 HD720
30
512-768
H.264 HD1080
15
768-2048
H.264 HD1080
30 *
2048-4096
H.264 HD1080
60
* For Collaboration Server 1800 with no DSP cards and Collaboration Server 1800 Entry Level, maximum
content resolution is HD1080p30.
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H.264 Highest Common Content Rates Tables
The tables below summarize the Highest Common maximum content rates as negotiated by the MCU.
These tables apply only to single MCU (non-cascading) and non-SVC enabled conferences.
Note: Platforms and Content Transcoding Resolution Limitations
• For Collaboration Server 1800 with no DSP cards and Collaboration Server 1800 Entry Level,
maximum content resolution is HD1080p30.
• For Transcoding, maximum content resolution is HD1080p15
Highest Common Content Bit Rate for H.264 Base Profile
1152
Content
Settings
/ MCU
64
128
256
768
96
192
320
384
512
832
64
64
128
128
256
1280
1536
3072
1024
1472
1728
1920
2048
2560
3584
4096
6144
*
256
384
512
512
512
768
768
1280
1280
Graphics 33%
≤1080p15^
1080p30 **
2048
Hi-res Graphics 50%
≤1080p15^
64
128
192
256
384
512
512
768
1024
1024
1280
1536
1080p30 **
1536
2048
1080p60 †
2048
3072
Live Video 66%
≤1080p15^
1080p30 **
64
128
256
384
512
512
768
1024
1280
1280
1536
1536
2048
2560
1080p60 †
3072
4096
* This line rate is applicable only for RealPresence Collaboration Server 1800 with 1 or 3 DSP cards, or with MPMRx card(s). In addition, it is
applicable only for non-SVC-enabled conferencing
** This resolution is applicable only for RealPresence Collaboration Server 1800 with no DSP cards, or for Entry Level.
† This resolution is applicable only for RealPresence Collaboration Server 1800 with 1 or 3 DSP cards, or with MPMRx card(s).
^ This limitation applies to content transcoding scenarios.
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Note: High Profile Applicability
• High Profile is not applicable for Collaboration Server 1800 with no DSP cards.
• High profile is not applicable in content transcoding scenarios.
Highest Common Content Bit Rate for H.264 High Profile
Content
Settings
/ MCU
64
128
256
768
1152
1920
3072
96
192
320
384
512
832
1024
1280
1472
1536
1728
2048
2560
3584
64
64
128
128
256
256
384
384
512
512
512
768
768
768
768
1280
2048
1280
1280
4096
6144
*
Graphics 33%
≤1080p15^
1080p30 **
1280
1080p60 †
Hi-res Graphics 50%
≤1080p15^
64
128
192
256
384
512
512
512
1080p30 **
768
768
1024
1280
1280
768
768
1024
1280
1536
1080p60 †
2048
2048
3072
1280
1280
Live Video 66%
≤1080p15AVC only ^
64
128
256
384
512
512
768
768
1024
≤1080p15SVC/Mixed
^
256
512
768
1080p30 **
384
768
1024
1024
1280
1280
1280
N/A
1024
1280
1536
2048
1080p60 †
2048
2560
4096
* This line rate is applicable only for non-SVC-enabled conferencing
** This resolution is applicable only for RealPresence Collaboration Server 1800 with no DSP cards, or for Entry Level.
† This resolution is applicable only for RealPresence Collaboration Server 1800 with 1 or 3 DSP cards, or with MPMRx card(s).
^ This limitation applies to content transcoding.
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H.264 Cascade Optimized (Fixed) Content Rates Tables
The tables below summarize the content rates as they are determined by the MCU.
For information on the supported resolutions for each of the MCU types see H.264 Resolution per Content
Rate Tables.
H.264 Cascade Optimized Content Bit Rate for H.264 Base Profile
1472
Cascade
Resolution
64
128
256
832
1536
2048
96
192
320
384
512
768
1024
1152
1280
1728
1920
2560
3072
3584
4096
6144
*
64
64
128
128
256
256
256
256
256
256
512
512
512
512
512
512
512
512
512
512
512
768
768
768
768
1152
1152
1152
Graphics 33%
720p5
720p30
1080p15 ^
1080p30 **
2048
1080p60 **
Hi-res Graphics 50%
720p5
64
128
192
256
384
384
720p30
384
512
512
512
512
512
512
512
512
512
512
512
512
768
768
768
768
768
768
768
768
768
768
1152
1152
2048
3072
1080p15 ^
1080p30 **
1080p60 **
3072
Live Video 66%
720p5
720p30
1080p15 ^
1080p30 **
1080p60 **
64
128
256
384
512
512
768
768
768
768
768
768
768
768
768
512
768
768
768
768
768
768
768
768
768
768
768
768
768
1152
1152
1152
1152
1152
2048
2048
2560
3072
4096
* This line rate is applicable only for RealPresence Collaboration Server 1800 or one with MPMRx card(s), and for non-SVC-enabled conferencing
** These resolutions are applicable only for RealPresence Collaboration Server 1800 or one with MPMRx card(s). For RMX 1800 with no DSP cards
or for Entry Level, maximum content resolution is HD1080p30
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Note: High Profile support
High profile is supported only by RealPresence Collaboration Server 2000/4000, or 1800 with 1 or 3
DSP cards.
H.264 Cascade Optimized Content Bit Rate for H.264 High Profile
Cascade
Resolution
1152
2560
1280
3072
64
128
256
96
192
320
384
512
768
832
1024
1472
1536
1728
1920
2048
3584
4096
6144
*
64
64
128
128
256
256
256
384
384
512
512
512
512
512
512
384
512
512
512
512
768
768
768
512
512
512
512
768
1280
1280
768
1280
2048
Graphics 33%
720p5
720p30
1080p15
1080p30
1080p60
2048
Hi-res Graphics 50%
720p5
64
128
192
256
720p30
384
384
384
512
512
512
512
512
512
512
512
384
384
512
512
512
768
768
768
768
768
768
512
512
768
768
768
768
1280
1280
1280
768
768
768
1024
1024
2048
2048
2048
3072
1080p15
1080p30
1080p60
Live Video 66%
720p5
64
128
256
256
384
512
512
512
512
512
512
512
512
512
512
720p30
512
512
512
768
768
768
768
768
768
768
768
1080p15
512
512
512
768
768
768
1280
1280
1280
1280
1280
768
1024
1024
1280
1280
1280
2048
2048
2560
4096
1080p30
1080p60
* This line rate is applicable only for non-SVC enabled conferencing
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Implementing Media Encryption for
Secured Conferencing
Encryption is available at the conference and participant levels, based on AES 128 (Advanced Encryption
Standard) and is fully H.233/H.234 compliant and the Encryption Key exchange DH 1024-bit
(Diffie-Hellman) standards.
Media Encryption Guidelines
● Encryption is not available in all countries and it is enabled in the MCU license. Contact Polycom
Support to enable it.
● Media encryption is supported in CP, SVC Only and mixed CP and SVC Conferencing Modes.
● Endpoints must support both AES 128 encryption and DH 1024 key exchange standards which are
compliant with H.235 (H.323) to encrypt and to join an encrypted conference.
● The encryption mode of the endpoints is not automatically recognized, therefore the encryption mode
must be set for the conference or the participants (when defined).
● Media Encryption for ISDN/PSTN participants is not supported in cascaded conferences.
● Conference level encryption must be set in the Profile, and cannot be changed once the conference
is running.
● If an endpoint connected to an encrypted conference stops encrypting its media, it is disconnected
from the conference.
● In Cascaded conferences, the link between the cascaded conferences must be encrypted in order to
encrypt the conferences.
● The recording link can be encrypted when recording from an encrypted conference to the Polycom®
RealPresence® Media Suite that is set to encryption. For more information, see Dial Out Recording
Link Encryption.
● Encryption of SIP Media is supported using SRTP (Secured Real-time Transport Protocol) and the
AES key exchange method.
● Encryption of SIP Media requires the encryption of SIP signaling - TLS Transport Layer must be used.
● Encryption of SIP Media is supported in conferences as follows:
 All media channels are encrypted: video, audio and FECC.
 Collaboration Server SRTP implementation complies with Microsoft SRTP implementation.
 LPR is not supported with SRTP.
 The ENABLE_SIRENLPR_SIP_ENCRYPTION System Flag enables the SirenLPR audio
algorithm when using encryption with the SIP protocol. The default value of this flag is NO
meaning SirenLPR is disabled by default for SIP participants in an encrypted conference. To
enable SirenLPR the System Flag must be added to system.cfg and its value set to YES.
 The SEND_SRTP_MKI System Flag enables or disables the inclusion of the MKI field in SRTP
packets sent by the Collaboration Server. The default value of the flag is YES.
Add the flag to system.cfg and set its value set to NO to disable the inclusion of the MKI field in SRTP
packets sent by the Collaboration Server when using endpoints that cannot decrypt SRTP-based
audio and video streams if the MKI (Master Key Identifier) field is included in SRTP packets sent by
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the Collaboration Server. When all conferences on the RMX will not have MS-Lync clients
participating and will have 3rd party endpoints participating. This setting is recommended for
Maximum Security Environments.
Add the flag to system.cfg and set its value set to YES when Microsoft Office Communicator and Lync
Clients. When any conferences on the RMX will have both MS-Lync clients and Polycom endpoints
participating. Some 3rd party endpoints may be unsuccessful in participating in conferences with this
setting.
Polycom endpoints function normally regardless of the setting of this flag.
For more information, see Modifying System Flags.
● In compliance with UC_APL_SEC_0013, the Collaboration Server 2000/4000 supports an additional
Privacy Protocol AES_CM_128_HMAC_SHA1_32, in addition to AES_CM_128_HMAC_SHA1_80.
For more information see Media Encryption and Authentication.
Note: ISDN/PSTN
ISDN/PSTN connections are not supported by Collaboration Server Virtual Edition.
Mixing Encrypted and Non-encrypted Endpoints in one
Conference
Mixing encrypted and non-encrypted endpoints in one conference is possible, based on the Encryption
option Encrypt When Possible in the Conference Profile - Advance dialog box. The behavior is different for
H.323/SIP and ISDN participants.
The option Encrypt When Possible enables the negotiation between the MCU and the endpoints and let
the MCU connect the participants according to their capabilities, where encryption is the preferred setting.
Defined participants that cannot connect encrypted are connected non-encrypted, with the exception of
dial-out SIP participants.
Note: Encrypt when possible
• When the conference encryption is set to Encrypt when possible, SIP dial out participants
whose encryption is set to AUTO can only connect with encryption, otherwise they are
disconnected from the conference.
• In CISCO TIP environments, dial in endpoints that are registered to CUCM can only connect as
non-encrypted when the conference encryption is set to Encrypt when possible as the CUCM
server sends the Invite command without SDP.
• When the conference encryption is set to Encrypt when possible, SIP dial out participants
whose encryption is set to AUTO can only connect with encryption, otherwise they are
disconnected from the conference.
The same system behavior can be applied to undefined participants, depending on the setting of the System
Flag FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE:
● When set to NO and the conference encryption in the Profile is set to Encrypt when possible, both
Encrypted and Non-encrypted undefined participants can connect to the same conferences, where
encryption is the preferred setting.
● When set to YES (default), undefined participants must connect encrypted, otherwise they are
disconnected.
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For defined participants, connection to the conference is decided according to the encryption settings in the
conference Profile, the Defined Participant’s encryption settings.
For undefined participants, connection to the conference is decided according to the encryption settings in
the conference Profile, the System Flag setting and the connecting endpoint’s Media Encryption
capabilities.
Direct Connection to the Conference
The following table summarizes the connection status of participants, based on the encryption settings in
the conference Profile, the Defined Participant’s encryption settings or the System Flag setting for undefined
participants and the connecting endpoint’s Media Encryption capabilities.
Connection of Participants to the Conference based on Encryption Settings
Conference
Encryption
Setting
No Encryption
Encrypt All
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Defined Participant
Encryption
Setting
Connection status
Auto
Connected,
non-encrypted
No
Connected,
non-encrypted
Yes
Connected only if encrypted.
Non-encrypted endpoints are
disconnected as encryption is
forced for the participant.
Auto
Connected, encrypted.
Non-encrypted endpoints are
disconnected
No
Disconnected (cannot be
added to the conference)
Yes
Connected, encrypted
Undefined Participant
Connection Status
*Flag = No
Connection Status
*Flag = YES
Connected
non-encrypted
(Encryption is not
declared by the
Collaboration
Server, therefore the
endpoint does not
use encryption)
Connected
non-encrypted
(Encryption is not
declared by the
Collaboration
Server, therefore the
endpoint does not
use encryption)
Connect only if
encrypted.
Non-encrypted
endpoints are
disconnected
Connect only if
encrypted.
Non-encrypted
endpoints are
disconnected
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Connection of Participants to the Conference based on Encryption Settings
Conference
Encryption
Setting
Encrypt When
Possible
Defined Participant
Encryption
Setting
Auto
Connection status
All defined participants except
dial-out SIP participants:
• Connect encrypted Endpoints with encryption
capabilities.
• Connect non-encrypted endpoints without
encryption capabilities.
Defined dial-out SIP
participant:
• Connect only if encrypted.
• Non-encrypted endpoints
are disconnected.
No
Connected, non-encrypted
Yes
Connected, encrypted
Undefined Participant
Connection Status
*Flag = No
Connection Status
*Flag = YES
Connect encrypted Endpoints with
encryption
capabilities.
Connect
non-encrypted endpoints without
encryption
capabilities
Connect only if
encrypted.
Non-encrypted
endpoints are
disconnected.
* Flag = FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE
Connection to the Entry Queue
An undefined participant connecting to an Entry Queue inherits the encryption characteristics of the Entry
Queue as defined in the Entry Queue’s profile.
Participants can be moved from the Entry Queue and the destination conference if both conferencing
entities have the same Profile settings, i.e. from SVC Only Entry Queue to SVC Only conference and from
mixed CP and SVC Entry Queue to a mixed CP and SVC conference, etc.
The following table summarizes the connection possibilities for a participant that is to be moved from an
Entry Queue to a destination conference for each of the conference Profile and Entry Queue encryption
options.
Connection of Undefined Participants to the Entry Queue Based on Encryption Settings
Entry Queue Encryption
Setting
Undefined Participant Connection to the Entry Queue
*Flag = No
*Flag = YES
No Encryption
Connected, non-encrypted
(Encryption is not declared by the
Collaboration Server, therefore endpoint
does not use encryption)
Connected, non-encrypted
(Encryption is not declared by the
Collaboration Server, therefore endpoint
does not use encryption)
Encrypt All
Connected only if encrypted.
Non-encrypted endpoints are
disconnected
Connected only if encrypted.
Non-encrypted endpoints are
disconnected
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Connection of Undefined Participants to the Entry Queue Based on Encryption Settings
Entry Queue Encryption
Setting
Encrypt When Possible
Undefined Participant Connection to the Entry Queue
*Flag = No
*Flag = YES
Connected encrypted - Endpoints with
encryption capabilities.
Connected non-encrypted - endpoints
without encryption capabilities
Connected only if encrypted.
Non-encrypted endpoints are
disconnected.
* Flag = FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE
Moving from the Entry Queue to Conferences or Between Conferences
Participants can be moved from the Entry Queue and the destination conference if both conferencing
entities have the same Profile settings, i.e. from SVC Only Entry Queue to SVC Only conference and from
mixed CP and SVC Entry Queue to a mixed CP and SVC conference, etc.
When moving participants from the Entry Queue to the destination conference, or when the Collaboration
Server user moves AVC participants from one conference to another (SVC participants cannot be moved
between conferences), the connection rules are similar and they are summarized in the table below:
Moving participants from entry queue to conference / between conferences depending on encryption settings
Destination
Conference
Encryption Setting
Current Participant Encryption Status
Encrypted
*Flag = NO
*Flag = YES
Non-Encrypted
*Flag = NO
*Flag = YES
No Encryption
Move succeeds, connected encrypted
Move succeeds, connected
non-encrypted
Encrypt All
Move succeeds, connected encrypted.
Move fails, disconnected.
Encrypt When Possible
Move succeeds,
connected
encrypted
Move succeeds,
connected
non-encrypted
Move succeeds,
connected
encrypted
Connected only if
endpoint was a
defined participant
in the source
conference.
Otherwise, move
fails.
* Flag = FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE
Recording Link Encryption
Recording Links are treated as regular participants, however the system flag
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF must be set to YES if a
non-encrypted Recording Link is to be allowed to connect to an encrypted conference.
The following table summarizes the connection possibilities for a Recording Link that is to be connected to
a conference for each of the conference profile and Entry Queue encryption options.
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Connections by Recording Link and Conference Encryption Settings
Conference
Profile Setting
Recording Link Connection Status according to flag:
ALLOW_NON_ENCRYPT_RECORDING_ LINK_IN_ENCRYPT_CONF
YES
NO
Encrypt All
Connected encrypted if possible,
otherwise connected non-encrypted.
Connected only if encrypted, otherwise
disconnected
No Encryption
Connected non-encrypted
Connected non-encrypted
Encrypt when possible
Connected encrypted if possible,
otherwise connected non-encrypted.
Connected encrypted if possible,
otherwise connected non-encrypted.
Enabling Media Encryption for a Conference
Media encryption is enabled at three levels:
● MCU level - Setting the Encryption Flags
● Conference level - Enabling Encryption in the Profile
● Participant level - Enabling Encryption at the Participant Level
You must first set the system flags for the MCU before media encryption can be enabled for the conference
and participants.
Setting the Encryption Flags
Enabling the media encryption for the MCU is usually performed once an it is applicable to all conferences
running on the MCU.
To modify the Encryption flags:
1 Click Setup>System Configuration.
The System Flags dialog box opens.
2 Set the
FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE flag
to YES or NO.
3 If recording will be used in encrypted conferences, set the
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF flag to YES or NO.
4 Click OK.
For more information, see Modifying System Flags.
5 Reset the MCU for flag changes to take effect.
Enabling Encryption in the Profile
Encryption for the conference is in the Profile and cannot be changed once the conference is running.
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To enable encryption at the conference level:
» In the Conference Profile Properties – Advanced dialog box, select one of the following Encryption
options:
 Encrypt All - Encryption is enabled for the conference and all conference participants must be
encrypted.
 No Encryption - Encryption is disabled for the conference.
 Encrypt when possible - Enables the negotiation between the MCU and the endpoints and let
the MCU connect the participants according to their capabilities, where encryption is the preferred
setting. For connection guidelines see Mixing Encrypted and Non-encrypted Endpoints in one
Conference.
For more information about recording encrypted conferences, see Dial Out Recording Link
Encryption.
Enabling Encryption at the Participant Level
You can select the encryption mode for each of the defined participants. Encryption options are affected by
the settings of the flag in the system configuration. Undefined participants are connected with the Participant
Encryption option set to Auto, inheriting the conference/Entry Queue encryption setting.
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To enable encryption at the participant level:
» In the Participant Properties – Advanced dialog box, in the Encryption list, select one of the
following options: Auto, On, or Off.
 Auto - The participant inherits the conference/Entry Queue encryption setting. The participant
connects as encrypted only if the conference is defined as encrypted.
 Yes - The participant joins the conference/Entry Queue as encrypted.
 No - The participant joins the conference/Entry Queue as non-encrypted.
Monitoring the Encryption Status
The conference encryption status is indicated in the Conference Properties - General dialog box.
The participant encryption status is indicated by a check mark in the Encryption column in the Participants
list pane.
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The participant encryption status is also indicated in the Participant Properties – SDP tab, where SRTP
indication is listed for each encrypted channel (for example, audio and video).
An encrypted participant who is unable to join a conference is disconnected from the conference. The
disconnection cause is displayed in the Participant Properties – Connection Status dialog box, Security
Failure indication, and the Cause box identifies the encryption related situation.
For more information about monitoring, see Conference and Participant Monitoring.
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Setting Conferences for Telepresence
Mode (AVC CP)
Collaboration Server supports the Telepresence Mode in AVC CP conferences allowing multiple participants
to join a telepresence conference from RPX and OTX high definition rooms as well as traditional, standard
definition video conferencing systems.
Note: Irrelevancy of Chapter to a Platform
Telepresence Mode and its associated layouts are not supported by Polycom RealPresence
Collaboration Server (RMX) 1800 with no DSP cards.
OTX (Telepresence) and RPX (RealPresence) room systems are configured with high definition cameras
and displays that are set up to ensure that all participants share a sense of being in the same room.
Participants using two RealPresence RPX HD 400 Room Systems
The following are examples of situations where an Collaboration Server is needed for Telepresence
configurations:
● RPX to OTX
● RPX 2-cameras/screens to RPX 4-cameras/screens
● 3 or more RPXs
● 3 or more OTXs
Collaboration Server Telepresence Mode Guidelines
System Level
● The Collaboration Server system must be licensed for Telepresence Mode.
● The system must be activated with a Telepresence enabled license key.
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Conference Level
● The Telepresence Mode and Telepresence Layout Mode fields are only displayed in the Conference
Profile dialog box if the Collaboration Server has a Telepresence license installed.
● A Telepresence conference must have Telepresence Mode enabled in its profile.
● In Telepresence Mode, ITP sites are automatically detected.
● When Telepresence Mode is selected in a conference profile, the following options are disabled:
 Borders
 Site Names
 Speaker Indication
 Skins
 Same Layout
 Presentation Mode
 Auto Layout
 Lecture Mode
● The master (center) camera is used for video, audio and content.
● Conference Templates can be used to simplify the setting up Telepresence conferences where
precise participant layout and video forcing settings are crucial. Conference Templates:
 Save the conference Profile.
 Save all participant parameters including their Personal Layout and Video Forcing settings.
● An ongoing Telepresence conference can be saved to a Conference Template for later re-use.
For more information see Conference Templates.
● Telepresence Mode and its associated layouts are not supported by Collaboration Server (RMX)
1800 with no DSP cards.
Automatic Detection of Immersive Telepresence (ITP) Sites
When the conference Telepresence Mode is set to Auto (Default) ITP endpoints are automatically detected.
If an ITP endpoint is detected in such conference, ITP features are applied to all endpoints and the
Collaboration Server sends conference video with the following options disabled:
● Borders
● Site names
● Speaker indication
● Skins
● Same Layout
● Presentation Mode
● Auto Layout
● Lecture Mode
The ITP features are dynamic, and if all ITP endpoints disconnect from the conference, normal conference
video is resumed for the remaining all participants. ITP features are re-applied to all participants should an
ITP endpoint re-connects to that conference.
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Horizontal Striping
Horizontal Striping is used by the Collaboration Server in order to prevent cropping and preserve the aspect
ratio of video for all Telepresence Modes.
Cropping
Cropping is used by the Collaboration Server in order to preserve the aspect ratio of video for all
Telepresence Modes.
Cropping is controlled by the ITP_CROPPING system flag in the system configuration, providing different
cropping options according to the endpoints participating in the Telepresence conference:
● ITP (defult) - In this mode, the area is cropped symmetrically from both top and bottom.
● CP - In this mode, the area is cropped symmetrically from both top-bottom and right-left, where the
width of the cropped area is calculated separately.
● Mixed - In this mode, the area to be stripped is cropped symmetrically cropped from right-left areas,
and asymmetrically from top-bottom areas (16% from top, 84% from bottom).
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Gathering Phase with ITP Room Systems
When a conference is configured to include a Gathering Phase, only one endpoint name is displayed for the
ITP room in the connected participant list of the Gathering slide. The ITP room endpoint with the suffix 1 in
its name receives the Gathering slide.
Aspect ratio for standard endpoints
Standard endpoints (non-ITP) receive video from the Collaboration Server with the same aspect ratio as that
which they transmitted to the Collaboration Server.
Skins and Frames
When Telepresence Mode is enabled, no Skin is displayed and the system uses a black background.
Frames around individual layout windows and the speaker indication are disabled.
RPX and OTX Video Layouts
Additional video layouts have been created to give Telepresence operators more video layout options when
configuring OTX and RPX room systems. These additional video layout options are available to all
endpoints on both conference layout and Personal Layout levels.
OTX / RPX – Additional Video Layouts
Number of Endpoints
Layouts
1
2
4:3
4:3
3
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4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
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OTX / RPX – Additional Video Layouts
Number of Endpoints
Layouts
4
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
4:3
5
9
10+
The following example illustrates the use of standard and additional Collaboration Server Telepresence
layouts when connecting four Room Systems as follows:
● Two OTX Room Systems
 2 active cameras
 6 screens
● Two RPX Room Systems
 8 cameras
 8 screens
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RPX and OTX Room System connected using the RealPresence Collaboration Server
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Room Switch Telepresence Layouts
The Room Switch Telepresence layouts normally controlled by the MLA can be managed by the MCU to
speed updating the conference layouts in large conferences with many endpoints.
Whether the MLA or the MCU controls the Room Switch Telepresence layouts is determined by the
MANAGE_TELEPRESENCE_ROOM_SWITCH_LAYOUTS flag. This flag must be manually added before
changing its value. No system reset is required.
The values are:
● NO (Default) - The MCU does not manage Telepresence Room Switch Layouts and they continue to
be managed by the MLA.
● YES - The MCU manages Telepresence Room Switch Layouts.
When the MCU controls the Telepresence Room Switch layouts
(MANAGE_TELEPRESENCE_ROOM_SWITCH_LAYOUTS = YES) the display is affected according to the
Telepresence Mode Settings in the Conference Profile as follows:
● If the Telepresence Mode = ON
 If no ITP endpoints are connected to the conference, the RMX Room Switch layout applies, in
which case only the speaker is seen.
 When a single participant using an ITP endpoint with either single or multiple screens connects to
the conference, the participant will see black screens.
● If the Telepresence Mode = AUTO
 If no ITP endpoints are connected to the conference, the RMX CP layout applies (unless the
conference layout is defined).
 When a single participant using an ITP endpoint with multiple screens connects to the conference,
the participant will see black screens.
 When a single participant using an ITP endpoint with a single screen connects to the conference,
the MCU will display a self-view of the participant.
● When a TIP system with 3 screens joins a conference, the layout is updated on all screen
simultaneously.
● When a Polycom ITP system with 2, 3, or 4 screens joins the conference, the layout is updated on all
screens simultaneously.
Telepresence Display Decision Matrix
How the speaker video is displayed on the screens of the conference participants is dependent on the
relationship between the number of screens the speaker endpoint contains and the number of screens of
the endpoints of the other conference participants.
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The following Telepresence Display Decision Matrix table below indicates how the speaker video will be
displayed on the various participant endpoints, when the MCU is managing Telepresence Room Switch
conference layouts.
For example, if the speaker’s endpoints has two screens and the participant’s endpoint only one, the
participant’s display is divided into two video layout cells with each video layout cell showing the input of one
of the speaker’s screens (endpoint).
If the participant endpoint has two screens, and the speaker endpoint only one, the speaker’s video will be
displayed on one of the participant’s screens, while the second screen remains black.
Guidelines for Managing the Room Switch Telepresence Layouts by
the MCU
● Only Room Switch layouts can be managed by the MCU. CP (Continuous Presence) layouts continue
to be managed by the MLA.
● Only CP-AVC conferences are supported.
● Lync Clients (with Polycom® ContentConnect™ add-in) are supported. Microsoft Lync is not
supported by Collaboration Server (RMX) 1800 with no DSP cards.
● SVC endpoints are not supported.
● It is recommended that the Speaker Change Threshold be set to 3 seconds.
● Telepresence endpoints are named using a text name followed by a number. For example, if an OTX
Telepresence room is named Oak, the three endpoint names would be Oak1, Oak2, and Oak3.
● Lecture mode is not supported in Telepresence Room Switch conferences managed by the MCU.
(This is because in Lecture mode, unlike Room Switch mode, the lecturer receives the CP layout of
conference participants.)
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● Personal layouts are disabled. Therefore, any features that use personal layouts like Click&View can
not be used to change the layout, and Click&View DTMF digits will be ignored.
● Changing the flag affects only future conferences. Conferences currently running are not affected.
● The Send Content To Legacy Endpoints feature is enabled by default when Telepresence mode is
enabled.
● Layout attributes (no skins, no site names and no borders) should continue for Telepresence layouts
managed by the RMX.
Speaker Priority in CP Video Layouts
The purpose of Speaker Priority Mode is to provide high visibility to the speaker's room and to ensure that
the active speaker in the conference is always displayed in the video layout, and displayed in the best way
possible. If there is space in the layout while the active speaker is displayed, previous speakers are also
displayed.
Reserved Screens
When Speaker Priority mode is selected each Room System reserves screens to provide high visibility
according to maximum number of room-screens in the conference, displaying the active speaker in the
largest video layout cell available.
The Speaker Priority option is selected in the Video Settings tab of the Profile dialog. For more information
see Selecting Speaker Priority
The number of reserved screens depends on the maximum number of room-cameras connected to the
conference. Typically, two room-cameras are displayed one screen, three room-cameras on three screens,
and four room-cameras on two screens.
Reserved screens include an Overlay Layout (Filmstrip) that may be populated with other conference
participants after the Grid Screen(s) have been fully populated with additional conference participants. See
Video Layout Examples.
Grid Screens
Grid screens are symmetric video layouts (2x2, 3x3, 4x4) that are populated with other conference
participants after the Reserved Screens are populated with the current and previous speakers. Grid screens
are only available when the room system has more screens than the number of reserved screens—grid
screens typically exist within 2 or 4 screen systems. See Video Layout Examples.
Video Layout Examples
● A three-screen Room System will reserve three screens if another three-camera Room System
participates in the conference. If the active speaker is using a single camera endpoint, the active
speaker is displayed on a full screen while the two previous speakers are displayed on the other two
screens.
● Irrespective of whether the receiving Room System has more screens than the active speaker’s
Room System:
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•
An active speaker’s two-camera room is displayed on one screen.
•
An active speaker’s four-camera room is displayed on two screens.
● Three-screen Room Systems (e.g. OTX / TIP) will not zoom out when Speaker Priority is selected.
The layout below is used when the active speaker is using a 3-screen room system and the viewing
endpoint is single-screen endpoint.
The layout below is used when the active speaker is using a single-screen endpoint and the viewing
endpoint is single-screen endpoint.
The layout below is used when the active speaker is using a single-screen endpoint and the viewing
endpoint is 3-screen endpoint.
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The layout below is used when the active speaker is using a 3-screen room system and the viewing
endpoint is 3-screen endpoint.
● If a two-screen Room System is displaying a active speaker using a one-camera endpoint, and a
previous speaker also using a one-camera endpoint, the following layouts will be displayed on the
Room System’s two screens.
The layout below is used when the conference includes 4-screen rooms where both the active
speaker and the previous active speaker are using single-screen endpoints.
The layout below is used when the conference includes 1, 2, and 3-screen rooms where the active
speaker is using a single-screen endpoint.
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Selecting Speaker Priority
Speaker Priority is selected in the Telepresence Layout Mode menu in the Video Settings tab of the Profile
dialog. For more information see Defining AVC CP Conferencing Profiles.
Telepresence Mode License
The Telepresence Layout Mode field is enabled only if the Collaboration Server system is licensed for
Telepresence Mode.
AVC endpoints only
Telepresence features are currently only supported in AVC-CP conferences.
SVC or SVC and CP Mixed Mode conferences are not supported.
Sending Content to Legacy Endpoints in Telepresence
Conferences
The Collaboration Server can be configured to manage the layouts of to H.323/SIP/ISDN endpoints that do
not support H.239 Content (legacy endpoints) over the video (people) channel in Telepresence conferences
when Content is being sent. This feature is controlled using the
FORCE_LEGACY_EP_CONTENT_LAYOUT_ON_TELEPRESENCE flag. This flag must be added to
change the value.
The values of the flag are:
● NO (Default) - The MCU does not manage the layouts while Content is sent. Personal layout
changes, for example, by MLA, override the default MCU layout. Legacy endpoints may not display
Content in Telepresence conferences due to layout changes.
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● YES - The MCU manages the layouts while Content is sent. Personal layout changes, for example,
by MLA, are ignored. The layouts for legacy endpoints are managed by the MCU.
Guidelines for Sending Content to Legacy Endpoints in Telepresence
Conferences
● MLA layout change requests for legacy endpoints will be ignored until Content is stopped. At that
point, MLA can be used again.
● Click&View can not be used to change the layout while Content is being sent.
● The Polycom Touch Control can not be used to change the layout while Content is being sent.
Content Display on Legacy Endpoints in Telepresence Conferences
When Content is sent to legacy endpoints in Telepresence conferences, their video layout automatically
changes to the Content layout which is defined by the system flag
LEGACY_EP_CONTENT_DEFAULT_LAYOUT. If MLA is managing the Telepresnce layout prior to Content
being sent, the MCU takes over managing the layout of Legacy endpoints once Content is started. The video
layouts of the other conference participants continue to be managed by MLA.
If MLA was managing the Telepresence layouts, when Content ends, control of the layouts for legacy
endpoints goes back to the MLA after a short time.
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Enabling Telepresence Mode
Telepresence Mode must be configured in a new or existing Conference Profile.
To enable Telepresence in a new or existing Conference Profile:
1 In the RMX Management pane, click Conference Profiles.
2 Click the New Profiles button or open an existing Conference Profile.
3 Define the various profile General, Advanced, Gathering Settings and Video Quality parameters.
For more information on defining Profiles, see Defining New Profiles.
4 Click the Video Settings tab.
5 In the Telepresence Mode field, select one of the following options:
 OFF - When OFF is selected, normal conference video is sent by the Collaboration Server.
 AUTO (Default) - The ITP features are dynamic. When AUTO is selected and an ITP endpoint is
detected, ITP features are applied to the conference video for all participants. If all ITP endpoints
disconnect from the conference, normal conference video is resumed for all remaining
participants. ITP features are re-applied for all participants should an ITP endpoint re-connect to
the conference.
When Telepresence Mode is set to Auto and a one-screen Telepresence unit is in use, the
Collaboration Server controls layouts instead of the MLA. For more information see Polycom
Multipoint Layout Application (MLA) User’s Guide for Use with Polycom Telepresence Solutions.
 ON - ITP features are always applied to the conference video for all participants regardless of
whether there are ITP endpoints connected or not.
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6 In the Telepresence Layout Mode field, select the Telepresence Layout Mode to be used in the
conference. This field is used by VNOC operators and Polycom Multi Layout Applications to retrieve
Telepresence Layout Mode information from the Collaboration Server.
The following modes can be selected (as required by the VNOC and Polycom Multi Layout
Applications):
 Manual
 Continuous presence (MLA) - Room Continuous Presence (default for API backward
compatibility)
 Speaker Priority - Speaker always displayed using the best quality (default)
 Room Switch - Voice Activated Room Switching
7 Select the required video layout.
8 Click OK.
Note: Skins and System Flags in Telepresence Mode
• When Telepresence Mode is enabled, the Skins options are disabled as the system uses a black
background and the frames and speaker indication are disabled.
• It is recommended that the PRESERVE_PARTY_CELL_ON_FORCE_LAYOUT System Flag be
set to YES if the Collaboration Server is used primarily for ITP conferences with MLA.
• For more information see the flag description in System Configuration Flags.
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Monitoring Telepresence Mode
Monitoring Ongoing Conferences
An additional status indicator, Telepresence Mode Enabled, is displayed in the Conference Properties Video Settings tab when monitoring ongoing conferences.
Note: Telepresence Mode License Requirement
The Telepresence Mode Enabled, Telepresence Mode and Telepresence Layout Mode fields are
only enabled if the Collaboration Server has a Telepresence license installed.
If Telepresence Mode is enabled, a check mark is displayed in the check box. This option is grayed as this
is a status indicator and cannot be used to enable or disable Telepresence Mode.
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Monitoring Participant Properties
An additional status indicator, Telepresence, is displayed in the Participant Properties - Advanced tab when
monitoring conference participants.
The Telepresence mode of the participant is indicated:
● RPX - the participant’s endpoint is transmitting 4:3 video format.
● OTX - the participant’s endpoint is transmitting 16:9 video format.
● None.
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Creating Multiple Cascade Links Between
Telepresence Conferences
You can create multiple Cascading links between Collaboration Servers hosting conferences that include
Immersive Telepresence Rooms (ITP) such as Polycom’s OTX and RPX Room Systems.
Guidelines for Creating Multiple Cascading Links between
Conferences
● Basic Cascading topology is used. For more information see Basic Cascading Using IP Cascaded
Link.
● Multiple Cascade Links between conferences are implemented by creating a Link Participant which
consists of a main link and sub-links which are automatically generated and sequentially numbered.
For more information see Creating a Link Participant, Creating a Link Participant.
● All cascaded links must use H.323 protocol.
● Multiple Cascade Links are supported in CP conferencing mode.
● The number of cascading links is defined manually according to the maximum number of Room
System cameras in the cascaded conference.
● When the active speaker is in an Immersive Telepresence Room, Multiple Cascade Links are used,
one link for each of the Room System’s cameras.
 An RPX 4xx Room System requires 4 Cascaded Links to carry the video of its 4 cameras.
 An RPX 2xx Room System requires 2 Cascaded Links to carry the video of its 2 cameras.
 An OTX 3xx Room System requires 3 Cascaded Links to carry the video of its 3 cameras. The
OTX Room System must be configured as Room Switch in order to send multiple streams. When
configured in CP Mode, its cameras zoom out and all 3 screens are sent as one stream.
● The number of links is defined when creating the Link Participant. Each conference in the cascade
must have a Link Participant with the same number of Multiple Cascade Links defined. Calls from
Link Participants not defined with the same number of links are rejected. Number of cascading links
is not identical for all conferences is listed as the Call Disconnection Cause. For more information see
Creating a Link Participant and Monitoring Multiple Cascade Links.
● Although it is possible to disconnect and reconnect specific Multiple Cascade Links using the
RealPresence Collaboration Server Web Client / RealPresence Collaboration Server Manager it not
advisable to do so.
 If the main link is disconnected all sub-links are disconnected and deleted.
Reconnecting the main link reconnects all sub-links.
 If a sub-link is disconnected it remains disconnected until it is manually reconnected.
 The number of Multiple Cascade Links cannot be modified while any of the links are in a
disconnected state. All previous links must be deleted before modification is possible.
For more information see Monitoring Multiple Cascade Links.
●
A Link Participant can be dragged from the address book into a conference.
 If it is the first Link Participant in the conference, the number of Multiple Cascade Links defined
for the participant are created and connected.
 If it is not the first Link Participant in the conference, the number of Multiple Cascade Links defined
for the participant is ignored.
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● If there are insufficient resources to connect all Multiple Cascade Links in either of the RMXs, none
of the links are connected and resources deficiency -0 is listed as the Call Disconnection Cause. For
more information see Monitoring Multiple Cascade Links.
● Multiple Cascade Links that are not used by MLA are inactive but continue to consume resources.
● All RMXs participating in the cascade must have the same Telepresence Mode definitions, either all
defined as CP or all defined as Room Switch.
● When Multiple Cascade Links are defined in the Conference Profile, the Layout Type field of the Link
Participant’s Participant Properties - Media Sources dialog box is set to Conference and cannot be
modified.
● TIP Telepresence Rooms (CTS) are supported without Content.
For more information see Appendix I - Polycom Open Collaboration Network (POCN).
Enabling and Using Multiple Cascade Links
The settings required to enable Multiple Cascade Links on the RMX are minimal and are described in
Creating a Link Participant.
Most of the layout configuration is performed using Polycom’s Multipoint Layout Application (MLA).
The figures RMX Telepresence Layout Mode - Room Switch and RMX Telepresence Layout Mode Continuous Presence show example layouts and media flows when MLA is configured for a cascading
conference between two RMXs.
In the figure RMX Telepresence Layout Mode - Room Switch:
● The OTX Room System connects to RMX A.
● The RPX Room System connects to RMX B.
● This layout requires the Telepresence Layout Mode to be set to Room Switch in the Conference
Profiles of the Cascading Conferences in each RMX.
● The current speaker is a participant in the RPX ITP Room.
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● Directional media flows, A  B, are shown separately for readability purposes.
RMX Telepresence Layout Mode - Room Switch
In the RMX Telepresence Layout Mode - Continuous Presence figure:
 An HDX endpoint and an OTX Room System connects to RMX A.
 An HDX endpoint and an RPX Room System connects to RMX B.
 This layout requires that the Telepresence Layout Mode to be set to Continuous Presence in the
Conference Profiles of the Cascading Conferences in each RMX.
 The current speaker is the HDX endpoint connected to RMX A.
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RMX Telepresence Layout Mode - Continuous Presence
For more information see:
● Setting Conferences for Telepresence Mode (AVC CP).
● Polycom® Multipoint Layout Application (MLA) User’s Guide for Use with Polycom Telepresence
Solutions.
● Polycom® Immersive Telepresence (ITP) Deployment Guide
Creating a Link Participant
Link Participant in the Dial Out RMX
The Link Participant is defined in the New Participant dialog box.
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To create a Link Participant in the dial-out RMX:
1 In the General tab:
 In the Dialing Direction drop-down menu, select Dial out.
 In the Type drop-down menu, select H.323.
For more information see the Creating a Cascade Enabled Dial-out/Dial-in Participant Link.
2 In the Advanced tab:
(This field is only enabled if the RMX system is licensed for Telepresence Mode.)
 The Cascade dial-out participant must be defined as Slave. (The dial-in undefined participant
should be defined as the Master.)
 In the Number of cascaded links (for ITP) drop-down menu, select the maximum number of
Multiple Cascade Links required according to the number of Room System endpoints in the
cascaded conference.
This field enables the administrator to select the maximum number of Multiple Cascade Links
required according to the number of Room System endpoints in the cascaded conference.
For example if an RPX 4xx is included, the number of links required is 4.
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The RMX automatically adds a number suffix to the name of the Link Participant, for example if
the Participant Link Name is CascadeLink and the Number of cascaded links (for ITP) field
is set to 4, the following Multiple Cascade Links are created:
 CascadeLink-1
 CascadeLink-2
 CascadeLink-3
 CascadeLink-4
Participant Link in the Dial In RMX
The call from Participant Link defined in the Dial-out RMX is identified by the Dial-in RMX as having been
initiated by a Participant Link.
Suffixes are appended to the Multiple Cascade Links according to the Number of cascaded links (for ITP)
field depending on whether the Dial -In Participant Link is defined or un-defined:
An Undefined Participant Link
The Multiple Cascade Link names are automatically assigned by the RMX.
For example on a RMX 2000 the names of the links are:
● POLYCOM RMX 2000-1
● POLYCOM RMX 2000-2
● POLYCOM RMX 2000-3, etc.
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A Defined Participant Link
The Multiple Cascade Link names are assigned according to the name of the defined participant that is to
function as the cascade link and the Number of cascaded links (for ITP) information sent by the calling
Dial-Out Participant Link.
For example if the defined participant that is to function as the cascade link is named
Cascade_Link_From_B the names of the links are:
● Cascade_Link_From_B-1
● Cascade_Link_From_B-2
● Cascade_Link_From_B-3, etc.
Monitoring Multiple Cascade Links
Multiple Cascade Links connections can be monitored in the Participants list of the RMX Web Client / RMX
Manager main screen:
Disconnection Causes
● If there are insufficient resources to connect all the required links:
 None of the links are connected.
 The first link is listed as Disconnected in the Participants list of the RMX Web Client / RMX
Manager main screen.
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▪
Resource deficiency is listed as the Call Disconnection Cause in the Participant Properties Connection Status dialog box.
● If a calling Link Participant is not defined with same number of links as all the other Link Participants
in the cascaded conferences:
 The call is rejected.
 The Call Disconnection Cause is: Number of cascading links is not identical for
all conferences.
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Note: Virtual Platform Requirement
When using RealPresence Platform, the conferencing parameters should be defined in the
RealPresence DMA system component and not directly in the RealPresence Collaboration Server
(RMX) 1500/1800/2000/4000 component.
Various conferencing modes and video features require additional settings, such as system flag settings,
conference parameters and other settings. In depth explanations of these additional settings are described
in the following sections:
● Video Preview
● Auto Scan and Customized Polling in Video Layout
● Packet Loss Compensation - LPR and DBA
● Layout Indications
● Lecture Mode
● Audio Algorithm Support
● Automatic Muting of Noisy Endpoints
● Permanent Conference
● Closed Captions
Video Preview
Note: Feature Applicability
Video Preview is applicable for AVC endpoints only.
Collaboration Server users can preview the video sent from the participant to the conference (MCU) and the
video sent from the conference to the participant. It enables the Collaboration Server users to monitor the
quality of the video sent and received by the participant and identify possible quality degradation.
The video preview is displayed in a separate window independent to the Collaboration Server Web Client.
All Web Client functionality is enabled and conference and participant monitoring as well as all other user
actions can be performed while the video preview window is open and active. Live video is shown in the
preview window as long as the window is open. The preview window closes automatically when the
conference ends or when participant disconnects from the conference. It can also be closed manually by
the Collaboration Server user.
Video Preview Guidelines
● Video Preview is supported in CP Conferencing Mode only.
● Video preview is available for AVC participants. It is not available for SVC participants.
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● Video preview window size and resolution are adjusted to the resolution of the PC that displays the
preview.
● Video Preview of the video sent from the conference to the participant is shown according to the line
rate and video parameters of the level threshold to which the participant is connected.
● All users can view a video preview.
● Only one preview window can be displayed for each Collaboration Server Web Client connection
(workstation).
● Only one preview window can be displayed for a single conference and up to four preview windows
can be displayed for each media card on different workstations (one per workstation and one per
conference). For example, if the Collaboration Server contains two media cards, and there are 5
conferences running on the Collaboration Server, if five conferences are running on the same media
card, only four conferences can be previewed from four different workstations. If four or less
conferences are running on one media card and the remaining conferences are running on the other
media card, all five conferences can be previewed.
● Live video that is shown in the preview window does not include the Content when it is sent by the
participant.
● Video Preview is supported in cascaded conferences.
● If the video preview window is opened when the IVR slide is displayed to the participant, it will also
be displayed in the video preview window.
● Video Preview is supported with H.264 High Profile.
● Video Preview is not supported for endpoints using the RTV protocol.
● Video Preview is disabled in encrypted conferences.
● Video preview cannot be displayed when the participant’s video is suspended.
● Participant’s video preview and the Polycom Desktop application (such as CMAD) window cannot be
open and running simultaneously on the same PC as both require the same DirectDraw resource.
Workstation Requirements to Display Video Preview
To be able to display the video preview window, the following minimum requirements must be met:
● Windows XP, Windows Vista and Windows 7
● Internet Explorer 7 and later
● DirectX is installed
● DirectDraw Acceleration must be enabled and no other application is using the video resource
● Hardware acceleration must be enabled
Testing your Workstation
To ensure that your workstation can display the video preview window:
1 In Windows, click Start > Run.
The Run dialog box opens.
2 In the Open field, type dxdiag, and press Enter or click OK.
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A confirmation message is displayed.
3 Click Yes to run the diagnostics.
The DirectX Diagnostic Tool dialog box opens.
4 Click the Display tab.
To be able to display the video preview window, the DirectDraw Acceleration and Direct3D
Acceleration options must be Enabled.
If the video card installed in the PC does not support DirectDraw Acceleration, a black window may
be viewed in the Video Preview window.
5 Click the Exit button.
Previewing the Participant Video
You can preview the video sent from the participant to the conference (MCU) and the video sent from the
conference to the participant by selecting the appropriate option from the Participant’s pop-up menu.
To preview the participant video:
1 List the conference participants in the Participants pane.
2 Right-click the participant whose video you want to preview, and select one of the following options:
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 View Participant Sent Video - To display the video sent from the participant to the conference.
 View Participant Received Video - To display the video sent from the conference to the
participant.
The Video Preview window opens.
If the video card installed in the PC does not support DirectDraw Acceleration, a black window may
be viewed.
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Auto Scan and Customized Polling in Video Layout
Note: Feature Applicability
Auto scan and customized polling in video layout is applicable for AVC CP conferences only.
Auto Scan enables you to define a single cell in the conference layout to cycle the display of participants
that are not in the conference layout.
Customized Polling allows the cyclic display to be set to a predefined order for a predefined time period.
The cyclic display only occurs when the number of participants is larger than the number of cells in the
layout.
Guidelines for Using Auto Scan and Customized Polling
● Auto Scan and Customized Polling are supported in AVC CP conferences only.
● Participants that are in the conference layout will not appear in the Auto Scan enabled cell.
● If Customized Polling is not used to define the order of the Auto Scan it will proceed according to order
in which the participants connected to the conference.
● If the user changes the conference layout, the Auto Scan settings are not exported to the new layout.
If the user changes the conference layout back to the layout in which Auto Scan was enabled, Auto
Scan with the previous settings will be resumed.
Enabling the Auto Scan and Customized Polling
Auto Scan and Customized Polling are enabled during the ongoing conference, in the Conference
Properties - Video Settings dialog box.
Enabling the Auto Scan
You enable the Auto Scan feature by selecting it in the Video Layout cell.
To enable Auto Scan:
1 In the Collaboration Server Web Client Main Screen - Conference list pane, double-click the
conference or right-click the conference and then click Conference Properties.
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2 In the Conference Properties - General dialog box, click Video Settings.
The Video Settings dialog box is displayed.
3 If Auto Layout check box is selected, clear it.
4 In the video layout cell to be designated for Auto Scan, select from the drop-down menu Auto Scan.
5 Select from the Auto Scan Interval(s) list the scanning interval in seconds.
6 Click the Apply button to confirm and keep the dialog box open, or Click OK.
Customized Polling
The order in which the Auto Scanned participants are displayed in the Auto Scan enabled cell of the video
layout can be customized.
To define the scanning order in the Customized Polling tab:
1 Open the Conference Properties dialog box.
2 Click the Customized Polling tab.
The Customized Polling dialog box is displayed.
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All conference participants are listed in the left pane (All Participants) whereas the participants to be
displayed in the Auto Scan enabled cell are listed in the right pane (Scanning Order).
The dialog box buttons are summarized in the following table:
Customized Polling Dialog Box Command Buttons
Button
Description
Add
Select a participant and click this button to add the participant to the list of
participants to be Auto Scanned.
The participants name is removed from the All Participants pane.
Delete
Select a participant and click this button to delete the participant from the list
of participants to be Auto Scanned.
The participants name is moved back to the All Participants pane.
Add All
Add all participants to the list of participants to be Auto Scanned.
All participants’ names are removed from the All Participants pane.
Delete All
Delete all participants from the list of participants to be Auto Scanned.
All participants’ names are moved back to the All Participants pane.
Up
Select a participant and click this button to move the participant up in the
Scanning Order.
Down
Select a participant and click this button to move the participant down in the
Scanning Order.
3 Click the Apply button to confirm and keep the dialog box open, or click OK.
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Packet Loss Compensation - LPR and DBA
Note: Feature Applicability
LPR and DBA are applicable for AVC conferencing only.
Lost Packet Recovery (LPR) and Dynamic Bandwidth Allocation (DBA) help minimize media quality
degradation that can result from packet loss in the network. Packet loss Compensation is available in AVC
CP Conferencing Mode only and is not supported in SVC Conferencing Mode or CP and SVC Conferencing
Mode.
Packet Loss
Packet Loss refers to the failure of data packets, transmitted over an IP network, to arrive at their
destination. Packet Loss is described as a percentage of the total packets transmitted.
Causes of Packet Loss
Network congestion within a LAN or WAN, faulty or incorrectly configured network equipment or faulty
cabling are among the many causes of Packet Loss.
Effects of Packet Loss on Conferences
Packet Loss affects the quality of:
● Video – frozen images, decreased frame rate, flickering, tiling, distortion, smearing, loss of lip sync
● Audio – drop-outs, chirping, audio distortion
● Content – frozen images, blurring, distortion, slow screen refresh rate
Lost Packet Recovery
The Lost Packet Recovery (LPR) algorithm uses Forward Error Correction (FEC) to create additional
packets that contain recovery information. These additional packets are used to reconstruct packets that are
lost, for whatever reason, during transmission. Dynamic Bandwidth Allocation (DBA) is used to allocate the
bandwidth needed to transmit the additional packets.
Lost Packet Recovery Guidelines
● If packet loss is detected in the packet transmissions of either the video or Content streams:
 LPR is applied to both the video and Content streams.
 DBA allocates bandwidth from the video stream for the insertion of additional packets containing
recovery information.
● Due to the LPR increasing the required bandwidth, content rate might be reduced (to no less than
64K) to preserve it.
When this reduction is insufficient, either the video rate or the LPR strength are reduced as well,
depending on the value of the system flag LPR_CONTENT_RATE_ADJUST_WEAK_LPR. For more
information on that flag see LPR_CONTENT_RATE_ADJUST_WEAK_LPR.
● LPR is supported in H.323 and SIP networking environments only.
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● In LPR-enabled Continuous Presence conferences:
 Both LPR-enabled and non-LPR-enabled endpoints are supported.
 The LPR process is not applied to packet transmissions from non-LPR-enabled IP (H.323 and
SIP) and ISDN endpoints.
 Non-LPR-enabled endpoints can be moved to LPR-enabled conferences.
 LPR-enabled endpoints cannot be moved to non-LPR-enabled conferences.
● In LPR-enabled Video Switched conferences:
 H.323 and SIP endpoints are supported.
 When cascading between conferences running on Collaboration Server and MGC (Polycom
legacy MCU), LPR is not supported over the link between the two conferences.
 Non-H.323 participants cannot be created, added or moved to LPR-enabled Video Switched
conferences.
● When connecting via an Entry Queue:
 A participant using an LPR-enabled endpoint can be moved to a non-LPR-enabled conference.
The participant is connected with LPR enabled.
 SIP and ISDN/PSTN participants cannot be moved to LPR-enabled Video Switched conferences.
Enabling Lost Packet Recovery
LPR is enabled or disabled in the Conference Profile dialog box.
● CP Conferences – LPR is enabled by default in the New Profile – Advanced dialog box.
● VSW Conferences – If Video Switching is selected, the LPR check box is automatically cleared and
LPR is disabled. LPR can be enabled for VSW conferences but H.320 and SIP participants will not
be able to connect.
For more information, see Defining New Profiles.
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Monitoring Lost Packet Recovery
In the Participant Properties – H.245 tab, LPR activity is displayed in all three panes.
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In the Participant Properties – Channel Status tab, check box indicators show LPR activation in the local
and remote (transmit and receive) channels.
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Layout Indications
Note: Layout Indications Applicability
• Layout Indications are applicable to AVC endpoints only.
• Layout Indications are not supported on Collaboration Server 1800, Entry Level.
Network Quality Indication
If network quality issues occur, the Network Quality icon provide information to participants about their own
network quality and that of other participants displayed in the cells of the conference Video Layout.
Network Quality Levels
Network quality is determined by the percentage of packet loss according to the following default threshold
values:
● Packet loss less than 1% is considered Normal
● Packet loss in the range of 1% - 5% is considered Major
● Packet loss above 5% is considered Critical.
Major and Critical states are indicated by the following icons:
When network quality improves from Critical to Major remaining stable for 5 seconds, the Network Quality
Indicator is changed accordingly and when network quality improves from Major to Normal, remaining stable
for 5 seconds, the Network Quality Indicator is no longer displayed.
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Indication Threshold Values
The default Major and Critical indication threshold values can be modified by manually adding the following
System Flags and modifying their values as required.
Network Quality Icon - Indication Threshold Flags
Flag
Description
NETWORK_IND_MAJOR_PERCENTAGE
The percentage degradation due to packet loss required to
change the indicator from Normal to Major.
Default: 1
NETWORK_IND_CRITICAL_PERCENTAGE
The percentage degradation due to packet loss required to
change the indicator from Major to Critical.
Default: 5
For more information see .
Displaying the Network Quality icons
Network Quality icons are not supported in SVC Conferencing Mode and AVC - Video switched
conferences.
Network Quality icons are displayed for:
● The video channel only in AVC Conferencing Mode.
Content, audio and FECC channel quality issues are not indicated.
● The participant’s own endpoint:
 Network Quality icons are displayed by default and can be disabled
 For media transmitted to and received from the Collaboration Server (Video in / Video out).
● Participants displayed in the cells of the conference video layout:
 Network Quality icons are not displayed by default and can be enabled
 The media transmitted from the endpoint to the Collaboration Server (Video in).
Customizing the Network Quality Icon Display
Display of the Network Quality icon can be customized for the participant’s own endpoint or for the
Participants displayed in the cells of the conference Video Layout.
The display of Network Quality icon (showing or hiding the icon) and the position of the icon in the video
layout cell can be customized by manually adding the following System Flags and modifying their values as
required.
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Network Quality Icon - Display Customization Flags
Flag
Description
DISABLE_SELF_NETWORK_IND
Disable the display of the Network Quality icon of the
participant’s own endpoint.
Default: NO
Range: YES / NO
NOTE: This flag’s function has been replaced by the
Network Quality check box in the Layout Indications tab of
the New Profile / Profile Properties dialog.
For more information see New AVC CP Profile - Layout
Indications Parameters.
DISABLE_CELLS_NETWORK_IND
Disable the display of Network Quality icons displayed in
the cells of the conference Video Layout.
Default: YES
Range: YES / NO
SELF_IND_LOCATION
Change the location of the display of the Network Quality
icon of the participant’s own endpoint.
Default: BOTTOM_RIGHT
Range:
• TOP_ LEFT
• TOP
• TOP_RIGHT
• BOTTOM_ LEFT
• BOTTOM
• BOTTOM_RIGHT
NOTE: This flag’s function has been replaced by the
Network Quality check box in the Layout Indications tab of
the New Profile or Profile Properties dialog.
For more information see New AVC CP Profile - Layout
Indications Parameters.
CELL_IND_LOCATION
Change the location of the display of Network Quality icon
displayed in the cells of the conference Video Layout.
Default: TOP_RIGHT
Range:
• BOTTOM_LEFT
• BOTTOM_RIGHT
• TOP_LEFT
• TOP_RIGHT
For more information see the .
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Recording Indications
When Display Recording Icon is selected in either the Recording or Layout Indications tab of the New
Profile or Profile Properties dialog, the recording status is indicated by the following icons:
Recording
For more information, see Recording and Streaming Conferences and New AVC CP Profile
- Layout Indications Parameters.
Recording Paused
For more information, see Recording and Streaming Conferences and New AVC CP Profile
- Layout Indications Parameters.
Note: Recording Indication Duplication
The Recording Indication icon is a duplicate of the Display Recording icon of the Recording tab of the
Profile dialog. For more information, see Recording and Streaming Conferences and New AVC CP
Profile - Layout Indications Parameters..
Audio and Video Participants Indication
During an ongoing conference, Audio Participants Indication icon provides an indication to all participants
that audio-only endpoints and secondary video devices are connected to the conference.
Similarly, video participant(s) presence is indicated by a Video Participant Indication icon, which is useful
when not all the connected participants can be displayed over the used layout. The icon indicating video
participant presence is:
The Audio/Video Participant indication icon is displayed in the conference video screen along with the an
adjacent number (to the right) of the connected endpoints of this type. Up to 99 participants of each type
can be indicated; when more than 99 participants are in the conference, the number is indicated as 99+.
These icons are displayed as part of a group that includes a network quality indication and a recording
indication displayed when recording is active. The icon group is displayed for AVC endpoints only.
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In the example below, all indication types are selected, and indicate 10 audio and 14 video participants.
In addition this figure demonstrates the order of the Layout Indications, which are ordered from the outside,
towards the inside as follows:
● Recording indication
● Audio participant indication
● Video participant indication
● Network quality indication
Setting Layout Indication via User Interface
● Audio/video participants indication appearance, is dependent on selecting the appropriate
check-boxes in the Conference Profile Properties > Layout Indications tab.
Backwards Compatibility
Conferences launched using Profiles that were defined in versions that do not include the Layout
Indications tab will behave as if all the parameters in the tab are unchecked.
● Indications may appear permanently, or for a short duration upon an endpoint joining or disconnecting
the conference, for the specified duration, all depending on the Collaboration Server user
configuration in that tab.
Note that when the audio participant indication is set to display permanently, it is displayed only if
audio participants are connected to the conference.
● The position of the icon can be customized, as shown in the figure below:
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Layout Indications Icon Display Positions
Following are examples of the Indication Icons display positions (97 audio participants are connected to the
conference).
Top-center (default) with all indication icons enabled, except video
Top-left, with all indication icons enabled, except video
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Top-left, with all indication icons enabled, except video, recording enabled but not active
Top-left, audio participants, network quality enabled, recording not enabled
Top-right with all indication icons enabled, except video, recording active
Layout Indications Guidelines
● Layout indications appear only on the video display of AVC endpoints, and are applicable for AVC CP
Only (Continuous Presence) and Mixed CP and AVC conferencing modes.
● Lync users can view layout indications provided the video sent to the Lync endpoint is transcoded.
However, layout indications are not embedded in the video sent to the link towards the AV MCU, to
preserve the Lync user experience.
● When indications are set to appear upon a change in connected participants, all selected indications
appear together, even when the change occurred only in one type of participants.
However, when the change occurs in participants whose type is unselected, and the indication
display is not permanent, no indications appear.
● Layout indications are displayed only in endpoints with resolutions of CIF and up.
● Layout indications are not displayed during the gathering phase.
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● Remote participants (that is, connected to a cascaded conference) are not included in the count, as
well as the cascading link itself. Layout indications are displayed within each Collaboration Server
participants independently, thus no layout indications are embedded in the video sent towards the
cascading link.
In that context, Lync endpoints connected via the AV MCU are included in the count, although other
video participants connected to the AV MCU conference are not included.
● The Layout Indications tab is hidden in SVC, VSW, and TIP-enabled conferences.
● Layout indications can be set only in conference profiles, and not for ongoing conferences.
● The same indication display duration is applicable for all indications.
● For TelePresence endpoints:
 The layout indications appear only on the main screen.
 Only non-TIP endpoints can view the layout indications.
 TelePresence rooms with three screens are considered as a single endpoint in the video
endpoints count.
● Upgrade of endpoints from audio to video, or downgrade from video to audio, is reflected in the layout
indications count.
● PSTN/H.323 audio participants calling the Collaboration Server Gateway profile, and dialing-out a
VMR on the DMA via SIP/H.323, are counted as either an audio or a video endpoint, regardless of
the Collaboration Server they are connected to.
● AVC endpoints connected to a Polycom® RealPresence® WebSuite™ client, can view layout
indications, depending on the conference profile configuration.
1x1 Layout Scenario Adaptation
To display indications according to the local endpoint’s capability (4CIF and up), Collaboration Servers
(2000/4000) should disable the system’s 1x1 layout functionality that includes sending the current speaker’s
resolution.
For example, in a conference between two video endpoints the layout is 1x1 by default. If the resolution of
one of the endpoints is lower than 4CIF and the other higher than 4CIF, RealPresence Collaboration Servers
2000/4000 will follow the speaker and not transmit the Indication Icons when the lower resolution endpoint
is the active speaker. (RealPresence Collaboration Servers 1800 never follow the speaker.) As a result, the
endpoint capable of higher resolution will not receive the Indication Icons.
For further details on how to determine the layout indications position, see New AVC CP Profile - Layout
Indications Parameters.
This behavior can be modified by manually adding the
HW_FOLLOW_SPEAKER_RESOLUTION_ON_1X1_LAYOUT flag to the System Configuration and
modifying its value.
For more information, see HW_FOLLOW_SPEAKER_RESOLUTION_ON_1X1_LAYOUT.
Lecture Mode
Lecture Mode enables all participants to view the lecturer in full screen while the conference lecturer sees
all the other conference participants in the selected layout while he/she is speaking. When the number of
sites/endpoints exceeds the number of video windows in the layout, switching between participants occurs
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every 15 seconds. Conference participants cannot change their Personal Layouts while Lecture Mode is
enabled.
Automatic switching is suspended when one of the participants begins talking, and it is resumed
automatically when the lecturer resumes talking.
Note: Feature applicability
Lecture mode is applicable for AVC CP conferencing only.
Enabling Lecture Mode
Lecture Mode is enabled at the conference level by selecting the lecturer. Conference participants cannot
change their Personal Layouts while Lecture Mode is enabled.
Automatic switching between participants viewed on the lecturer’s screen is enabled in the conference
Profile.
Selecting the Conference Lecturer
Selecting a lecturer for the ongoing conference, enables the Lecture Mode. You can select the lecturer:
● during the definition of the ongoing conference
● after the conference has started and the participants have connected to the conference.
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To select the lecturer and enable the Lecture Mode while starting the conference:
» In the Conference Properties - Participant dialog box, enable the Lecture Mode using one of the
following methods:
To automatically select a defined participant:
1 Add participants to the conference either from the Address book or by defining new participants.
2 In the Lecturer field, select the lecturer from the list of the defined participants.
To automatically select the lecturer:
» In the Lecturer field, select [Auto].
In this mode, the conference speaker becomes the lecturer.
To select the lecturer and enable the Lecture Mode during the ongoing conference:
1 Make sure that the participant you want to designate as the lecturer has connected to the
conference.
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2 In the Conference Properties - Video Settings dialog box, in the Lecturer field, select the lecturer
from the list of the connected participants.
Note: Dial-in/out Participants
Defined dial out and dial in participants are considered to be two separate participants even if they
have the same IP address/number. Therefore, if a defined dial-out participant is added to the
conference and the same participant then dials in (before the system dialed out to that participant)
the system creates a second participant in the Participants list and tries to call the dial-out
participant. If the dial-out participant was designated as the conference lecturer, the system will not
be able to replace that participant with the dial-in participant that is connected to the conference.
Enabling the Automatic Switching
Automatic switching between participants viewed on the lecturer’s screen is enabled in the conference
Profile, or during the ongoing conference, in the Conference Properties.
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● In the Profile Properties - Video Settings tab, select the Lecturer View Switching check box.
This option is activated when the conference includes more sites than windows in the selected layout.
If this option is disabled, the participants will be displayed in the selected video layout without
switching.
For more information about Profile definition, see Defining AVC-Based Conference Profiles.
● Once the conference is running, in the Conference Properties - Video Settings dialog box, select
the Lecturer View Switching check box.
Lecture Mode Monitoring
A conference in which the Lecture Mode is enabled is started as any other conference. The conference runs
as an audio activated Continuous Presence conference until the lecturer connects to the conference. The
selected video layout is the one that is activated when the conference starts. Once the lecturer is connected,
the conference switches to the Lecture Mode.
When Lecturer View Switching is activated, it enables automatic switching between the conference
participants in the lecturer’s video window. The switching in this mode is not determined by voice activation
and is initiated when the number of participants exceeds the number of windows in the selected video
layout. In this case, when the switching is performed, the system refreshes the display and replaces the last
active speaker with the current speaker.
When one of the participants is talking, the automatic switching is suspended, showing the current speaker,
and it is resumed when the lecturer resumes talking.
If the lecturer is disconnected during an Ongoing Conference, the conference resumes standard
conferencing.
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Forcing is enabled at the Conference level only. It applies only to the video layout viewed by the lecturer as
all the other conference participants see only the lecturer in full screen.
If an asymmetrical video layout is selected for the lecturer (i.e. 3+1, 4+1, 8+1), each video window contains
a different participant (i.e. one cannot be forced to a large frame and to a small frame simultaneously).
When Lecture Mode is enabled for the conference, the lecturer is indicated by an icon in the Role column
of the Participants list.
To control the Lecture Mode during an Ongoing Conference:
1 During the Ongoing Conference, in the Conference Properties - Video Settings dialog box you
can:
 Enable or disable the Lecture Mode and designate the conference lecturer in the Lecturer list;
select None to disable the Lecture Mode or select a participant to become the lecturer to enable it.
 Designate a new lecturer.
 Enable or disable the Lecturer View Switching between participants displayed on the lecturer
monitor by selecting or clearing the Lecturer View Switching check box.
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 Change the video layout for the lecturer by selecting another video layout.
Restricting Content Broadcast to Lecturer
Content broadcasting can be restricted to the conference lecturer only, when one of the conference
participants is set as the lecturer (and not automatically selected by the system). Restricting the Content
Broadcast prevents the accidental interruption or termination of H.239 Content that is being shared in a
conference.
Content Broadcast restriction is enabled by setting the
RESTRICT_CONTENT_BROADCAST_TO_LECTURER system flag to ON. When set to OFF (default) it
enables all users to send Content.
When enabled, the following rules apply:
● Content can only be sent by the designated lecturer. When any other participant tries to send Content,
the request is rejected.
● If the Collaboration Server user changes the designated lecturer (in the Conference Properties Video Settings dialog box), the Content of the current lecturer is stopped immediately and cannot be
renewed.
● The Collaboration Server User can abort the H.239 Session of the lecturer.
● Content Broadcasting is not implemented in conferences that do not include a designated lecturer
and the lecturer is automatically selected by the system (for example, in Presentation Mode).
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Muting Participants Except the Lecturer
Note: Feature Applicability
Muting participant except the lecturer is possible in AVC CP conferencing only.
When the Mute Participants Except Lecturer option in the Conference Profile is enabled, the audio of all
participants in the conference except for the lecturer can be automatically muted upon connection to the
conference. This prevents other conference participants from accidentally interrupting the lecture, or from a
noisy participant affecting the audio quality of the entire conference. Muted participants cannot unmute
themselves unless they are unmuted from the Collaboration Server Web Client/RMX Manager.
Guidelines for Muting all the Participants Except the Lecturer
● Both administrators and operators (users) are allowed to set the Mute Participants Except Lecturer
option.
● When the Mute Participants Except Lecturer option is enabled, the mute indicator on the participant
endpoints are not visible because the mute participants was initiated by the MCU. Therefore, it is
recommended to inform the participants that their audio is muted by using the Message Overlay
functions.
● When the Mute Participants Except Lecturer option is enabled in the Conference Profile settings, all
conferences to which this profile is assigned will start with this option enabled. All participants, except
for the designated lecturer, are muted.
● The Mute Participants Except Lecturer option can be enabled or disabled at any time after the start
of the conference. When enabled, it allows all the conference participants to converse before the
lecturer joins the conference or before they are muted. When disabled, it unmutes all the participants
in the conference.
● If the endpoint of the designated lecturer is muted when the lecturer connects to the conference, the
lecturer remains muted until the endpoint has been unmuted.
● When you replace a lecturer, the MCU automatically mutes the previous lecturer and unmutes the
new lecturer.
● When you disconnect a lecturer from the conference or the lecturer leaves the conference, all
participants remain muted but are able to view participants in regular video layout until the you disable
the Mute Participants Except Lecturer option.
● A participant can override the Mute Participants Except Lecturer option by activating the Mute All
Except Me option using the appropriate DTMF code, provided the participant has authorization for
this operation in the IVR Services properties. The lecturer audio is muted and the participant audio is
unmuted. You can reactivate the Mute Participants Except Lecturer option after a participant has
previously activated the Mute All Except Me option. The participant is muted and the lecturer, if
designated, is unmuted.
● In cascaded conferences, all participants (including the link participants) except the lecturer are
muted. Only the lecturer is not muted.
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Enabling the Mute Participants Except Lecturer Option
The Mute Participants Except Lecturer option is enabled or disabled (default) in the Conference Profile or
in an ongoing conference in the Profile Properties - Audio Settings tab.
Note: Audio Clarity In Capture
The Audio Clarity features is not applicable for Collaboration Servers 1800 and Virtual Edition, thus
the relevant check-box does not appear in this tab.
When the Mute Participants Except Lecturer option is enabled and a conference has started, the Mute by
MCU icon is displayed in the Audio column in the Participants pane of each participant that is muted.
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Audio Algorithm Support
The Collaboration Server supports the following audio algorithms in AVC conferences: G.711, G.722,
G.722.1, G.722.1C, G.729A, G.719, G. 728, G.723.1, siren 7, Siren14, Siren 22 (in mono or stereo) and
SirenLPR.
Polycom’s proprietary Siren 22 and industry standard G.719 audio algorithms are is supported for
participants connecting with Polycom endpoints.
The Siren 22 audio algorithm provides CD-quality audio for better clarity and less listener fatigue with audio
and visual communication applications. Siren 22 requires less computing power and has much lower
latency than alternative wideband audio technologies.
The Siren 7 audio codec is the preferred codec for SIP/Lync calls, depending on the value of the system
flag ALLOW_SIREN7_CODEC. This preference is aimed at preventing disconnection of Lync clients using
audio rates smaller than 42Kbps, when the Lync server is configured to allow 33Kbps audio rate. For more
information see ALLOW_SIREN7_CODEC.
The SirenLPR audio algorithm provides CD-quality audio for better clarity and less listener fatigue with audio
and visual communication applications.
In SVC conferences, the system supports SAC (Scalable Audio Coding) audio algorithm.
Audio Algorithm Support Guidelines
● Siren 22 and G.719 are supported in both mono and stereo.
● Stereo is supported in H.323 calls only.
● Siren 22 is supported by Polycom HDX and Group series endpoints, version 2.0 and later.
● Siren 7 is the preferred audio codec for SIP/Lync clients, provided the system flag
ALLOW_SIREN7_CODEC is set to YES.
● Where low line rates affect audio quality, it is possible to disable G.729 codec in favor of G.711 by
modifying the ENABLE_G729 system flag value to NO.
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● G.728 is supported in H.323, SIP and ISDN environments.
● SirenLPR is disabled by default and can be enabled by setting the system flag ENABLE_SIRENLPR,
to YES.
● SirenLPR is supported:
 In IP (H.323, SIP) calls only.
 In CP and VSW conferences.
 With Polycom CMAD and HDX 3.0.1 and later and Group series endpoints.
 For mono audio at audio line rates of 32Kbps, 48Kbps and 64Kbps.
 For stereo audio at audio line rates of 64Kbps, 96Kbps and 128Kbps.
SIP Encryption
The ENABLE_SIRENLPR_SIP_ENCRYPTION System Flag enables the SirenLPR audio algorithm when
using encryption with the SIP protocol.
The default value of this flag is NO meaning SirenLPR is disabled by default for SIP participants in an
encrypted conference. To enable SirenLPR the System Flag must be added to system.cfg and its value set
to YES.
Mono
The Siren 22, G.719, and SirenLPR mono audio algorithms are supported at the following bit rates.
:
Siren22, G.719, and SirenLPR Mono vs Bitrate
Audio Algorithm
Minimum Bitrate (kbps)
Siren22 64k
Siren22 48K
Siren22_32k
G.719_64k
384
G.719_48k
G.719_32k
G.728 16K
G.719_64k
384
SirenLPR_48k
256
Siren22_48K
G.719_48k
G.7221C_48k
Siren14_48k
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Siren22, G.719, and SirenLPR Mono vs Bitrate
Audio Algorithm
Minimum Bitrate (kbps)
SirenLPR_32k
Siren22_32k
G.719_32k
128
G.7221C_32k
Siren14_32k
SirenLPR
64
SirenLPR
48
SirenLPR
32
Siren7
16
Stereo
The Siren 22Stereo, G.719Stereo, and SirenLPR audio algorithms are supported at the following bit rates.
Siren22Stereo, G.719Stereo, and SirenLPR vs Bitrate
Audio Algorithm
Minimum Bitrate (kbps)
Siren22Stereo_128k
SirenLPRStereo_128k
1024
G.719Stereo_128k
Siren22Stereo_96k
SirenLPRStereo_96k
512
G.719Stero_96k
Siren14Stero_96k
SirenLPRStereo_64k
G.719Stereo_64k
384
Siren22Stereo_64k
Siren14Stereo_64k
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Audio algorithms supported for ISDN
Supported Audio Algorithm vs Bitrate
Audio Algorithm
Minimum Bitrate (kbps)
G.722.1C 48K
G.722.1C 32K
G.722.1C 24K
Siren14 48K
256
Siren14 32K
Siren14 24K
G.722.1 32K
G.722.1 24K
G.722.1 16K
G.722 48K
G.722 56K
G.722 64K
256
G.711 56K
G.711 64K
G.728 16K
G.722.1C 32K
G.722.1C 24K
Siren14 32K
Siren14 24K
G.722.1 32K
G.722.1 24K
128
G.722 48K
G.722 56K
G.722 64K
G.711 56K
G.711 64K
G.728 16K
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Supported Audio Algorithm vs Bitrate
Audio Algorithm
Minimum Bitrate (kbps)
G.722.1 16K
G.722.1C 24K
Siren14 24K
G.722 48K
G.722 56K
96
G.722 64K
G.711 56K
G.711 64K
G.728 16K
G.728 16K
64
Monitoring Participant Audio Properties
The audio algorithm used by the participant’s endpoint can be verified in the Participant Properties Channel Status dialog box.
To view the participant’s properties during a conference:
1 In the Participants list, right click the desired participant and select Participant Properties.
2 Click the Channel Status - Advanced tab.
The Participant Properties - Channel Status - Advanced dialog box is displayed.
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3 In the Channel Info field, select Audio In or Audio Out to display the audio parameters.
4 Click the OK button.
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Automatic Muting of Noisy Endpoints
Note: Feature Applicability
• Automatic muting of noisy endpoint is not applicable for Collaboration Servers 1800 / Virtual
Edition.
• Automatic muting of noisy endpoints is applicable only in AVC video conferences and audio-only
(non-SAC) conferences.
The Collaboration Server can detect AVC-enabled endpoints with a noisy audio channel and automatically
mute them, reducing the noise heard by other conference participants. When the auto muted endpoint
becomes the speaker the endpoint is automatically un-muted by the system. If the speaker halts his/her
conversation and the line still emits noises, the endpoint will be automatically muted again.
When the endpoints are automatically muted by the MCU, no indication is displayed in the Collaboration
Server Web Client or RMX Manager as the system does not consider it as a deliberate mute.
Guidelines for Automatically Muting Noisy Endpoints
The automatic muting of noisy AVC-enabled endpoints can be used according to the following guidelines:
● The Auto mute noisy endpoints check box in the Profile Properties - Audio Settings dialog box is
enabled only when the ENABLE_SELECTIVE_MIXING flag is set to YES (default).
● It affects only AVC-based and audio only endpoints (non-SAC endpoints)
● It does not affect SVC-based endpoints
● It is supported in CP conferences and in Mixed CP and SVC conferences.
● In a mixed CP and SVC conferences, only the AVC-based endpoints can be automatically muted. If
the noisy endpoint is SVC-based, its audio channel will not be sent to the AVC-based endpoints, but
it will be sent to the other SVC-based endpoints.
● MCU reset is not required when changing the ENABLE_SELECTIVE_MIXING flag setting.
● When upgrading from a version prior to 8.1, the Auto mute noisy endpoints option is not automatically
enabled in the existing Profiles and it has to be manually enabled, if required.
In new Profiles that are created after the upgrade, the Auto mute noisy endpoints option is
automatically enabled.
● If your conferencing environment includes the Polycom DMA, the conferences that are started from
the DMA will not include the Auto mute noisy endpoints parameter as it is not part of the DMA Profiles.
In such a case, when the parameter setting is unknown, the system will enable or disable the
automatic muting of noisy endpoints according to the flag setting - if the flag is set to YES, it will be
enabled in the conference.
● Sounds with the frequency of noise, such as applause and certain background music, may be also
be muted.
The following table summarizes the state (enabled or disabled) of the Automatic muting of noisy endpoints
feature depending on the ENABLE_SELECTIVE_MIXING flag setting and the Auto mute noisy endpoints
setting in the Profile Properties - Audio Settings:
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Conditions for enabling/disabling the automatic muting of noisy endpoints
ENABLE_SELECTIVE_MIXING flag
Setting
Auto mute noisy endpoints
setting
Automatic muting of noisy
endpoints State
YES
Yes (check box selected)
Enabled
YES
No (check box cleared)
Disabled
YES
Unknown (for example, the
conference is started from the
DMA)
Enabled
NO
Yes (check box selected)
Disabled
NO
No (check box cleared)
Disabled
NO
Unknown (for example, the
conference is started from the
DMA)
Disabled
Automatic Muting of Noisy Endpoints
The automatic muting of noisy endpoints can be enabled or disabled at the conference level (in the
Conference Profile) or at the system level, by changing the ENABLE_SELECTIVE_MIXING flag setting.
In new MCU installations, the automatic muting of noisy endpoints is automatically enabled on the MCU as
the ENABLE_SELECTIVE_MIXING flag is set to YES and the Auto mute noisy endpoints check box in the
Profile Properties - Audio Settings tab is selected.
You can disable the automatic muting of noisy endpoints by either setting the system flag to NO or clearing
the Auto mute noisy endpoints check box in the Profile Properties - Audio Settings tab.
If required, it is recommended to disabled the automatic muting of noisy endpoints at the conference level,
in the conference Profile without changing the flag settings.
In existing MCU sites, following the software upgrade the automatic muting of noisy endpoints is disabled
at the conference level in the existing conference Profile and has to be manually enabled in these profiles.
This option is automatically enabled when creating a new Profile.
Automatic Muting of Noisy Endpoints at the Conference Level
If the ENABLE_SELECTIVE_MIXING flag is set to YES, the automatic muting of noisy endpoints can be
enabled or disabled at the conference level in the Conference Profile - Audio Settings dialog box.
If the ENABLE_SELECTIVE_MIXING flag is set to NO, the automatic muting of noisy endpoints is disabled
at the conference level and cannot be enabled in the Conference Profile - Audio Settings dialog box.
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Additional Conferencing Information
To disable/enable the automatic muting of noisy endpoints in the Conference Profile:
1 In a new or existing Conference Profile, click the Audio Settings tab.
 In new Profiles, the Auto mute noisy endpoints check box is selected by default.
 In existing profiles (after software upgrade from a version prior to 8.1), the Auto mute noisy
endpoints check box is cleared.
2 To enable the automatic muting of noisy endpoints, click the Auto mute noisy endpoints check
box.
3 Click OK.
Automatic Muting of Noisy Endpoints at the MCU Level
You can disable the automatic muting of noisy endpoints at the MCU level by changing the
ENABLE_SELECTIVE_MIXING flag setting to NO.
In such a case, the automatic muting of noisy endpoints at the conference level (in the Conference Profile
- Audio Settings dialog box) is disabled.
To modify the system flag setting:
● To modify ENABLE_SELECTIVE_MIXING flag setting to NO, manually add it to system.cfg file and
set its value to NO.
For more details, see the Modifying System Flags.
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Permanent Conference
A Permanent Conference is any ongoing conference with no pre-determined End Time continuing until it is
terminated by an administrator, operator or chairperson.
Guidelines
● Resources are reserved for a Permanent Conference only when the conference has become
ongoing.
● Resources are allocated to a Permanent Conference according to the Reserve Resources for Video
Participants field. If the number of defined dial-out participants exceeds the value of this field, the
Collaboration Server automatically replaces the number in the Reserve Resources for Video
Participants field with the number of defined dial-out participants in the Permanent Conference.
● Auto Terminate is disabled in Permanent Conferences.
● If participants disconnect from the Permanent Conference, resources that were reserved for its video
and audio participants are released.
● Entry Queues, Conference Reservations and SIP Factories cannot be defined as Permanent
Conferences.
● Additional participants can connect to the conference, or be added by the operator, if sufficient
resources are available.
● The maximum size of the Call Detail Record (CDR) for a Permanent Conference is 1MB.
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Additional Conferencing Information
Enabling a Permanent Conference
The Permanent Conference option is selected in the New Conference, New Meeting Room or New
Conference Templates dialog boxes.
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Closed Captions
Note: Feature Applicability
• This option is not supported for Collaboration Servers 1800 / Virtual; Edition.
• This feature is applicable only for AVC endpoints.
Endpoints can provide real-time text transcriptions or language translations of the video conference by
displaying captions. The captions for a conference may be provided by the captioner who is present in the
conference, or the captioner may use a telephone or web browser to listen to the conference audio. When
the captioner sends a unit of text, all conference participants see it on the main monitor for 15 seconds. The
text then disappears automatically.
The captioner may enter caption text using one of the following methods:
● Remotely, via a dial-up connection to the system’s serial RS-232 port.
● In the room using equipment connected directly to the serial port.
● In the room or remotely, using the Polycom HDX web interface.
Closed Captions Guidelines
● The captions display properties are configured on the endpoint sending the captions.
● Closed Captions content is defined from the endpoint. The Collaboration Server only transmits it to
the endpoints.
● When enabled, captions are available to all endpoints supporting FECC.
● Captions are supported in H.323 and SIP connections.
● The FECC indications during ongoing conferences are used when sending captions.
● When Closed Captions option is enabled for the MCU, muting an endpoint may cause the display of
the Far Mute indication on all the screens of the endpoints connected to the conference.
● The Closed Captions option is not supported in cascading conferences (captions they can only be
viewed in the local conference) as FECC is not supported in cascading links.
● Site name display is not affected by captions display.
● Captions are supported by the Collaboration Server in the following configurations and conferencing
modes:
 CP Conferencing Mode.
 Encrypted and non-encrypted conferences.
 Conferences with content sharing.
Enabling Closed Captions
Captions are enabled by a system flag. By default, Closed Captions are disabled.
To change the flag value:
1 On the Collaboration Server menu, click Setup > System Configuration.
The System Flags dialog box opens.
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Additional Conferencing Information
2 In the MCMS_PARAMETERS tab, click the New Flag button.
The New Flag dialog box is displayed.
3 In the New Flag field enter ENABLE_CLOSED_CAPTION.
4 In the Value field enter YES to enable or NO to disable Closed Captions display.
5 Click OK to close the New Flag dialog box.
The new flag is added to the flags list.
6 Click OK to close the System Flags dialog box.
Note: System Restart Requirement
For flag changes (including deletion) to take effect, reset the MCU. For more information, see
Resetting the Collaboration Server.
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Defining Cascading Conferences
Cascading enables administrators to connect one conference directly to one or several conferences,
depending on the topology, creating one large conference. The conferences can run on the same MCU or
different MCUs.
There are many reasons for cascading conferences, the most common are:
● Connecting two conferences on different MCUs at different sites.
● Utilizing the connection abilities of different MCUs, for example, different communication protocols,
such as, serial connections and ISDN, etc.
Conferences are Cascaded when a link is created between two conferences, usually running on two
different MCUs.
Note: Cascading support
• Cascading information applies to AVC Conferencing Mode (CP and mixed CP and SVC) only.
• Cascading Conferences are not supported by Collaboration Server (RMX) 1800 with no DSP cards,
and Collaboration Server 1800, Entry Level.
Cascading Link Properties
Cascade links are treated as endpoints in CP conferences. They are allocated resources as any other
endpoint according to Default Minimum Threshold Line Rates and Resource Usage Summary.
They transmit audio, video and content between conferences as well as DTMF codes input from other
endpoints in the conference.
Setting the Video Layout in Cascading conferences require additional consideration.
Setting the Video Layout in Cascading conferences
When cascading two conferences, the video layout displayed in the cascaded conference is determined by
the selected layout in each of the two conferences. Each of the two conferences will inherit the video layout
of the other conference in one of their windows.
In order to avoid cluttering in the cascaded window, it is advised to select appropriate video layouts in each
conference before cascading them.
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Video Layouts in Cascaded Conferences
Conference A
Conference B
Video layout of conference
B within window of
conference A
Video layout of conference
A within window of
conference B
Without cascading
During cascading
Guidelines for Setting the Video Layout in Cascading Conferences
To ensure that conferences can be cascaded and video can be viewed in all conferences the following
guidelines are recommended:
● The same version installed on all MCUs participating the cascading topology
● The same license installed on all MCUs participating the cascading topology
● Same Conference Parameters are defined in the Profile of the conferences participating in the
cascading topology
 Conference line rates should be identical
 Content rate should be identical
 Same encryption settings
● DTMF codes should be defined with the same numeric codes in the IVR services assigned to the
cascading conferences
● DTMF forwarding is suppressed
● The video layout of the link is set to 1x1 by the appropriate system flag.
● Cascaded links in 1x1 video layout are in SD resolution.
● When the Mute Participants Except Lecturer option is enabled in the Conference Profile, all
participants (including the link participants) except the lecturer are muted. Only the lecturer is not
muted.
● Gathering is not supported in Cascading Conferences.
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Flags Controlling Cascading Layouts
● Setting the FORCE_1X1_LAYOUT_ON_CASCADED_LINK_CONNECTION System Flag to YES
(default) automatically forces the cascading link to Full Screen (1x1) in CP conferences, hence
displaying the speaker of one conference to a full window in the video layout of the other conference.
Set this flag to NO when cascading between an RealPresence Collaboration Server and an MCU that
is functioning as a Gateway, if the participant layouts on the MCU are not to be forced to 1X1.
● Setting the AVOID_VIDEO_LOOP_BACK_IN_CASCADE System Flag to YES (default) prevents the
speaker’s image from being sent back through the participant link from the cascaded conference.
This can occur in cascaded conferences with conference layouts other than 1x1. It results in the
speaker’s own video image being displayed in the speaker’s video layout.
This option is supported with:
 In IP (H.323, SIP) and ISDN environments.
 Basic Cascading of Continuous Presence and Video Switched conferences. If a Master MCU has
two slave MCUs, participants connected to the slave, MCUs will not receive video from each other
until a video end point connects to the Master MCU.
 Video resolution will be according to the Resolution Configuration, or VSW profile.
For more details on defining system flags, see Modifying System Flags.
DTMF Forwarding
When two conferences are connected over an IP link, DTMF codes from one conference are not forwarded
to the second conference with the exception of the following operations that are available throughout the
conference and the forwarding of their DTMF codes is not suppressed (i.e. they will apply to both
conferences):
● Terminate conference.
● Mute all but me.
● Unmute all but me.
● Secure conference.
● Unsecure conference.
Note: DTMF Codes Forwarding
During cascading between a gateway and a conference all DTMF codes are forwarded from the
gateway to the conference and vice versa.
Play Tone Upon Cascading Link Connection
The RealPresence Collaboration Server can be configured to play a tone when a cascading link between
conferences is established. The tone is played in both conferences.
This tone is not played when the cascading link disconnects from the conferences.
The tone used to notify that the cascading link connection has been established cannot be customized.
The option to play a tone when the cascading link is established is enabled by setting the System Flag
CASCADE_LINK_PLAY_TONE_ON_CONNECTION to YES.
Default value: NO.
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The tone volume is controlled by the same flag as the IVR messages and tones:
IVR_MESSAGE_VOLUME.
Possible Cascading Topologies
The following cascading topologies are available for setting cascading conferences:
● Basic Cascading - only two conferences are connected (usually running on two different
RealPresence Collaboration Servers). The cascaded MCUs reside on the same network.
● Star Cascading - one or several conferences are connected to one master conference. Conferences
are usually running on separate MCUs. The cascaded MCUs reside on the same network.
● MIH (Multi-Hierarchy) Cascading - several conferences are connected to each other in
Master-Slave relationship. The cascaded MCUs can reside on different networks.
System configuration and feature availability change according to the selected cascading topology.
Note: Sharing Content in Cascaded Conferences
For properly sharing content in cascaded conferences, predefined dial in and out link participants
must be defined with Master/Slave settings in the conferences.
When Cascading between the Collaboration Server and third party MCUs, the participant defined in
the Collaboration Server conference must be defined as Master.
When cascading between the Collaboration Server and a Codian MCU, set the system flag
ENABLE_CODIAN_CASCADE to YES to ensure that the Collaboration Server is defined as a Master
in cascading conferences at all times.
Basic Cascading
In this topology, a link is created between two conferences, usually running on two different MCUs. The
MCUs are usually installed at different locations (states/countries) to save long distance charges by
connecting each participant to their local MCU, while only the link between the two conferences is billed as
long distance call.
● This is the only topology that enables both IP and ISDN cascading links.
● Cascading between Collaboration Server Virtual Edition and Collaboration Servers (RMX)
2000/1800/4000 is supported.
● When linking two conferences using an IP cascading link:
 The destination MCU can be indicated by:
 IP address
 H.323 Alias
 Both MCUs must be located in the same network.
● The MCU can be used as a gateway.
● The configuration can include two RealPresence Collaboration Servers or one RealPresence
Collaboration Server and one MGC.
● Multiple Cascade Links enabling Cascading between RMXs hosting conferences that include
Immersive Telepresence Rooms (ITP) such as Polycom’s OTX and RPX Room Systems can be
defined. For more information see Creating Multiple Cascade Links Between Telepresence
Conferences.
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Basic Cascading Using IP Cascaded Link
In this topology, both MCUs can be registered with the same gatekeeper or the IP addresses of both MCUs
can be used for the cascading link. Content can be sent across the Cascading Link.
Basic Cascading Topology - IP Cascading Link
For example, MCU B is registered with the gatekeeper using 76 as the MCU prefix.
The connection between the two conferences is created when a dial out IP participant is defined (added) to
conference A whose dial out number is the dial-in number of the conference or Entry Queue running on
MCU B.
Dialing Directly to a Conference
Dial out IP participant in conference A dials out to the conference running on MCU B entering the number
in the format:
[MCU B Prefix/IP address][conference B ID].
For example, if MCU B prefix is 76 and the conference ID is 12345, the dial number is 7612345.
Dialing to an Entry Queue
When dialing to an Entry Queue, the dial out participant dials the MCU B prefix or IP address of MCU B and
the Entry Queue ID in the format:
[MCU B Prefix/IP address][EQ B ID]
For example, if MCU B prefix is 76 and the Entry Queue ID is 22558, the dial number is 7622558.
When the participant from conference A connects to the Entry Queue, the system plays to all the participants
in Conference A the IVR message requesting the participant to enter the destination conference ID.
At this point, the Conference A organizer or any other participant in the conference can enter the required
information for the IVR session using DTMF codes. For example, the meeting organizer enters the
destination conference ID - 12345.
Any DTMF input from conference A is forwarded to the Entry Queue on MCU B to complete the IVR session
and enable the move of the participant to the destination conference B.
Once the DTMF codes are entered and forwarded to the Entry Queue on MCU B, the IVR session is
completed, the participant moved to the destination conference and the connection between the two
conferences is established.
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Automatic Identification of the Cascading Link
In both dialing methods, the system automatically identifies that the dial in participant is an MCU and creates
a Cascading Link and displays the link icon for the participant (
). The master-slave relationship is
randomly defined by the MCUs during the negotiation process of the connection phase.
Basic Cascading Using ISDN Cascaded Link
ISDN connection can be used to link between two MCUs or MCU and gateway and create a cascading
conference. Content can be sent across the ISDN Cascading Link.
Network Topologies Enabling Content Sharing Over ISDN Cascaded Links
ISDN Cascaded links that support content sharing can be created between two gateways, gateway-to-MCU
or between two MCUs in the following network topologies:
● Gateway to Gateway
Gateway to Gateway Topology
In this topology, an IP participant calls another IP participant over an ISDN link between two
gateways.
● Gateway to MCU
Gateway to MCU/ MCU to Gateway Topology
In this topology, an IP participant calls a conference running on an MCU via a gateway and over an
ISDN link.
● MCU to MCU
Cascading Between Two MCUs Using an ISDN Link
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In this topology, an ISDN participant from conference running on MCU A calls a conference running
on MCU B over an ISDN link.
Guidelines
● In Collaboration Servers with versions prior to 8.6, content is restricted, in the sense that when
another endpoint wants to send content, the first endpoint must stop sending content before the
second endpoint can initiate or send content.
● Endpoints that do not support H.239 can receive the Content using the Send Content to Legacy
Endpoints option.
● When a participant joins a conference with active Content, content cannot be viewed by the new
participant. Restart the Content.
● Cascaded MCUs/Gateways must be registered with the same Gatekeeper or neighboring
Gatekeepers. MCUs and endpoints must also be registered with Gatekeepers.
● Gateway/MCU calls require definition of IVR Services. For more information see Defining the IVR
Service for Gateway Calls.
Note: Content Protocol over ISDN Cascaded link
The content sharing protocol is H.263 when sent over ISDN Cascading link.
Gateway to Gateway Calls via ISDN Cascading Link
When H.323 participants connects to another IP participants via a Gateway to Gateway call over an ISDN
link, the dialing string includes the following components:
[GW A prefix in GK] - The prefix with which the RealPresence Collaboration Server (gateway) is
registered to the gatekeeper.
[GW Profile ID] - The ID of the Gateway Profile defined on Gateway A to be used for routing the call to
the Gateway B.
[GW Profile ISDN/PSTN number] - The dial-in number assigned to the Gateway Profile defined on
Gateway B, including the required country and area codes.
Information required that is not part of the dialing string:
[Destination number] - The destination number as alias, IPv4 address or ISDN/PSTN number of
participant B.
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The dialing string format:
H.323 Participants connecting to another IP participant via a Gateway to Gateway call over an ISDN link
enter a dial string using the format:
<GW A Prefix in GK><Gateway Profile_ID on GW A>*<Destination ISDN Dial-in number
assigned to the Gateway Session Profile GW B>*<Destination Number, participant>
For example:
GW A prefix in Gatekeeper - (not used with SIP)
22
Gateway Profile ID in GW A
9999
ISDN Dial-in Number assigned to the Gateway Session
Profile GW B
4444103
IP Participant Alias
3456
H.323 participant dials: 229999*4444103, and when prompted for the Destination number enters 3456
followed by the pound key (#) using DTMF codes
SIP Participants connecting to another IP participant via a Gateway to Gateway call over an ISDN link enter
a dial string using the format:
<Gateway Profile_ID on GW A>@<Central Signaling IP GW A>*<Destination ISDN
Dial-in number assigned to the Gateway Session Profile GW B>*<Destination
Number, participant>
For example:
If Central Signaling IP address of Gateway A is 172.22.177.89, SIP participant dials: 9999@
172.22.177.89* 4444103 and when prompted for the Destination number enters 3456 followed by the
pound key (#) using DTMF codes.
Gateway to MCU Calls via ISDN Cascading Link
When H.323 participants connects to a conference/Meeting Room via a Gateway to MCU call over an ISDN
link, the dialing string includes the following components:
[GW A prefix in GK] - The prefix with which Gateway A is registered to the gatekeeper.
[GW Profile ID on GW A] - The ID of the Gateway Profile on GW A to be used for routing the call to
the Meeting Room/conference running on MCU B.
[Conference/Meeting Room/Entry Queue ISDN/PSTN number] - The dial-in number assigned to
the Entry Queue/Meeting Room/Conference defined on MCU B, including the required country and area
codes.
Information required that is not part of the dialing string:
[Destination Conference ID] - Only if using the Entry Queue on MCU B for routing calls or creating
new ad hoc conferences. The ID of the destination conference on MCU B.
The dialing string format:
<GW A Prefix in GK><Gateway Profile_ID on GW A>*<ISDN Number assigned to the
Meeting Room/Conference/Entry Queue>
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For Example:
GW A prefix in Gatekeeper - (not used with SIP)
22
Gateway Profile ID in GW A
9999
ISDN Dial-in Number assigned to the Entry Queue/MR/conference
4444100
H.323 participant dials
229999*4444100
SIP participant dials (if Central Signaling IP address of Gateway A is 172.22.177.89) 9999@
172.22.177.89 IP* 4444100
If dialing an Entry Queue, when prompted for the Destination number enters 3456 followed by the pound
key (#) using DTMF codes to create a new conference or join an ongoing conference with that ID.
MCU to MCU Calls via ISDN Cascading Link
A dial out ISDN participant is defined (added) to conference A running on MCU A. The participant’s dial out
number is the dial-in number of the Entry Queue or conference running on MCU B (for example 54145106).
MCU A dials out to an Entry Queue or conference B running on MCU B using the Entry Queue number (for
example 54145106) or the conference number.
When the participant, who is a dial-in participant in conference B, connects to the Entry Queue, the system
plays to all the participants in Conference A the IVR message requesting the participant to enter the
destination conference ID (or if connecting to a conference directly, the participant is requested to enter the
conference password).
At this point the Conference A organizer or any other participant in the conference can enter the required
information for the IVR session using DTMF codes. For example, the meeting organizer enters the
destination conference ID - 12345.
Any DTMF input from conference A is forwarded to the Entry Queue on MCU B to complete the IVR session
and enable the move of the participant to the destination conference B.
Once the DTMF codes are entered and the IVR session is completed, the participant is connected to the
conference and the connection between the conferences is established. The system automatically identifies
the calling participant as an MCU and the connection is identified as a cascading link and the cascading link
icon is displayed for the participant (
).
RealPresence Collaboration Server Configuration Enabling ISDN Cascading Links
To enable Gateway-to-Gateway, Gateway-to-MCU and MCU-to-MCU calls over ISDN Cascading links, the
following configurations are required:
● Modifying the IP Network Service to include the MCU Prefix in the Gatekeeper (in the Gatekeepers
dialog box). For more details, see Modifying the Default IP Network Service.
● ISDN Network Service is configured in both MCUs. For more details, Modifying an ISDN/PSTN
Network Service.
● Configuring a Gateway Profile and assigning dial-in ISDN/PSTN numbers. For details, see Defining
the Gateway Profile.
● Configure the Entry Queue or conference (for direct dial-in) as enabled for ISDN connection and a
dial-in number is assigned (for example 54145106).
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● Defining the dial-in ISDN participant in MCU B and Dial-out ISDN participant in MCU A (for
MCU-to-MCU cascading conferences).
A dial out ISDN participant is defined (added) to conference A. The participant’s dial out number is
the dial-in number of the Entry Queue or conference running on MCU B (for example 54145106).
MCU A dials out to an Entry Queue or conference B running on MCU B using the Entry Queue
number (for example 54145106) or the conference number.
Conference Profile Definition
The following table lists the recommended Meeting Room/Conference Profile parameters setting when
routing ISDN cascaded calls.
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Recommended Conference Profile Options Setting
Line Rate
Motion
128

Sharpness
Encryption
LPR

128
128

128

256





256
256

256

384





384
384

384

512





512
512

512

768





768
768

768




Note: Line Rate Recommendation
Since the remote participant settings are unknown, it is recommended that the gateway or endpoint
be configured to support a higher line rate (for example, 768 Kbps) to allow flexibility during endpoint
capability negotiations.
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MCU Interoperability Table
The following table lists the different MCU and Gateway configurations that are supported or implemented
when routing Cascaded ISDN calls.
MCU Interoperability Table
Source
Destination
Scenario
Version(s)
RealPresence
Collaboration
Server Gateway
RealPresence
Collaboration
Server MCU
User calls via a Gateway to a Remote
Conference (user to conference)
RealPresence Collaboration
Server v. 7.1 or later
RealPresence
Collaboration
Server Gateway
RealPresence
Collaboration
Server Gateway
User calls via a Gateway to a Remote
User behind Gateway (user to user)
RealPresence Collaboration
Server v. 7.1 or later
RealPresence
Collaboration
Server MCU
RealPresence
Collaboration
Server MCU
A dial out participants calls to a remote
conference (conference to conference)
RealPresence Collaboration
Server v. 7.1 or later
RealPresence
Collaboration
Server MCU
RealPresence
Collaboration
Server Gateway
A dial out participants calls to a remote
User behind a Gateway (Conference to
User)
RealPresence Collaboration
Server v. 7.1 or later
Endpoint
RealPresence
Collaboration
Server Gateway
User calls directly to a remote user behind
a Gateway (User to User)
RealPresence Collaboration
Server v. 7.1
RealPresence
Collaboration
Server MCU
Codian
Gateway
Dial out participants use a fixed rule
behind the Codian Gateway.
RealPresence Collaboration
Server v. 7.1
Latest Codian version
RealPresence
Collaboration
Server Gateway
Codian
Gateway
Dial out participants use a fixed rule
behind the Codian Gateway.
RealPresence Collaboration
Server v. 7.1
Latest Codian version
Codian Gateway
RealPresence
Collaboration
Server MCU
User calls via a Codian Gateway to a
Remote Conference (user to conference)
RealPresence Collaboration
Server v. 7.1
Latest Codian version
Codian Gateway
RealPresence
Collaboration
Server Gateway
User calls via a Codian Gateway to a
Remote User behind RealPresence
Collaboration Server Gateway (user to
user)
RealPresence Collaboration
Server v. 7.1
Latest Codian version
RealPresence
Collaboration
Server MCU
Radvision
Gateway
User calls via a Radvision Gateway to a
Remote User behind RealPresence
Collaboration Server Gateway (user to
user)
RealPresence Collaboration
Server v. 7.1
Latest Radvision version
RealPresence
Collaboration
Server Gateway
Radvision
Gateway
User calls via a Radvision Gateway to a
Remote User behind RealPresence
Collaboration Server Gateway (user to
user)
RealPresence Collaboration
Server v. 7.1
Latest Radvision version
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MCU Interoperability Table
Source
Destination
Scenario
Version(s)
Radvision
Gateway
RealPresence
Collaboration
Server MCU
User calls via a Radvision Gateway to a
Remote Conference (user to conference)
RealPresence Collaboration
Server v. 7.1
Latest Radvision version
Radvision
Gateway
RealPresence
Collaboration
Server Gateway
User calls via a Radvision Gateway to a
Remote User behind RealPresence
Collaboration Server Gateway (user to
user)
RealPresence Collaboration
Server v. 7.1
Latest Radvision version
Endpoint
RealPresence
Collaboration
Server Gateway
User calls directly to a DMA controlled
environment
RealPresence Collaboration
Server v. 7.1
RealPresence
Collaboration
Server MCU
RealPresence
Collaboration
Server Gateway
A dial out participants calls to a remote
conference on a DMA controlled
environment
RealPresence Collaboration
Server v. 7.1
Notes:
• On the Codian gateway Content is not supported with line rates of 128Kbps and below.
• To send Content from a participant over Radvision Gateway to a conference/participant, the
GWP20 patch must be installed in the RadVision gateway:
On the Radvision gateway, open the GWP20 User Interface.
Select Settings > Advanced Commands.
In the Command box, enter H239OlcPatch.
In the Parameters box, enter Enable and click Send.
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Suppression of DTMF Forwarding
Forwarding of the DTMF codes from one conference to another over an ISDN cascading link is not
automatically suppressed as with IP cascading link and it can be limited to basic operations while
suppressing all other operations by a system flag DTMF_FORWARD_ANY_DIGIT_TIMER_SECONDS.
System Flag Settings
The DTMF_FORWARD_ANY_DIGIT_TIMER_SECONDS flag determines the time period (in seconds) that
MCU A will forward DTMF inputs from conference A participants to MCU B.
Once the timer expires, most of the DTMF codes (excluding five operations as for IP links) entered in
conference A will not be forwarded to conference B. This is done to prevent an operation requested by a
participant individually (for example, mute my line) to be applied to all the participants in conference B.
Flag range (in seconds): 0 - 360000
This flag is defined on MCU A (the calling MCU).
If a flag is not listed in the System Flags list it must be added to the system.cfg file before it can be modified.
For more details on defining system flags, see Modifying System Flags.
Star Cascading Topology
In the Star topology (as well as in the Basic topology), the MCUs are usually installed at different locations
(states/countries) and participants connect to their local MCU to facilitate the connection and save long
distance call costs. Star Topology Cascading requires that all cascaded MCUs reside on the same network.
Note: H.323 Cascaded Link Requirement
Although participants in Star Cascading conferences can connect to their local conference using IP
(H.323, SIP, and ISDN), the Cascading Links between conferences must connect via H.323.
Content sharing is available to all conferences over the H.323 Cascading Link. Snatching content token
between different Collaboration Servers is supported over an H.323 cascading link, provided all
RealPresence Collaboration Servers are:
● Version 8.6 and up.
● Configured with Exclusive Content Mode set to off.
● The system flag ENABLE_CONTENT_SNATCH_OVER_CASCADE is set to YES.
Note that when a Master MCU is not in Exclusive Content Mode, another participant in the same MCU may
snatch the content, even if the value of the ENABLE_CONTENT_SNATCH_OVER_CASCADE system flag
is NO.
In this topology, the MCUs are networked together using two modes:
● Master-Slave Cascading
● Cascading via Entry Queue
Master-Slave Cascading
It is similar to MIH (Multi Hierarchy) cascading, with only two levels: one Master MCU on level 1 and several
Slave MCUs on level 2.
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The cascading hierarchy topology can extend to four levels (MIH Cascade - a Sample 3-Level Cascading
Configuration) and should be deployed according to the following guidelines:
● If an RealPresence Collaboration Server is deployed on level 1:
 RealPresence Collaboration Server systems can be used on level 2
 MGC with version 9.0.4 can be used on level 2 if RealPresence Collaboration Server version 7.0.2
and higher is deployed in level 1
● If an MGC is deployed on level 1:
 MGC or RealPresence Collaboration Server can be used on level 2.
Master-Slave Star Cascading Topology
● When creating a cascading link between two RealPresence Collaboration Servers:
 The RealPresence Collaboration Servers operate in CP (Continuous Presence) mode.
● When creating a cascading link between MGCs and RealPresence Collaboration Servers:
 The MGCs can only operate in VSW mode.
The following table summarizes Video Session Modes line rate options that need to be selected for each
conference in the cascading hierarchy according to the cascading topology:
MIH Cascading – Video Session Mode and Line Rate
Topology
MCU Type
Video Session Mode
Line Rate
Endpoint
Level 1
RealPresence
Collaboration Server
CP - HD
1.5Mb/s, 1Mb/s, 2Mb/s
HDX
Level 2
RealPresence
Collaboration Server
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MIH Cascading – Video Session Mode and Line Rate
Topology
MCU Type
Video Session Mode
Line Rate
Endpoint
Level 1
RealPresence
Collaboration Server
CP - CIF
768Kb/s, 2Mb/s
VSX
Level 2
RealPresence
Collaboration Server
Level 1
MGC
CP - CIF 263
768Kb/s, 2Mb/s
HDX, VSX
Level 2
RealPresence
Collaboration Server
CP - CIF 264
Level 1
MGC
VSW - HD
1.5Mb/s
HDX
Level 2
RealPresence
Collaboration Server
VSW - HD
To establish the links between two RealPresence Collaboration Servers requires the following procedures
be performed:
● Establish the Master-Slave relationships between the cascaded conferences by defining the dialing
direction.
● Create the Master and Slave conferences, defining the appropriate line rate.
● Create a cascade-enabled Dial-out Participant link in the Master conference
● Create a cascade-enabled Dial-in Participant link in the Slave conference.
Creating a Cascade Enabled Dial-out/Dial-in Participant Link
The connection between two cascaded conferences is established by a cascade enabled dial-out and dial-in
participants, acting as a cascades link.
The dialing direction determines whether the dial-out participant is defined in the conference running on the
Master MCU or the Slave MCU. For example, if the dialing direction is from the Master conference on level
1 to the Slave conference on level 2, the dial-out participant is defined in the Master conference on level 1
and a dial-in participant is defined in the Slave conference running on the MCU on level 2.
If the cascade-enabled dial-out participant always connects to the same destination conference on the other
(second) MCU, the participant properties can be saved in the Address Book of the MCU for future repeated
use of the cascaded link.
To define the dial-out cascade participant link:
1 In the Conferences pane, select the conference.
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2 In the Participants pane, click New Participant (
).
The New Participant - General dialog box is displayed.
3 Define the following parameters:
New Participant – Dial-out Cascade Link
Field
Description
Name
Enter the participant’s name.
This field may not be left blank. Duplicate participant names, comma, and
semi-colon characters may not be used in this field.
Dialing Direction
Select Dial-out.
Type
Select H.323.
IP Address
Enter the IP address of the Signaling Host of the MCU running the other (second)
conference, where the cascade enabled Entry Queue is defined.
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New Participant – Dial-out Cascade Link
Field
Description
Alias Name
If you are using the target MCU IP address, enter the Conference ID of the target
conference. For example: 24006
If a gatekeeper is used, instead of the IP address, you can enter the prefix of the
target MCU as registered with the gatekeeper, as part of the dialing string and the
conference ID in the format:
<Target MCU Prefix><Conference_ID>
For example: 92524006
If the conference has a password and you want to include the password in the dial
string, append the password to in the dial string after the Conference ID.
For example: 92524006##1234
If the conference has a password and you do not want to include the password in
the dial string, set the
ENABLE_CASCADED_LINK_TO_JOIN_WITHOUT_PASSWORD flag to YES.
For more information see Modifying System Flags.
Alias Type
Select E.164 (digits 0-9, *, #).
4 Click the Advanced tab.
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5 In the Cascade field, select:
 Slave (if the participant is defined in a conference running on a Slave MCU).
 Master (if the participant is defined in a conference running on the Master MCU).
6 Click OK.
To define a Dial-in Participant as the cascade link:
1 In the Participants list, click the New Participant button(
).
The New Participant - General dialog box opens.
2 Define the following parameters:
New Participant – Dial-in Cascade Link
Field
Description
Display Name
Enter the participant’s name.
This field may not be left blank. Duplicate participant names, comma, and
semi-colon characters may not be used in this field.
Dialing Direction
Select Dial-in.
Type
Select H.323.
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New Participant – Dial-in Cascade Link (Continued)
Field
Description
IP Address
If a gatekeeper is used: This field is left empty.
If a gatekeeper is not used: Enter the IP address of the Signaling Host of the
MCU running the other conference.
Alias Name
If a gatekeeper is used: Enter the name of the other (second) conference.
If a gatekeeper is not used: Enter the ID of the MCU running the other (second)
conference.
Alias Type
If a gatekeeper is used: H.323 ID
If a gatekeeper is not used: Select E.164 (digits 0-9, *, #).
3 Click the Advanced tab.
The Advanced dialog box opens.
4 In the Cascaded Link field, select:
 Slave, if the participant is defined in a conference running on a Slave MCU.
 Master, if the participant is defined in a conference running on the Master MCU.
5 Click the OK button.
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Cascading via Entry Queue
The link between the two conferences is created when a participant that is defined as a dial-out cascaded
link in one conference (Conference A) connects to the second conference (Conference B) via a special
cascaded Entry Queue (EQ). When MCU A dials out to the cascaded link to connect it to conference A, it
actually dials out to the cascaded Entry Queue defined on MCU B.
Cascaded Conferences - Star Topology
Though the process of cascading conferences mentioned in this section refers to conferences running on
two different RealPresence Collaboration Server units, it is possible to cascade conferences running
between RealPresence Collaboration Server units and other MCUs.
The following features are not supported by the cascaded link and therefore are not supported in the
combined conference:
● DTMF codes are enabled in cascaded conference, but only in their local conference. The operations
executed via DTMF codes are not forwarded between linked conferences.
● FECC (Far End Camera Control will only apply to conferences running in their local MCU).
Enabling Cascading
Cascading two conferences requires that the following procedures are implemented:
● Creating the cascade-enabled Entry Queue
A cascade-enabled Entry Queue must be created in the MCU hosting the destination conference
(Conference B). The cascade-enabled Entry Queue is used to establish the dial-in link between the
destination conference and the linked conference and bypassing standard Entry Queue, IVR prompt
and video slide display.
● Creating a cascade-enabled Dial-out link
The creation of a cascade-enabled dial-out link (participant) in the linked conference (Conference A).
This dial-out participant functions as the link between the two conferences.
● (Optional) Enabling the cascaded linked participant to connect to the linked conference (Conference
A) without entering the conference password. This can be done by modifying the default settings of
the relevant system flag.
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Creating the Cascade-enabled Entry Queue
Note: Limitation on Cascade-Enabled Entry Queue
The cascade-enabled Entry Queue should be used only to connect cascaded links and should not be
used to connect standard participants to conferences.
The cascade-enabled Entry Queue maintains the correct behavior of the cascaded link when it dials into it.
When cascading High Definition (HD) conferences, the cascade-enabled Entry Queue must have the same
settings as both cascaded conferences and the participants in both conferences must use the same line rate
and HD capabilities as set for the conferences and Entry Queue.
To Define a Cascade-Enabled Entry Queue:
1 In the RealPresence Collaboration Server Management pane, click the Entry Queues button.
The Entry Queues list pane is displayed.
2 Click the New Entry Queue
button.
The New Entry Queue dialog box is displayed.
3 Define the standard Entry Queue parameters (as described in Defining a New Entry Queue).
4 In the Cascade field, select Master or Slave depending on the Master/Slave relationship.
 Set this field to Master if the Entry Queue is defined on the MCU that is at the center of the
topology and other conferences dial into it (acting as the Master).
 Set this field to Slave if the Entry Queue is defined on the MCU acting as a Slave, that is, to which
the link from the Master MCU (MCU at the center of the topology) is dialing.
If you are defining an HD cascaded Entry Queue, it is recommended to select the same Profile that
is selected for both conferences.
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5 Click OK.
The new Entry Queue enabling cascading is created.
Creating the Dial-out Cascaded Link
The dial-out link (participant) is created or added in the linked conference (Conference A). The dial-out string
defined for the participant is the dialing string required to connect to the destination conference (Conference
B) Entry Queue defined on the MCU hosting the destination cascaded conference. The dial-out participant
can be defined in the Address Book and added to the conference whenever using the same
cascade-enabled Entry Queue and a destination conference (with the same ID and Password).
To define the Dial-out Cascaded Link:
1 Display the list of participants in the linked conference (Conference A).
2 In the Participant List pane, click the New Participant
button.
The New Participant - General dialog box is displayed.
3 In the Name field, enter a participant name.
4 In the Dialing Direction field, select Dial-out.
5 In the Type list field, verify that H.323 is selected.
6 There are two methods to define the dialing string:
a Using the MCU’s IP Address and the Alias string - see Method A.
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b Using only the Alias string (requires a gatekeeper) - see Method B.
Method A
In this method no gatekeeper is used.
In the IP Address field, enter the IP address of the Signaling Host of the MCU hosting the
destination conference (in the example, MCU B).
In the Alias Name/Type field, enter the ID of the cascade-enabled Entry Queue (EQ), the Conference
ID and Password of the destination conference (MCU B) as follows:
<EQ ID>#<Destination Conference ID>#<Password> (Password is optional).
For Example: 78485#24006#1234
Cascade-enabled
EQ ID
Destination
Confer
Password (optional)
Method B
In this method a gatekeeper is used:
In the Alias Name field, enter the Prefix of MCU B, EQ ID, Destination Conference ID, and Password,
as follows:
<MCU Prefix EQ ID>#<Conference ID>#<Password> (Password is optional)
For Example: 92578485#24006#1234
MCU Prefix as
registered in
the
gatekeeper
Password (optional)
Cascade-enabled
EQ ID
Conference ID
7 Click the Advanced tab.
8 In the Cascade field, select:
 Slave, if the participant is defined in a conference running on a Slave MCU and will connect to the
Master MCU (in the center of the topology).
 Master, if the participant is defined in a conference running on the Master MCU (in the center of
the topology) dialing from the Master MCU to the Slave MCU.
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9 Click OK.
The cascade-enabled dial-out link is created and the system automatically dials out to connect the
participant to the linked conference, as well as the destination conference.
Enabling Cascaded Conferences without Password
If a password is assigned to the linked conference, cascaded links will be prompted for a password when
connecting to it (Conference A). Administrators have the option of altering the MCU settings to enable
cascaded links to connect without a password.
To enable cascaded links to connect without a password:
1 In the RealPresence Collaboration Server web client connected to MCU A (where the linked
conference is running), click Setup >System Configuration.
The System Flags dialog box opens.
2 Set the ENABLE_CASCADED_LINK_TO_JOIN_WITHOUT_PASSWORD flag to YES.
3 Click OK.
For more information, see Modifying System Flags.
Reset the MCU for flag changes to take effect.
Monitoring Star Cascaded Conferences
To monitor both conferences at the same time, two instances of the RealPresence Collaboration Server
Web Clients must be opened (one for each MCU) by entering the IP Address of each MCU. If both
conferences are running on the same MCU, only one RealPresence Collaboration Server Web Client
window is required.
When conferences are cascaded, the Participant List pane of each of the two conferences will display a
linked icon ( ); a dial-in linked icon in the destination conference (Conference B) and a dial-out linked icon
in the linked conference (Conference A).
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The Conferences List panes in each of the two conferences will display a cascaded conference icon ( )
indicating that a conference running on the MCU is presently cascading with another conference running
on the same or another MCU. The cascaded conference icon will be displayed for a short period of time and
then disappear.
Creating the Dial-out Link from a Conference Running on the MGC to
the Conference Running on the RealPresence Collaboration Server
In the same way that the dial-out cascaded link is created in the RealPresence Collaboration Server, you
can create a dial-out participant in the MGC.
In the MGC Manager application, define a new participant as follows:
1 In the Participant Properties dialog box, enter a Participant Name, select Dial-out and H.323.
2 Define the dialing string as described in step 6 in the procedure for defining the dial-out cascaded
link.
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3 In the Advanced tab, in the Node Type field, select MCU.
4 Click OK.
Cascading Conferences - H.239-enabled MIH Topology
Note: Cascading Topology Applicability for Platform
MIH cascading topology is not applicable for Collaboration Server Virtual Edition.
H.239 Multi-Hierarchy (MIH) cascading is available to RealPresence Collaboration Server users enabling
them to run very large conferences on different MCUs in multiple levels of Master-Slave relationships using
an H.323 connection.
Multi-Hierarchy (MIH) Cascading is implemented where the cascaded MCUs reside on different networks,
whereas Star Topology Cascading requires that all cascaded MCUs reside on the same network.
MIH Cascading allows:
● Opening and using a content channel (H.239) during conferences.
● Full management of extremely large, distributed conferences.
● Connecting conferences on different MCUs at different sites.
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● Utilizing the connection abilities of different MCUs, for example, different communication protocols,
such as, serial connections, ISDN, etc.
● Significant call cost savings to be realized by having participants call local MCUs which in turn call
remote MCUs, long distance.
.
Note: H.323 Cascading Link Requirement
Although participants in MIH Cascading conferences can connect using IP (H.323, SIP) and ISDN, the
MIH Cascading Links must connect via H.323.
MIH Cascading Levels
The cascading hierarchy topology can extend to up to four levels (as shown below), where the most
common configuration includes up to three levels.
MIH Cascade - a Sample 3-Level Cascading Configuration
Cascading Topologies
The cascading hierarchy topology should be deployed according to the following guidelines:
● If an RMX is deployed on level 1 (recommended deployment):
 Any RMX can be used on level 2, 3 and 4 (recommended deployment),
 MGC version 9.0.4 can be used on level 2 and level 3,
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 DST MCS 4000 and other MCUs can be deployed on levels 3 and 4.
● If an MGC is deployed on level 1:
 MGC or RMX can be used on level 2.
 DST MCS 4000 and other MCUs can be deployed on levels 3 and 4.
● DST MCS 4000 MCUs connect as endpoints to the RMXs or MGCs on higher levels.
MIH Cascade Levels
MIH Cascading Guidelines in CP Licensing
Master - Slave Conferences
● It is recommended to have RMX systems at all levels to leverage the high quality video and content
offered by the RMX.
● In MIH Cascading conferences, although there are multiple levels of Master and Slave relationships
between conferences, the conference that runs on the MCU on level 1 of the hierarchy must be the
Master for the entire cascading session. When an MGC is part of the cascading topology, it can be
configured at any level if MGC Version 9.0.4 is installed, otherwise, it must be set as Level 1 MCU.
● Conferences running on MCUs on levels 2 and 3 and can be both Masters and Slaves to conferences
running on MCUs on levels above and below them.
● All conferences running on MCUs on the lowest level in the configuration (for example, level 3 in a
3-level hierarchy configuration) are Slave conferences.
● When the DST MCS 4000 is on level 3 and acting as slave to level 2, the RMX on level 2 must dial
out to it in order for the DST MCS 4000 to be identified as slave. The link between the two MCU (dial
out participant) is defined as a standard participant and not as a cascading link.
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MIH Cascading – Master-Slave Relationship
Video Session Mode, Line Rate and Video Settings
The types of MCUs, their position in the cascade topology and the endpoint capabilities (HD/CIF and
H.263/H.264) determine the Video Session Type of the MIH Cascading conference.
● When creating a cascading link between two RMXs:
 The RMXs operate in CP (Continuous Presence) mode.
 DTMF codes should be defined with the same numeric codes in the IVR services assigned to the
cascading conferences.
● When creating a cascading link between MGCs and RMXs:
 If there are no MGCs on level 2, the MGCs can operate in either in CP or VSW (Video Switching)
mode.
 If there are MGCs on level 2, the MGCs can only operate in VSW mode.
 MGC does not support H.264 High Profile, therefore when MGC is part of the Cascading topology,
do not select High Profile on the RMX system.
 DTMF codes should be defined with the same numeric codes in the IVR services assigned to the
cascading conferences.
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● When creating a cascading link between two MGCs the MGCs must be configured to operate in VSW
mode.
For more details about the MGC to MGC connection, see the MGC Manager User’s Guide, Volume
II, Chapter 1, Ad Hoc Auto Cascading and Cascading Links.
● To enable the connection of the links between cascaded conferences, they must run at the same line
rate.
● To enable Content sharing between the RMX and the MGC, the rate allocated to the content must be
identical in both conferences. Make sure that the line rate set for both conferences, and the Content
Settings (Graphics, Hi-res Graphics or Live video) are selected correctly to ensure the compatible
rate allocation. For more details on the RMX rate allocation to the Content channel, see SIP BFCP
Content Capabilities.
The following table summarizes Video Session Modes line rate options that need to be selected for each
conference in the cascading hierarchy according to the cascading topology:
MIH Cascading – Video Session Mode and Line Rate
MCU Type
Video Session Type
Line Rate
Level 1
RMX
CP - HD
1.5Mb/s, 1Mb/s, 2Mb/s
Level 2
RMX
Level 1
RMX
CP - CIF
768Kb/s, 2Mb/s
Level 2
RMX
Level 1
RMX
CP
768Kb/s, 2Mb/s
Level 2
MGC
CP or VSW
Level 1
MGC
CP - CIF 263
Level 2
RMX
CP - CIF 264
Level 1
MGC
VSW - HD
Level 2
RMX
VSW HD
Level 2
RMX
CP - HD
1.5Mb/s, 1Mb/s, 2Mb/s
Level 3
RMX
Level 2
MGC
VSW*
384 kbps, 768 kbps
Level 3
MGC
Level 2
RMX
CP/VSW -HD
1.5Mb/s, 1Mb/s, 2Mb/s
Level 3
MCS 4000
Level 2
RMX
CP - CIF
768kb/s, 2Mb/s
Level 3
MCS 4000
Topology
768 kb/s, 2Mb/s
1.5Mb/s
* When MGC is on Level 3, Content cannot be shared between Level 2 and Level 3.
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MGC to RealPresence Collaboration Server Cascading
If MGC is running version 9.0.4, and RealPresence Collaboration Server is running version 7.0.2 and higher,
the RealPresence Collaboration Server can be set as Master on level 1 and MGC as Slave on level 2.
MGC running versions other than 9.0.4 is always on level 1 and must be set as the Master MCU.
If the cascading topology includes additional MGCs as well as RealPresence Collaboration Servers it is
recommended to define Video Switching conferences for all the cascading conferences running on the MGC
in the topology.
Two methods can be used to create the Cascading links between conferences running on the RealPresence
Collaboration Server and MGC:
● Method I - Establish the links by defining a dial-in and a dial-out participant in the Slave and Master
conference (where the Master conference is created on the MCU on Level 1 and the Slave
conference is created on the MCU on Level 2).
● Method II - Using a Cascading Entry Queue on either the MGC or the RealPresence Collaboration
Server depending on the dialing direction and the MCU Level. This is recommended when the
RealPresence Collaboration Server is on Level 1.
Method I
Depending on the dialing direction, the following procedures must be performed:
Set up Procedures according to the Dialing Direction
Dialing
Direction
RealPresence Collaboration Server Level 1
MGC to
RealPresence
Collaboration
Server
Set the appropriate flags (done once
only).
Set the appropriate flags (done once only).
Define the conference setting and its line
rate to be the same as the one set on the
RealPresence Collaboration Server.
Define the conference setting and its line rate to
be the same as the one set on the MGC.
Define the dial-in participant (Cascaded
Link) with the calling number from the
MGC.
The alias that will be used to identify the
dial-in participant can be the name of the
calling slave conference.
Set the Cascading option as Master.
Define the dial-out participant (Cascaded Link) to
the conference running on the RealPresence
Collaboration Server. Set the dial-out alias to be
the prefix of the MCU and the name of the master
conference running on the RealPresence
Collaboration Server.
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Set up Procedures according to the Dialing Direction
Dialing
Direction
RealPresence Collaboration Server Level 1
MGC - Level 2
RealPresence
Collaboration
Server to MGC
Set the appropriate flags (done once only)
Set the appropriate flags (done once only)
Define the conference setting and its line
rate to be the same as the one set on the
RealPresence Collaboration Server.
Define the conference setting and its line rate to
be the same as the one set on the MGC.
Define the dial-out participant (Cascaded
Link). Set the dial-out alias to be the prefix
of the MGC and the name of the slave
conference running on the MGC.
Set the Cascading option as Master.
Define the dial-in participant (Cascaded Link) to
the conference running on the RealPresence
Collaboration Server.
The alias that will be used to identify the dial-in
participant can be the name of the calling slave
conference.
For details on the participant definition on the RealPresence Collaboration Server, see Creating a Cascade
Enabled Dial-out/Dial-in Participant Link.
For a detailed description of the participant definition in the MGC, see the MGC Manager User’s Guide,
Volume II, Chapter 1, Cascading Conferences.
Note: Content Sharing between MGC and Collaboration Server
To enable Content sharing between the Collaboration Server and the MGC, the rate allocated to the
content must be identical in both conferences. Make sure that the line rate set for both conferences,
and the Content Settings (Graphics, Hi-res Graphics or Live video) are selected correctly to ensure
the compatible rate allocation. For more details on the Collaboration Server rate allocation to the
Content channel, see SIP BFCP Content Capabilities.
Method II
Depending on the dialing direction, the following procedures must be performed:
Set up Procedures according to the Dialing Direction
Dialing
Direction
MGC to
RealPresence
Collaboration
Server
MGC Level 1
Set the appropriate flags (done once
only).
RealPresence Collaboration Server (RMX)
1800/2000/4000 Level 2
Set the appropriate flags (done once only).
Define the cascade-enabled Entry Queue, setting
it as Slave.
Define the conference setting and its line
rate to be the same as the one set on the
RealPresence Collaboration Server.
Define the conference setting and its line rate to
be the same as the one set on the MGC.
Define the dial-out participant (Cascaded
Link) to the conference running on the
RealPresence Collaboration Server.
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Set up Procedures according to the Dialing Direction
Dialing
Direction
RealPresence
Collaboration
Server to MGC
MGC Level 1
RealPresence Collaboration Server (RMX)
1800/2000/4000 Level 2
Set the appropriate flags (done once only)
Set the appropriate flags (done once only)
Define the cascade-enabled Entry Queue.
Define the conference setting and its line
rate to be the same as the one set on the
RealPresence Collaboration Server.
Define the conference setting and its line rate to
be the same as the one set on the MGC.
Define the dial-out participant (Cascaded Link) to
the conference running on the MGC, setting the
participant Cascade parameter to Slave.
Setting Flags on the RealPresence Collaboration Server
When running conferences in mixed environment (RealPresence Collaboration Server and MGC) there may
be small differences between the line rates each MCU is sending. In the RealPresence Collaboration
Server, several flags must be set to ensure that these differences will not cause the cascaded link to connect
as Secondary and that Content flows correctly between the cascaded conferences. This procedure is
performed once per RealPresence Collaboration Server.
To modify the flags:
1 In the RealPresence Collaboration Server Web Client menu, click Setup > System Configuration.
2 In the System Flags dialog box, add the following new flags and values:
 MIX_LINK_ENVIRONMENT=YES
Setting this flag to YES will adjust the line rate of HD Video Switching conferences run on the
RealPresence Collaboration Server 1800 from 1920Kbps to 18432 Kbps to match the actual rate
of the HD Video Switching conference running on the MGC. In such case, the conference can
include IP and ISDN participants.
 IP_ ENVIRONMENT_LINK=NO
Note: Required System Flags Settings
The values (YES/NO) of the MIX_LINK_ENVIRONMENT and IP_LINK_ENVIRONMENT system
flags should be opposite.
 H263_ANNEX_T=YES (default)
This flag enables/disables the use of Annex T with H263. Set it to NO if the endpoints connecting
to the conference do not support this mode. In such a case, you must also change the MGC flag
ENABLE_H239_ANNEX_T setting to NO.
 FORCE_1X1_LAYOUT_ON_CASCADED_LINK_CONNECTION=YES (default).
Set this flag to NO If the MGC is functioning as a Gateway and participant layouts on the other
network are not to be forced to 1X1.
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3 If the MGC is dialing the RealPresence Collaboration Server and the cascaded link connects to the
conference via the Cascade-enabled Entry Queue without being prompted for the conference
password, set the flag to YES as follows:
ENABLE_CASCADED_LINK_TO_JOIN_WITHOUT_PASSWORD=YES
4 Click OK.
5 Reset the MCU to apply the changes.
Setting Flags in the MGC
Flag setting is required to ensure the correct MCU behavior for cascading conferences. It is performed once
per MCU.
To modify the flags:
1 In the MGC Manager, right-click the MCU icon and then select MCU Utils > Edit “system.cfg”.
2 In the H264 Section, ensure that the following flags are set to:
 ENABLE_HD_SD_IN_FIXED_MODE=YES
Setting this flag to YES enables H.264 Standard Definition (SD), High Definition (HD) and VSX
8000 (Version 8.0) support in Video Switching conferences.
 H264_VSW_AUTO=NO
Setting this flag to NO disables the highest common mechanism in H.264 and enables the
selection of H.264 Video Protocol in fixed mode in Dual Stream Video Switching cascading
conferences.
 ENABLE_H239_ANNEX_T=YES
This flag should be set to the same value (YES/NO) as the settings of the RealPresence
Collaboration Server flag H263_ANNEX_T.
Note: MIH Cascading with MGC Requirement
To use MIH Cascade in the MGC, the Conference Numeric ID routing mode must be used. It is
determined when the system.cfg flag in the GREET AND GUIDE/IVR section is set to
QUICK_LOGIN_VIA_ENTRY_QUEUE=NO.
3 Click OK.
4 If you changed the flags, reset the MCU.
Method II - Defining the Cascading Entry Queue in the MGC
The Entry Queue definition on the MGC is required if the dialing is done from the RealPresence
Collaboration Server to the MGC.
To define the Cascading Entry Queue in the MGC:
1 In the MGC Manager, expand the MCU tree.
2 Right-click the Meeting Rooms, Entry Queues and SIP Factories icon and click New Entry
Queue.
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3 In the New Entry Queue dialog box, set the Entry Queue parameters and select the Cascade check
box.
For more details on the definition of new Entry Queues refer to the MGC Manager User’s Guide,
Volume II, Chapter 1, Ad Hoc Auto Cascading and Cascading Links.
4 Click OK.
Creating the Dial-out Link between the Conference Running on the MGC and the
Conference Running on the RealPresence Collaboration Server
If the dialing is done from the MGC to the RealPresence Collaboration Server, you need to define the
cascaded link (dial-out participant) in the conference running on the MGC.
The dial-out string defined for the participant is the dialing string required to connect to the destination
conference via the Cascade-enabled Entry Queue defined on the RealPresence Collaboration Server
hosting the destination cascaded conference. The dial-out participant can be defined on the MGC as
template or assigned to the Meeting Room.
In the MGC Manager application, define a new participant as follows:
1 In the Participant Properties - Identification dialog box, enter a Participant Name
2 In the Connection Type field, select Dial-out.
3 In the Interface Type list field, select H.323.
4 There are two methods to define the dialing string to the other conference:
a Using the MCU’s IP Address and the Alias string - Method A
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b Using only the Alias string (requires a gatekeeper) - Method B
Method A
This method does not use any gatekeepers.
In the IP Address field, enter the IP address of the Signaling Host of the RealPresence
Collaboration Server hosting the destination conference.
In the Alias Name/Type field, enter the ID of the cascade-enabled Entry Queue (EQ), the Conference
ID and Password of the destination conference as follows:
<EQ ID>##<Destination Conference ID>##<Password> (Password is optional).
For Example:
Cascade-enabled
EQ ID
1002##12001##1234
Destination
Confer
Password (optional)
Method B
This method uses a gatekeeper.
In the Alias Name field, enter the Prefix of MCU B, EQ ID, Destination Conference ID, and Password,
as follows:
<MCU Prefix EQ ID>##<Conference ID>##<Password> (Password is optional)
For Example:
MCU Prefix as
registered in
the
gatekeeper
9251002##12001##1234
Password (optional)
Cascade-enabled
EQ ID
Conference ID
5 Click the Advanced tab and in the Node Type field, select Terminal.
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6 Click OK.
Defining the Cascade Enabled Entry Queue on the RealPresence Collaboration Server
If the dialing is done from the conference running on the MGC that is the Master MCU, a Cascade-enabled
Entry Queue must be defined on the RealPresence Collaboration Server setting it as Slave.
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For more details, see MGC to RealPresence Collaboration Server Cascading.
Defining the Cascading Conferences
The table below lists the line rates and the video settings that should be used when defining the conferences
on the MGC. The same line rates should be selected when defining the Conference Profiles on the
RealPresence Collaboration Server, as well as whether the conference is HD Video Switching. However,
the video settings will be automatically selected by the system.
Recommended Conference Line Rates for Cascaded Conferences
Topology
Video Session Mode
Conference Line Rate
MGC
MGC - CIF 263
RealPresence Collaboration Server CIF 264 CP
768Kb/s, 2Mb/s
MGC - HD VSW
RealPresence Collaboration Server HD VSW
1.5Mb/s

RealPresence
Collaboration
Server
In addition, the conference running on the MGC should be set as Meet Me Per Conference and select the
H.239 option in the Dual Stream Mode field. For more details on conference definition on the MGC, refer to
the MGC Manager User’s Guide, Volume I, Chapter 5.
Defining the Dial-out Participant on the RealPresence Collaboration Server
If the dialing is done from a conference running on the RealPresence Collaboration Server to the conference
running on the MGC, the dial-out participant is defined in the conference running on the RealPresence
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Collaboration Server, setting the Cascade field to Slave. This participant dials the Cascade-enabled Entry
Queue defined on the MGC.
To define the Dial-out Participant:
1 Display the list of participants in the linked conference (Slave conference).
2 In the Participant List pane, click the New Participant (
) button.
The New Participant - General dialog box is displayed.
3 In the Name field, enter a participant name.
4 In the Dialing Direction field, select Dial-out.
5 In the Type list field, verify that H.323 is selected.
6 There are two methods to define the dialing string:
a Using the MCU’s IP Address and the Alias string - Method A
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b Using only the Alias string (requires a gatekeeper) - Method B
Method A
This method does not used a gatekeeper.
In the IP Address field, enter the IP address of the MGC hosting the destination conference (Master
conference).
In the Alias Name/Type field, enter the ID of the cascade-enabled Entry Queue (EQ), the Conference
ID and Password of the destination conference (Master Conference) as follows:
<EQ ID>##<Destination Conference ID>##<Password> (Password is optional).
Method B
This method uses a gatekeeper.
In the Alias Name field, enter the MGC Prefix as registered in the gatekeeper, EQ ID, Destination
Conference ID, and Password, as follows:
<MGC Prefix EQ ID>##<Conference ID>##<Password> (Password is optional)
7 Click the Advanced tab and in the Cascade field, select the Slave option.
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8 Click OK.
The cascade-enabled dial-out link is created and the system automatically dials out to connect the
participant to the local conference, as well as the destination conference on the MGC.
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Meeting Rooms
A Meeting Room is a conference saved on the MCU in passive mode, without using any of the system
resources. A Meeting Room is automatically activated when the first participant dials into it. Meeting Rooms
can be activated as many times as required. Once activated, a Meeting Room functions as any ongoing
conference.
The conferencing Mode of the Meeting Room is determined by the Profile assigned to it.
In SVC Conferencing Mode, dial-in is available as follows:
● AVC-capable endpoints (participants) can only connect to an AVC CP Meeting Room. When dialing
into SVC Only Meeting Room the calls fail.
● SVC-capable endpoints support both AVC and SVC video protocols. When dialing into SVC Only
conferences, they connect as SVC endpoints. When dialing into AVC CP Only conferences, they
connect as AVC endpoints.
● Both AVC and SVC endpoints can connect to a mixed CP and SVC conference.
In AVC Conferencing Mode, ISDN/PSTN participants can dial-in directly to a Meeting Room without
connection through an Entry Queue. Up to two numbers can be defined per conference provided that they
are from the same ISDN/PSTN Network Service. When a dial-in number is allocated to a Meeting Room,
the number cannot be deleted nor can the ISDN/PSTN Network Service be removed. The dial-in number
must be communicated to the ISDN or PSTN dial-in participants.
In AVC CP Conferences, dial-out participants can be connected to the conference automatically, or
manually. In the automatic mode the system calls all the participants one after the other. In the manual
mode, the Collaboration Server user or meeting organizer instructs the conferencing system to call the
participant. Dial-out participants must be defined (mainly their name and telephone number) and added to
the conference. This mode can only be selected at the conference/Meeting Room definition stage and
cannot be changed once the conference is ongoing.
A Meeting Room can be designated as a Permanent Conference.
The system is shipped with four default Meeting Rooms:
Default Meeting Rooms List
Meeting Room Name
ID
Default Line Rate
Maple_Room
1001
384 Kbps
Oak_Room
1002
384 Kbps
Juniper_Room
1003
384 Kbps
Fig_Room
1004
384 Kbps
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
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Use Time Out as DTMF Delimiter
Users are able to change the behavior of the policy of number collection for VMR Entry Queues,
Conference-IDs and Passwords, allowing a time-out to be used as a stop indicator for the input string.
In previous versions, a # input at the end of the input string indicated completion of the input.
The administrator can configure the system, using the ENABLE_DTMF_NUMBER_WO_DELIMITER
system flag to change the previous system behavior, allowing a time-out to be used as a stop indicator for
the string input for the local IVR, when the MCU collects the Conference-ID in the local Entry Queue or the
Password (chairperson or participant) while routed to the conference.
The flag must be manually added to the System Configuration and its value modified as follows:
Value / Description
Flag Name
YES
ENABLE_DTMF_NUM
BER_WO_DELIMITER
If the timer expires, the received digits
validated even if there is no delimiter.
If the received number is not valid, the
system will prompt again for the number
according to number of retries. that are
configured.
NO
This is the default setting for backward
compatibility.
If the timer expires because no delimiter
is received, the number input is not valid.
The system will prompt again for the
number according to number of retries.
that are configured.
A System Reset in not required for the flag setting to take effect.
For more information see, Modifying System Flags.
Meeting Rooms List
Meeting Rooms are listed in the Meeting Room list pane.
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Meeting Rooms
To list Meeting Rooms:
» In the RMX Management pane, in the Frequently Used list, click the Meeting Rooms button
.
The Meeting Rooms list is displayed.
An active Meeting Room becomes an ongoing conference and is monitored in the same way as any
other conference.
The Meeting Room List columns include:
Meeting Rooms List Columns
Field
Description
Display Name
Displays the name and the icon of the Meeting Room in the Collaboration Server Web
Client.
(green)
(gray)
An active video Meeting Room that was
activated when the first participant connected to
it.
A passive video Meeting Room that is waiting to
be activated.
Routing Name
The ASCII name that registers conferences, Meeting Rooms, Entry Queues and SIP
Factories in the various gatekeepers and SIP Servers. In addition, the Routing Name is
also:
• The name that endpoints use to connect to conferences.
• The name used by all conferencing devices to connect to conferences that must be
registered with the gatekeeper and SIP Servers.
ID
Displays the Meeting Room ID. This number must be communicated to H.323
conference participants to enable them to dial in.
Duration
Displays the duration of the Meeting Room in hours using the format HH:MM
(default 01:00).
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Meeting Rooms List Columns
Field
Description
Conference Password
The password to be used by
participants to access the Meeting
Room. If blank, no password is
assigned to the conference. This
password is valid only in
conferences that are configured to
prompt for a conference password
in the IVR Service.
The Collaboration Server can be configured to
automatically generate conference and
chairperson passwords when these fields are
left blank. For more information, see the
Automatic Password Generation Flags.
Chairperson Password
Displays the password to be used
by the users to identify themselves
as Chairpersons. They are granted
additional privileges. If left blank,
no chairperson password is
assigned to the conference. This
password is valid only in
conferences that are configured to
prompt for a chairperson
password.
Profile
Displays the name of the Profile assigned to the Meeting Room. For more information,
see Defining New Profiles.
SIP Registration
The status of registration with the SIP server:
• Not configured - Registration with the SIP Server was not enabled in the
Conference Profile assigned to this conferencing Entity. In Multiple Networks
configuration, If one service is not configured while others are configured and
registered, the status reflects the registration with the configured Network Services.
The registration status with each SIP Server can be viewed in the Properties Network Services dialog box of each conferencing entity.
When SIP registration is not enabled in the conference profile, the Collaboration
Server's registering to SIP Servers will each register with a URL derived from its
own signaling address. In Collaboration Server 2000/4000, this unique URL
replaces the non-unique URL, dummy_tester, used in previous versions.
• Failed - Registration with the SIP Server failed. This may be due to incorrect
definition of the SIP server in the IP Network Service, or the SIP server may be
down, or any other reason the affects the connection between the Collaboration
Server or the SIP Server to the network.
• Registered - the conferencing entity is registered with the SIP Server.
• Partially Registered - This status is available only in Multiple Networks
configuration, when the conferencing entity failed to register to all the required
Network Services if more than one Network Service was selected.
Use Time Out as DTMF Delimiter
Users are able to change the behavior of the policy of number collection for VMR Entry Queues,
Conference-IDs and Passwords, allowing a time-out to be used as a stop indicator for the input string.
In previous versions, a # input at the end of the input string indicated completion of the input.
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The administrator can configure the system, using the ENABLE_DTMF_NUMBER_WO_DELIMITER
system flag to change the previous system behavior, allowing a time-out to be used as a stop indicator for
the string input for the local IVR, when the MCU collects the Conference-ID in the local Entry Queue or the
Password (chairperson or participant) while routed to the conference.
The flag must be manually added to the System Configuration and its value modified as follows:
Value / Description
Flag Name
YES
ENABLE_DTMF_NUM
BER_WO_DELIMITER
If the timer expires, the received digits
validated even if there is no delimiter.
If the received number is not valid, the
system will prompt again for the number
according to number of retries. that are
configured.
NO
This is the default setting for backward
compatibility.
If the timer expires because no delimiter
is received, the number input is not valid.
The system will prompt again for the
number according to number of retries.
that are configured.
A System Reset in not required for the flag setting to take effect.
For more information see Modifying System Flags.
Meeting Room Toolbar & Right-click Menu
The Meeting Room toolbar and right-click menus provide the following functionality:
Meeting Room Toolbar and Right-click Menus
Toolbar
button
Right-click menu
Description
New Meeting Room
Select this button to create a new Meeting Room.
Delete Meeting Room
Select any Meeting Room and then click this button to delete the
Meeting Room.
Dial out to AVC participants assigned to a Meeting Room will only start when the dial in participant
who has activated it has completed the connection process and the Meeting Room has become an
ongoing conference.
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Creating a New Meeting Room
To create a new meeting room:
● In the Meeting Rooms pane, click the New Meeting Room
in the pane, and select New Meeting Room.
button or right-click an empty area
The New Meeting Room dialog box is displayed.
The definition procedure is the same as for the new conference (with the exception of Reserved
Resources for Audio and Video participants in Collaboration Server 2000/4000 only).
If SIP Factories are being used do not assign a Meeting Room the ID 7001. This ID is reserved for
the default SIP Factory.
Microsoft Lync users can connect a Collaboration Server Meeting Room to a conference running on
the Microsoft A/V MCU. This allows Collaboration Server Lync users to connect with a conference in
progress on the A/V MCU and be an active participant in the conference.
Microsoft Lync is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
For more information, see Connecting a Collaboration Server Meeting Room to a Microsoft AV-MCU
Conference.
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Entry Queues, Ad Hoc Conferences and
SIP Factories
Entry Queues
An Entry Queue (EQ) is a special routing lobby to access conferences. Participants connect to a single-dial
lobby and are routed to their destination conference according to the Conference ID they enter. The Entry
Queue remains in a passive state when there are no callers in the queue (in between connections) and is
automatically activated once a caller dials its dial-in number.
Participants can be moved from the Entry Queue and the destination conference if both conferencing
entities are set to the same conferencing parameters: Conferencing Mode, Line rate and video parameters.
For example, participants can be moved from SVC Only Entry Queue to SVC Only conference, or from a
mixed CP and SVC Entry Queue to a mix CP and SVC conference, from CP only Entry Queue to CP only
conference.
The parameters (bit rate and video properties) with which the participants connect to the Entry Queue and
later to their destination conference are defined in the Conference Profile that is assigned to the Entry
Queue. For example, if the Profile Bit Rate is set to 384kbps, all endpoints connect to the Entry Queue and
later to their destination conference using this bit rate even if they are capable of connecting at higher bit
rates.
An Entry Queue IVR Service must be assigned to the Entry Queue to enable the voice prompts guiding the
participants through the connection process. The Entry Queue IVR Service also includes a video slide that
is displayed to the participants while staying in the Entry Queue (during their connection process).
Different Entry Queues can be created to accommodate different conferencing modes, conferencing
parameters (by assigning different Profiles) and prompts in different languages (by assigning different Entry
Queue IVR Services).
For more information, see IVR Services List.
The Entry Queue can also be used for Ad Hoc conferencing. If the Ad Hoc option is enabled for the Entry
Queue, when the participant enters the target conference ID the system checks whether a conference with
that ID is already running on the MCU. If not, the system automatically creates a new ongoing conference
with that ID. For more information about Ad Hoc conferencing, see Ad Hoc Conferencing.
An Entry Queue can be designated as Transit Entry Queue to which calls with dial strings containing
incomplete or incorrect conference routing information are transferred. For more information, see Transit
Entry Queue.
To enable ISDN/PSTN participants to dial in to the Entry Queue, an ISDN/PSTN dial-in number must be
assigned to the Entry Queue. Up to two dial-in numbers can be assigned to each Entry Queue. The dial-in
numbers must be allocated from the dial-in number range defined in the ISDN/PSTN Network Service. You
can allocate the two dial-in numbers from the same ISDN/PSTN Network Service or from two different
ISDN/PSTN Network Services. The dial-in number must be communicated to the ISDN or PSTN dial-in
participants.
The Entry Queue can also be used as part of the Gateway to Polycom® RealPresence DMA solution for
connecting Audio only PSTN, ISDN, SIP and H.323 endpoints to RealPresence DMA™ system.
Gateway functionality is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
For more information, see Dialing to Polycom® RealPresence DMA System.
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Default Entry Queue properties
The system is shipped with a default Entry Queue whose properties are shown in the following table.
Default Entry Queue Properties
Parameter
Value
Display Name
DefaultEQ
The user can change the name if required.
Routing Name
DefaultEQ
The default Routing Name cannot be changed.
ID
1000
Profile name
Factory_Video_Profile. Profile Bit Rate is set to 384 Kbps.
Entry Queue Service
Entry Queue IVR Service. This is default Entry Queue IVR Service shipped with the
system and includes default voice messages and prompts in English.
Ad Hoc
Enabled
Cascade
None (Disabled)
Enable ISDN/PSTN
Access
Disabled. You can modify the properties of this Entry Queue to enable ISDN/PSTN
participants to dial-in to a conference. Up to two dial-in numbers can be assigned.
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
Defining a New Entry Queue
You can modify the properties of the default Entry Queue and define additional Entry Queues to suit different
conferencing requirements.
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Entry Queues, Ad Hoc Conferences and SIP Factories
To define a new Entry Queue:
1 In the RMX Management pane, In the Rarely Used menu, click Entry Queues.
2 In the Entry Queues list pane, click the New Entry Queue button.
The New Entry Queue dialog box opens.
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3 Define the following parameters:
Entry Queue Definitions Parameters
Option
Description
Display Name
The Display Name is the conferencing entity name in native language character
sets to be displayed in the Collaboration Server Web Client.
In conferences, Meeting Rooms, Entry Queues and SIP factories the system
automatically generates an ASCII name for the Display Name field that can be
modified using Unicode encoding.
• English text uses ASCII encoding and can contain the most characters
(length varies according to the field).
• European and Latin text length is approximately half the length of the
maximum.
• Asian text length is approximately one third of the length of the maximum.
The maximum length of text fields also varies according to the mixture of
character sets (Unicode and ASCII).
Maximum field length in ASCII is 80 characters. If the same name is already
used by another conference, Meeting Room or Entry Queue, the Collaboration
Server displays an error message requesting you to enter a different name.
Routing Name
Enter a name using ASCII text only. If no Routing Name is entered, the system
automatically assigns a new name as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered
in Display Name, the ID (such as Conference ID) is used as the Routing
Name.
Profile
Select the Profile to be used by the Entry Queue.
The default Profile is selected by default. This Profile determines the Bit Rate
and the video properties with which participants connect to the Entry Queue and
destination conference.
To connect to a Video Switching conference via Entry Queue, the Profile
assigned to the Entry Queue must be set to Video Switching. It is recommended
to use the same profile for both the destination conference and Entry Queue.
In Ad Hoc conferencing, it is used to define the new conference properties.
ID
Enter a unique number identifying this conferencing entity for dial in. Default
string length is 4 digits.
If you do not manually assign the ID, the MCU assigns one after the completion
of the definition. The ID String Length is defined by the flag
NUMERIC_CONF_ID_LEN in the System Configuration.
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Entry Queue Definitions Parameters
Option
Description
Entry Queue Mode
Select the mode for the Entry Queue
Standard Lobby (default) - When selected, the Entry Queue is used as a
routing lobby to access conferences. Participants connect to a single-dial lobby
and are routed to their destination conference according to the Conference ID
they enter.
Ad Hoc - Select this option to enable the Ad Hoc option for this Entry Queue. In
this mode, when the participant enters the target conference ID the system
checks whether a conference with that ID is already running on the MCU. If not,
the system automatically creates a new ongoing conference with that ID.
IVR Only Service Provider - When selected, designates this Entry Queue as a
special Entry Queue that provides IVR Services to SIP calls on behalf of the
RealPresence DMA system. The IVR Only Service Provider Entry Queue does
not route the SIP calls to a target conference. Instead the RealPresence DMA
system handles the call. For more details, see IVR Provider Entry Queue
(Shared Number Dialing).
External IVR Control - IVR Services can be controlled externally from an
application server (such as the DMA) supporting the MCCF-IVR (Media Control
Channel Framework-Interactive Voice Response) package.
When selected, the connection process of the participant to the conference via
the Virtual Entry Queue is controlled and managed by an external IVR service of
an application server (for example, DMA).
Entry Queue IVR Service
The default Entry Queue IVR Service is selected. If required, select an alternate
Entry Queue IVR Service, which includes the required voice prompts, to guide
participants during their connection to the Entry Queue.
Cascade
Set this field to None for all Entry Queues other than cascading.
If this Entry Queue is used to connect dial-in cascaded links, select Master or
Slave depending on the Master/Slave relationship in the Cascading topology.
Set this field to Master if:
• The Entry Queue is defined on the MCU on level 1 and the dialing is done
from level 2 to level 1.
• The Entry Queue is defined on the MCU on level 2 and the dialing is done
from level 3 to level 2.
Set this field to Slave if the Entry Queue is defined on the MCU on level 2
(Slave) and the dialing is done from MCU level 1 to level 2.
Enable ISDN/PSTN
Access
Select this check box to allocate dial-in numbers for ISDN/PSTN connections.
To define the first dial-in number using the default ISDN/PSTN Network Service,
leave the default selection. When the Entry Queue is saved on the MCU, the
dial-in number will be automatically assigned to the Entry Queue. This number
is taken from the dial-in numbers range in the default ISDN/PSTN Network
Service.
ISDN/PSTN Network
Service
The default Network Service is automatically selected. To select a different
ISDN/PSTN Network Service in the service list, select the name of the Network
Service.
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Entry Queue Definitions Parameters
Option
Description
Dial-in Number (1)
Leave this field blank to let the system automatically assign a number from the
selected ISDN/PSTN Network Service. To manually define a dial-in number,
enter a required number from the dial-in number range defined for the selected
Network Service.
Dial-in Number (2)
By default, the second dial-in number is not defined. To define a second-dial-in
number, enter a required number from the dial-in number range defined for the
selected Network Service.
4 Click OK.
The new Entry Queue is added to the Entry Queues list.
Listing Entry Queues
To view the list of Entry Queues:
» In the RMX Management pane- Rarely Used menu, click Entry Queues.
The Entry Queues are listed in the Entry Queues pane.
You can double-click an Entry Queue to view its properties.
Modifying the EQ Properties
To modify the EQ:
» In the Entry Queues pane, either double-click or right-click, and select Entry Queue Properties of
the selected Entry Queue in the list.
The Entry Queue Properties dialog box is displayed. All the fields may be modified except Routing
Name.
Transit Entry Queue
A Transit Entry Queue is an Entry Queue to which calls with dial strings containing incomplete or incorrect
conference routing information are transferred.
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IP Calls are routed to the Transit Entry Queue when:
● A gatekeeper is not used, or where calls are made directly to the Collaboration Server’s Signaling IP
Address, with incorrect or without a Conference ID.
● When a gatekeeper is used and only the prefix of the Collaboration Server is dialed, with incorrect or
without a Conference ID.
● When the dialed prefix is followed by an incorrect conference ID.
When no Transit Entry Queue is defined, all calls containing incomplete or incorrect conference routing
information are rejected by the Collaboration Server.
In the Transit Entry Queue, the Entry Queue IVR Service prompts the participant for a destination
conference ID. Once the correct information is entered, the participant is transferred to the destination
conference.
Setting a Transit Entry Queue
The Collaboration Server factory default settings define the Default Entry Queue also as the Transit Entry
Queue. You can designate another Entry Queue as the Transit Entry Queue.
Only one Transit Entry Queue may be defined per Collaboration Server and selecting another Entry Queue
as the Transit Entry Queue automatically cancels the previous selection.
To designate an Entry Queue as Transit Entry Queue:
1 In the RMX Management pane, Rarely Used list, click Entry Queues.
2 In the Entry Queues list, right-click the Entry Queue entry, and then click Set Transit Entry Queue.
The Entry Queue selected as Transit Entry Queue is displayed in bold.
To cancel the Transit Entry Queue setting:
1 In the RMX Management pane, Rarely Used list, click Entry Queues.
2 In the Entry Queues list, right-click the Transit Entry Queue entry, and then click Cancel Transit
Entry Queue.
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IVR Provider Entry Queue (Shared Number Dialing)
In an environment that includes a RealPresence DMA system, the Collaboration Server Entry Queue can
be configured to provide the IVR Services on behalf of the RealPresence DMA system to SIP endpoints. It
displays the Welcome Slide, plays the welcome message and retrieves the destination conference ID that
is entered by the participant using DTMF codes.
To enable this feature, a special Entry Queue that is defined as IVR Only Service Provider is created. This
Entry Queue does not forward calls to conferences running on the Collaboration Server and its main
functionality is to provide IVR services.
Call Flow
The SIP participant dials the DMA Virtual Entry Queue number, for example 1000@dma.polycom.com.
The DMA forwards the SIP call to the Collaboration Server, to a special Entry Queue that is configured as
IVR Only Service Provider. The participant is prompted to enter the conference ID using DTMF codes.
Once the participant enters the conference ID, the conference ID is forwarded to the DMA, enabling the
DMA to connect the SIP endpoint to the destination conference or create a new conference and connect
the participant to that conference.
Guidelines for setting the Entry Queue as IVR Provider
● An Entry Queue defined as IVR Only Service Provider does not route the SIP call to a target
conference and it cannot be used to route calls on the Collaboration Server. In such a configuration,
the DMA handles the calls. Therefore, normal Entry Queues must be defined separately.
● Operator Assistance must be disabled in the IVR Service assigned to this Entry Queue.
● Only the conference ID prompts should be configured. Other prompts are not supported in IVR Only
Service Provider configuration.
● PSTN, ISDN, and H.323 calls to this Entry Queue are rejected.
● The DMA must be configured to locate the IVR Only Service Provider Entry Queue on the
Collaboration Server. To locate the Entry Queue the DMA requires the Entry Queue’s ID number and
the Collaboration Server Signaling IP address (xxx.xx.xxx.xx).
Configuring the Collaboration Server as IVR Provider
Entry Queue IVR Service
If required, create a special Entry Queue IVR Service in which the Operator Assistance option is disabled,
and only the Conference ID prompts are enabled.
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Entry Queue
» In the New Entry Queue dialog box, Entry Queue Mode list, select IVR Only Service Provider.
 Enter the Entry Queue ID to be used by the DMA for forwarding the SIP calls to this Entry Queue.
 Select the special Entry Queue IVR Service if one was created.
 The Cascade and Enable ISDN/PSTN Dial-in options should not be enabled with this type of
Entry Queue.
Using External IVR Services via the MCCF-IVR Package
IVR Services can be controlled externally from an application server supporting the MCCF-IVR (Media
Control Channel Framework-Interactive Voice Response) package. The external IVR service is currently
being implemented with the integration of the Polycom RealPresence Virtualization Manager (DMA) as the
application server. When the application server is deployed in the enterprise environment and the Polycom
RealPresence Collaboration Server (MCU) is deployed as a media server, the external IVR service can be
used to play audio messages, display slides, and collect DTMF input from the participant. The external IVR
service is managed by the application server at the pre-conference phase when the participant is placed
into a special external IVR-controlled Entry Queue in the Collaboration Server (MCU), collecting information
before connecting to the conference.
The external IVR-controlled Entry Queue plays recorded voice messages or sends video slides such as
splash screens to the participant and collects DTMF input from the participant such as conference ID and
conference password for various functions.
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IVR media files, WAV for voice messages and JPG for video slides, are stored on the application server. In
order to provide external IVR control, a TCP-based MCCF channel is created between the application
server and the media server. Because of real-time considerations, when the MCCF channel is established,
the application server notifies the media server about the media files. The media server downloads the
media files. The media server is notified by the application server when to download new or updated media
files.
When the call has completed the pre-conference phase in the external IVR-controlled Entry Queue, the
application server disconnects the call from the Entry Queue and routes the call to an ongoing conference
or creates a new VMR.
Call Flows
The external IVR-controlled Entry Queue can be initiated for various types of calls from SIP endpoints such
as standalone endpoints and Cisco TIP endpoints. Standalone endpoints are SIP or H.264 TIP endpoints.
These endpoints can include HDX systems, multiple Telepresence (ITP) screens, and RealPresence
Desktop client applications.
Call Flow for Standalone SIP Endpoints
The following describes how a standalone SIP endpoint call is placed into the IVR-controlled Entry Queue
and is then connected to a conference:
1 A SIP call is routed through the application server to the IVR-controlled Entry Queue.
2 The MCU answers the call and waits for the IVR media file requests from the application server. The
MCU does not control the call while the call is in the Entry Queue.
3 The application server may request, through the MCCF channel - IVR package, to play an audio file
and display a slide. When the audio file has finished playing, the MCU notifies the application server
that the audio file has been played for the call.
4 The application server may request, through the MCCF channel - IVR package, to collect DTMF
input such as a conference ID or password, from the caller. The DTMF input is transferred from the
MCU to the application server. When the application server receives the DTMF input, it validates the
input for the required conference ID or password. If the input is incorrect, the application server will
request the MCU to replay the audio file and collect the DTMF input again. The MCU transfers the
DTMF input to the application server for revalidation.
5 When the application server has completed the pre-conference IVR, the application server routes
the call to a VMR with the collected password appended to the following dial string:
<conf-id>**<password>@mcu-sig-ip.
The call is disconnected from the application server. The MCU now has control of the call.
6 The call is transferred to a conference, which can reside on another MCU.
Call Flow for Standalone TIP Endpoints
The following describes how a standalone TIP endpoint call is placed into the IVR-controlled Entry Queue
and is then connected to a conference:
1 A TIP call is routed through the application server to the IVR-controlled Entry Queue. TIP endpoints
can either have a single screen or multiple screens.
2 The MCU answers the call and waits for the IVR media file requests from the application server. The
MCU does not control the call while the call is in the Entry Queue.
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3 The application server may request, through the MCCF channel - IVR package, to play an audio file
and display a video slide. When the TIP endpoint uses multiple screens, the video slide is displayed
on the main screen only. When the audio file has finished playing, the MCU notifies the application
server that the audio file has been played for the call.
4 The application server may request, through the MCCF channel - IVR package, to collect DTMF
input such as a conference ID or password, from the caller. When the TIP endpoint uses multiple
screens, the DTMF input is collected only once from the main screen. The DTMF input is transferred
from the MCU to the application server. When the application server receives the DTMF input, it
validates the input for the required conference ID or password. Because TIP uses DTLS, it can
optionally enable re-keying of DTMF input and the calls to the Entry Queue and the conference can
be encrypted.
5 When the application server has completed the pre-conference IVR, the application server routes
the call to a VMR with the collected password appended to the following dial string:
<conf-id>**<password>@mcu-sig-ip.
The call is disconnected from the application server. The MCU now has control of the call.
6 The call is transferred to a conference, which can reside on another MCU.
Call Flow for TIP Endpoints from a Polycom ITP System
The following describes how a TIP call from Cisco TPS endpoints or TIP calls from a Polycom ITP system
working as a TIP call is placed into the IVR-controlled Entry Queue and is then connected to a conference:
1 A TIP call is routed through the application server to the IVR-controlled Entry Queue.
2 The MCU answers the call and waits for the IVR media file requests from the application server. The
MCU does not control the call while the call is in the Entry Queue.
3 While the call is in the Entry Queue, video is only displayed on the main screen.
4 DTMF input is collected only once from the main screen. Because TIP uses DTLS, it can optionally
enable re-keying of DTMF input and the calls to the Entry Queue and the conference can be
encrypted.
5 When the application server has completed the pre-conference IVR, the application server routes
the call to a VMR with the collected password appended to the following dial string:
<conf-id>**<password>@mcu-sig-ip. The MCU now has control of the call.
The call is transferred to a conference, which can reside on another MCU.
Guidelines for Using External IVR Services via the MCCF-IVR Package
● Only AVC SIP and TIP protocols are supported.
● MCCF channels support both IPV4 and IPV6.
● When the MCCF channel is disconnected, an alarm is displayed and all external IVR files are deleted.
When the MCCF channel is reconnected, the external IVR files are sent to the MCU.
● When the Collaboration Server (MCU) is restarted, all existing external IVR files are deleted. When
the MCCF channel connects to the Collaboration Server, the external IVR file are sent to the
Collaboration Server.
● H.323 and ISDN protocols are not supported.
● Video Switching conferences do not support the TIP protocol
● TIP-based conferencing does not support the following features during conferences:
 Gathering phase
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 Skin display
 Text messaging using Message Overlay
 Site Name display
 PCM
 Click&View
● To play audio messages and display the welcome slide during the participant connection to the
conference via the Virtual Entry Queue, the Media files have to meet the following requirements (as
defined in the Entry Queue IVR Service):
 Audio messages: WAV files - PCM, 16 KHz, 16 bit, Mono
 Video slides: JPG files - 1920 x 1088 resolution
Configuring the MCU to Support External IVR Services via the MCCF-IVR
The support of External IVR Services via the MCCF-IVR package is enabled by default in the Collaboration
Server (RMX) systems, by the flag ENABLE_MCCF which is set to YES.
However, in Ultra Secure Mode and in secured environments where the External IVR Services via the
MCCF-IVR package is not required and unused ports should be closed, this flag should be set to NO.
To change this flag value from YES to No, you must first add it to the System Configuration. For more details,
see .
Configuring the Entry Queue to Use External IVR Services
The Entry Queue can be configured to place a call in the external IVR-controlled Entry Queue.
To configure the Entry Queue for External IVR Services:
1 In the RMX Management pane, click Entry Queues.
2 In the Entry Queues pane, click the New Entry Queue icon.
The New Entry Queue dialog box is displayed.
3 In the Display Name field, type an appropriate display name for the Entry Queue.
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4 In the Entry Queue Mode field, select External IVR Control from the available options.
When External IVR Control is selected, the connection process of the participant to the conference
via the Virtual Entry Queue is controlled and managed by an external IVR service of an application
server (for example, DMA).
5 Click OK.
SIP Factories
A SIP Factory is a conferencing entity that enables SIP endpoints to create Ad Hoc conferences. The
system is shipped with a default SIP Factory, named DefaultFactory.
.
The default SIP Factory uses the conferencing ID 7001. If a SIP Factory is being used do not assign
this ID to any conferencing entity, including conferences, reservations, and meeting rooms.
When a SIP endpoint calls the SIP Factory URI, a new conference is automatically created based on the
Profile parameters, and the endpoint joins the conference.
The SIP Factory URI must be registered with the SIP server to enable routing of calls to the SIP Factory. To
ensure that the SIP factory is registered, the option to register Factories must be selected in the Default IP
Network Service.
Creating SIP Factories
To create a new SIP Factory:
1 In the RMX Management pane, Rarely Used list, click SIP Factories.
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2 In the SIP Factories list pane, click the New SIP Factory button.
The New Factory dialog box opens.
3 Define the following parameters:
:
New Factory Properties
Option
Description
Display Name
Enter the SIP Factory name that will be displayed.
The Display Name is the conferencing entity name in native language character
sets to be displayed in the Collaboration Server Web Client.
In conferences, Meeting Rooms, Entry Queues and SIP factories the system
automatically generates an ASCII name for the Display Name field that can be
modified using Unicode encoding.
• English text uses ASCII encoding and can contain the most characters
(length varies according to the field).
• European and Latin text length is approximately half the length of the
maximum.
• Asian text length is approximately one third of the length of the maximum.
The maximum length of text fields also varies according to the mixture of
character sets (Unicode and ASCII).
Maximum field length in ASCII is 80 characters. If the same name is already
used by another conference, Meeting Room or Entry Queue, the Collaboration
Server displays an error message requesting you to enter a different name.
Routing Name
The Routing Name is defined by the user, however if no Routing Name is
entered, the system will automatically assign a new name when the Profile is
saved as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered
in Display Name, the ID (such as Conference ID) is used as the Routing
Name.
Profile
The default Profile is selected by default. If required, select the conference
Profile from the list of Profiles defined in the MCU.
A new conference is created using the parameters defined in the Profile.
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New Factory Properties
Option
Description
Automatic Connection
Select this check box to immediately accept the conference creator endpoint to
the conference. If the check box is cleared, the endpoint is redirected to the
conference and then connected.
4 Click OK.
The new SIP Factory is added to the list.
SIP Registration & Presence for Entry Queues and SIP
Factories with SIP Servers
Entry Queues and SIP Factories can be registered with SIP servers. This enables Office Communication
Server or Lync server client users to see the availability status (Available, Offline, or Busy) of these
conferencing entities, and to connect to them directly from the Buddy List.
Guidelines for registering Entry Queues and SIP Factories with SIP Servers
● The Entry Queue or SIP Factory must be added to the Active Directory as a User.
● SIP Registration must be enabled in the Profile assigned to the Entry Queue or SIP Factory. For more
information see Defining New Profiles.
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Monitoring Registration Status
The SIP registration status can be viewed in the Entry Queue or SIP Factory list panes.
The following statuses are displayed:
● Not configured - Registration with the SIP Server was not enabled in the Conference Profile
assigned to the Entry Queue or SIP Factory.
When SIP registration is not enabled in the conference profile, the Collaboration Server's registering
to SIP Servers will each register with an URL derived from its own signaling address. In Collaboration
Server 2000/4000, this unique URL replaces the non-unique URL, dummy_tester, used in previous
versions.
● Failed - Registration with the SIP Server failed.
This may be due to incorrect definition of the SIP server in the IP Network Service, or the SIP Server
may be down, or any other reason the affects the connection between the Collaboration Server or the
SIP Server to the network.
● Registered - The conferencing entity is registered with the SIP Server.
● Partially Registered - This status is available only in Multiple Networks configuration, when the
conferencing entity failed to register to all the required Network Services, if more than one Network
Service was selected for Registration.
Ad Hoc Conferencing
The Entry Queue can also be used for Ad Hoc conferencing. If the Ad Hoc option is enabled for the Entry
Queue, when the participant enters the target conference ID the system checks whether a conference with
that ID is already running on the MCU. If not, the system automatically creates a new ongoing conference
with that ID. The conference parameters are based on the Profile linked to the Entry Queue. As opposed to
Meeting Rooms, that are predefined conferences saved on the MCU, Ad Hoc conferences are not stored
on the MCU. Once an Ad Hoc conference is started, it becomes an ongoing conference, and is monitored
and controlled as any standard ongoing conference.
An external database application can be used for authentication with Ad Hoc conferences. The
authentication can be done at the Entry Queue level and at the conference level. At the Entry Queue level,
the MCU queries the external database server whether the participant has the right to create a new
conference. At the conference level the MCU verifies whether the participant can join the conference and if
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the participant is the conference chairperson. The external database can populate certain conference
parameters.
For more information about Ad Hoc conferencing, see Appendix D - Ad Hoc Conferencing and External
Database Authentication.
Gateway to Polycom® Distributed Media Application™ (DMA™) 7000
Gateway to Polycom® Distributed Media Application™ (DMA™) 7000 enables audio only PSTN, ISDN
(video endpoints using only their audio channels), SIP and H.323 calls to connect to the Polycom DMA 7000
via gateway sessions running on the Collaboration Server. Each Collaboration Server conference acting as
a gateway session includes one connection to the endpoint and another connection to the DMA. The DMA
7000 enables load balancing and the distribution of multipoint calls on up to 10 Polycom Collaboration
Server media servers.
As part of this solution, the Collaboration Server acts as a gateway for the DMA that supports H.323 calls.
The PSTN, ISDN or SIP endpoint dials the virtual Meeting Room on the DMA via a special Entry Queue on
the Collaboration Server.
Gateway functionality is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
For more information, see Dialing to Polycom® RealPresence DMA System.
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Address Book
The Address Book stores information about the people and businesses you communicate with. The
Address Book stores, among many other fields, IP addresses, phone numbers and network communication
protocols used by the participant’s endpoint. By utilizing the Address Book you can quickly and efficiently
assign or designate participants to conferences. Groups defined in the Address Book help facilitate the
creation of conferences. Participants can be added to the Address Book individually or in Groups.
When using the Polycom® RealPresence® Resource Manager® Global Address Book, all entries are listed.
The Address Book can be organized into a multi-level hierarchical structure. It can be used to mirror the
organizational layout of the enterprises and it is especially suitable for large-scale enterprises with a
considerable number of conference participants and organizational departments and divisions. Groups in
the Address Book can contain sub-groups or sub-trees, and individual address book participant entities.
The Address Book provides flexibility in arranging conference participants into groups in multiple levels and
the capabilities to add groups or participants, move or copy participants to multiple groups within the address
book, and use the address book to add groups and participants to a conference or Conference Template.
Importing and exporting of Address Books enables organizations to seamlessly distribute up-to-date
Address Books to multiple Collaboration Server units. It is not possible to distribute Address Books to
external databases running on applications, such as the RealPresence Resource Manager. External
databases can run in conjunction with Collaboration Server units, but must be managed from the external
application. For example, new participants cannot be added to the external database from the Collaboration
Server Web Client. To enable the Collaboration Server to run with an external database such as
RealPresence Resource Manager, the appropriate system configuration flags must be set.
For more information, see Modifying System Flags.
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
Note: Defining Conference Profiles in the Polycom® RealPresence® Web Suite
In the Polycom® RealPresence® Web Suite solution, the Conference Profiles are defined in the
RealPresence DMA system component and should not be defined directly in the RealPresence
Collaboration Server (RMX) 1800/2000/4000 component.
Note: Global Address Book
• Integration with the Global Address Book of the RealPresence Resource Manager is supported.
For more information, see Integrating the Global Address Book (GAB) with the Collaboration
Server. Integration with the SE200 GAB (Global Address Book) is not supported.
• Integration with the Global Address Book is not supported by Collaboration Server (RMX) 1800
with no DSP cards.
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Viewing the Address Book
You can view the participants currently defined in the Address Book. The first time the Collaboration Server
Web Client is accessed, the Address Book pane is displayed.
The Address Book contains two panes:
● Navigation pane - Contains the hierarchical tree and All Participants list.
● List pane - Displays the list of all the members of the selected group and sub-groups.
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The Navigation pane of the Address Book contains the following types of lists:
● Hierarchical — Displays a multi-level hierarchical tree of groups and participants. Double-clicking a
group on the navigation pane displays the group participants and sub-groups in the List pane.
● All Participants — Double-clicking this selection displays the single unique entity of all the
participants in a single level. When adding a participant to a group, the system adds a link to the
participant’s unique entity that is stored in the All Participants list. The same participant may be added
to many groups at different levels, and all these participant links are associated with the same
definition of the participant in the All Participants list. If the participant properties are changed in one
group, they will be changed in all the groups accordingly.
Displaying and Hiding the Group Members in the Navigation Pane
The currently selected group, whose group members are displayed in the Address Book List pane is
identified by a special icon
.
To expand the group to view the group members:
» Double-click the group name or click the Expand
button.
The address book entities and sub-groups of the group is displayed in the right group list pane. You
can drill down the sub-group to view address book entities in the sub-group.
To move up to the next level and view the members in the upper level:
» Double-click the navigation arrow
button in the group members pane.
To collapse a group:
» Double-click the group name or click the Collapse
button.
Participants List Pane Information
The Participants List pane displays the following information for each participant:
Participants List Pane
Field/Option
Description
Type
Indicates whether the participant is a video (
Name
Displays the name of the participant.
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Field/Option
Description
IP Address/Phone
Enter the IP address of the participant’s endpoint.
• For H.323 participant define either the endpoint IP address or alias.
• For SIP participant define either the endpoint IP address or the SIP address.
Note: This field is removed from the dialog box when the ISDN/PSTN protocol is selected.
Network
The network communication protocol used by the endpoint to connect to the conference:
H.323 or SIP or ISDN/PSTN.
Dialing Direction
Dial-in – The participant dials in to the conference.
Dial-out – The Collaboration Server dials out to the participant.
Encryption
Displays whether the endpoint uses encryption for its media.
The default setting is Auto, indicating that the endpoint must connect according to the
conference encryption setting.
For information on adding and modifying participants in the Address Book, see Managing the Address Book.
Displaying and Hiding the Address Book
The Address Book can be hidden it by clicking the anchor pin ( ) button in the pane header. The Address
Book pane closes and a tab is displayed at the right edge of the screen.
» Click the tab to re-open the Address Book.
Adding Participants from the Address Book
You can add individual participants or a group of participants from the Address Book to a conference.
Adding Individual Participants from the Address Book to Conferences
You can add a participant or multiple participants to a new conference, ongoing conferences, or to
Conference Templates by using the drag-and-drop operation.
In SVC-based conferences, only dial-in participants can be added from the address book.
Multiple selection of group levels is not available.
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To add a participant to a new conference or an ongoing conference:
1 In the Address Book Navigation pane, select the group from which to add participants.
2 In the Address Book List pane, select the participant or participants you want to add to the
conference.
3 Click and hold the left mouse button and drag the selection to the Participants pane of the
conference.
The participants are added to the conference.
Adding a Group from the Address Book to Conferences
You can add a group of participants to a new conference, ongoing conferences, or to Conference Templates
by using the drag-and-drop operation.
To add a group to a new conference or an ongoing conference:
1 In the Address Book Navigation pane, select the group you want to add to the conference.
2 Click and hold the left mouse button and drag the selection to the Participants pane of the
conference.
The participants in the group level and all sub-levels are added to the conference.
Participant Groups
A group is a predefined collection of participants. A group provides an easy way to manage clusters of
participants that are in the same organizational structure and to connect a combination of endpoints to a
conference. For example, if you frequently conduct conferences with the marketing department, you can
create a group called “Marketing Team” that contains the endpoints of all members of the marketing team.
Groups can contain participants and sub-groups. You can define up to ten levels in the Main group.
Managing Groups in the Address Book
To manage the groups in the Address Book:
1 In the Address Book Navigation pane, right-click the group you want to manage.
The Groups menu is displayed.
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2 Select one of the following actions:
:
Address Book Navigation
Action
Description
New Group
Creates a new group within the current group.
New Participant
Adds a new participant within the current group.
Copy Group
Copies the current group to be pasted as an additional group.
Paste Group
Places the copied group into the current group. The group name of the
copied group is defined with “Copy” at the end of the group name. This
action is only available after a Copy Group action has been implemented.
Paste Participant
Places the copied participant into the current selected group. This action
is available after a Copy or Cut action was activated when selecting a
single participant or multiple participants.
Paste Participant as New
Pastes as a new participant into the selected group. This paste action
adds “Copy” at the end of the participant name. This action is only
available after a Copy action was activated for a single participant.
Rename Group
Renames the group name.
Delete Group
Deletes the group and all of its members. This action displays a message
requesting confirmation to delete the group and all members connected
with the group.
Additionally, you can drag a group from one location in the Address Book to another location, moving the
group and all its members, including sub-groups, to its new location using the drag-and-drop operation.
Moving a group to a new location can be done in the navigation pane or the list pane.
To drag a group from a location in the address book to another location:
1 Select the group you want to move.
2 Click and hold the left mouse button and drag the selection to the new location. The new location
can be either the Main root level or another group level.
The group and all its members (participants and groups) are moved to the new address book location.
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Managing the Address Book
Guidelines
● The multi-level Address Book can only be used in a local configuration on the Collaboration Server.
The hierarchical structure cannot be implemented with the Global Address Book (GAB).
● Up to ten levels can be defined in the hierarchical structure of the Address Book.
● The default name of the root level is Main. The Main root level cannot be deleted but the root level
name can be modified.
● Address Book names support multilingual characters.
● Participants in the Address Book can be copied to multiple groups. However, only one participant
exists in the Address Book. Groups that contain the same participants refer to the same definition of
the participant entity.
Adding a Participant to the Address Book
Adding participants to the Address Book can be performed by the following methods:
● Directly in the Address Book.
● Moving or saving a participant from an ongoing conference to the Address Book.
Only defined dial-out ISDN/PSTN participants can be added to the Address Book or ongoing conferences.
ISDN/PSTN participants are added to the Address Book in the same manner that H.323 and SIP
participants are added.
When adding dial-out participants to the ongoing conference, the system automatically dials out to the
participants using the Network Service (ISDN/PSTN or IP) defined for the connection in the participant
properties.
Adding a New participant to the Address Book Directly
You can add a new participant to the Main group or to a group in the Address Book. Additionally, you can
add a participant from a new conference, ongoing conference, or Conference Template.
To add a new participant to the Address Book:
1 In the Address Book - Navigation pane, select the group to where you want to add the new
participant.
2 Click the New Participant button ( ) or right-click the group to where you want to add the
participant and select the New Participant option.
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 Alternatively, click anywhere in the List pane and select the New Participant option.
The New Participant - General dialog box opens.
3 Define the following fields:
New Participant - General
Field
Description
Name
Enter the name of the participant or the endpoint as it will be displayed in the
Collaboration Server Web Client.
The Name field can be modified using Unicode encoding.
• English text uses ASCII encoding and can contain the most characters (length
varies according to the field).
• European and Latin text length is approximately half the length of the maximum.
• Asian text length is approximately one third of the length of the maximum.
Maximum field length in ASCII is 80 characters.
The maximum length of text fields varies according to the mixture of character sets
used (Unicode and ASCII).
This field may not be left blank. Duplicate participant names, comma, and semi-colon
characters may not be used in this field.
This name can also become the endpoint name that is displayed in the video layout.
For more details about endpoint (site) names, see Site Names Definition
Note: This field is displayed in all tabs.
Endpoint Website
(IP only)
Click the Endpoint Website hyperlink to connect to the internal website of the
participant’s endpoint. It enables you to perform administrative, configuration and
troubleshooting activities on the endpoint.
The connection is available only if the IP address of the endpoint’s internal site is
defined in the Website IP Address field.
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New Participant - General
Field
Description
Dialing Direction
Select the dialing direction:
• Dial-in – The participant dials in to the conference. This field applies to IP
participants only.
• Dial-out – The MCU dials out to the participant.
Note:
• Dial-out is forced when defining an ISDN/PSTN participant.
Type
The network communication protocol used by the endpoint to connect to the
conference: H.323, or SIP or ISDN/PSTN.
The fields in the dialog box change according to the selected network type.
IP Address
(H.323 and SIP)
Enter the IP address of the participant’s endpoint.
• For H.323 participant define either the endpoint IP address or alias.
• For SIP participant define either the endpoint IP address or the SIP address.
For Collaboration Servers registered to a gatekeeper, the Collaboration Server can be
configured to dial and receive calls to and from H.323 endpoints using the IP address
in the event that the Gatekeeper is not functioning.
Note: This field is hidden for ISDN/PSTN participants.
Participant Number
(ISDN/PSTN Only)
Enter the phone number of the ISDN/PSTN participant.
Note: This field is only displayed when the ISDN/PSTN protocol is selected.
Alias Name/Type
(H.323 Only)
If you are using the endpoint’s alias and not the IP address, first select the type of
alias and then enter the endpoint’s alias:
• H.323 ID (alphanumeric ID)
• E.164 (digits 0-9, * and #)
• Email ID (email address format, e.g. abc@example.com)
• Participant Number (digits 0-9, * and #)
Notes:
• Although all types are supported, the type of alias is dependent on the gatekeeper’s
capabilities. The most commonly supported alias types are H.323 ID and E.164.
• This field is used to enter the Entry Queue ID, target Conference ID and
Conference Password when defining a cascaded link.
• Use of the E.164 Number is dependent on the setting of the
REMOVE_IP_IF_NUMBER_EXISTS System Flag. For more information see
Substituting E.164 Number in Dial String.
• This field is removed from the dialog box when the ISDN/PSTN protocol is selected.
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New Participant - General
Field
Description
SIP Address/Type
(SIP Only)
Select the format in which the SIP address is written:
• SIP URI - Uses the format of an E-mail address, typically containing a user name
and a host name: sip:[user]@[host]. For example, sip:dan@polycom.com.
Note: If the SIP Address field contains an IPv6 address, it must be surrounded by
square brackets, for example, [::1].
• TEL URI - Used when the endpoint does not specify the domain that should
interpret a telephone number that has been input by the user. Rather, each domain
through which the request passes would be given that opportunity.
For example, a user in an airport might log in and send requests through an outbound
proxy in the airport. If the users enters 411 (this is the phone number for local
directory assistance in the United States), this number needs to be interpreted and
processed by the outbound proxy in the airport, and not by the user's home domain.
In this case, tel: 411 is the correct choice.
Note: This field is removed from the dialog box when the ISDN/PSTN protocol is
selected.
Endpoint Website
IP Address
(IP only)
Enter the IP address of the endpoint’s internal site to enable connection to it for
management and configuration purposes.
This field is automatically completed the first time that the endpoint connects to the
Collaboration Server. If the field is blank it can be manually completed by the system
administrator. The field can be modified while the endpoint is connected
Audio Only
Select this check box to define the participant as a voice participant, with no video
capabilities.
Extension/Identifier
String
Dial-out participants that connect to an external device such as Cascaded Links or
Recording Links may be required to enter a conference password or an identifying
string to connect. Enter the required string as follows:
[p]…[p][string]
For example: pp4566#
p (optional) - Indicates a pause of one second before sending the DTMF string. Enter
several concatenated [p]s to increase the delay before sending the string. The
required delay depends on the configuration of the external device or conference IVR
system.
String - Enter the required string using the digits 0-9 and the characters * and #. The
maximum number of characters that can be entered is identical to the H.323 alias
length.
If the information required to access the device/conference is composed of several
strings, for example, the conference ID and the conference password, this information
can be entered as one string, where pauses [p] are added between the strings for the
required delays, as follows:
[p]…[p][string][p]…[p] [string]...
For example: p23pp*34p4566#
Extension/Identifier
String
The Collaboration Server automatically sends this information upon connection to the
destination device/conference. The information is sent by the Collaboration Server as
DTMF code to the destination device/conference, simulating the standard IVR
procedure.
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4 Usually, additional definitions are not required and you can use the system defaults for the
remaining parameters. In such a case, click OK.
To modify the default settings for advanced parameters, click the Advanced tab.
5 Define the following Advanced parameters:
New Participant - Advanced
Field
Description
Video Bit Rate / Auto
(IP Only)
The Auto check box is automatically selected to use the Line Rate defined for
the conference.
Note: This check box cannot be cleared when defining a new participant during
an ongoing conference.
To specify the video rate for the endpoint, clear this check box, and then select
the required video rate.
Video Protocol
Select the video compression standard that will be forced by the MCU on the
endpoint when connecting to the conference: H.261, H.263, H.264 or RTV.
Select Auto to let the MCU select the video protocol according to the endpoint’s
capabilities.
Resolution
The Auto check box is automatically selected to use the Resolution defined for
the conference.
To specify the Resolution for the participant, select the required resolution from
the drop-down menu.
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New Participant - Advanced
Field
Description
Broadcasting Volume +
Listening Volume
To adjust the volume the participant broadcasts to the conference or the volume
the participant hears the conference, move the slider; each unit represents an
increase or decrease of 3 dB (decibel). The volume scale is from 1 to 10, where
1 is the weakest and 10 is the strongest. The default connection value is 5.
Encryption
Select whether the endpoint uses encryption for its connection to the
conference.
Auto (default setting) indicates that the endpoint will connect according to the
conference encryption setting.
AGC
AGC (Auto Gain Control) mechanism regulates noise and audio volume by
keeping the received audio signals of all participants balanced. Select this check
box to enable the AGC mechanism for participants with weaker audio signals.
Notes:
• To be enable AGC, set the value of the ENABLE_AGC System Flag in
system.cfg to YES. The flag’s default value is NO.
• If the System Flag does not exist in the system, it must be manually added to
the System Configuration. For more information see Modifying System Flags.
• Enabling AGC may result in amplification of background noise.
Cascaded (IP Only)
If this participant is used as a link between conferences select:
• Slave, if the participant is defined in a conference running on a Slave MCU.
• Master, if the participant is defined in a conference running on the Master
MCU.
It enables the connection of one conference directly to another conference using
an H.323 connection only. The conferences can run on the same MCU or
different MCU’s. For more information, see Basic Cascading Using IP Cascaded
Link.
Precedence Domain
Name
(Dial-out SIP Only)
When Multi Level Precedence and Preemption is used, this is the Precedence
Domain Name for the participant.
For more information see MLPP (Multi Level Precedence and Preemption).
Precedence Level
(Dial-out SIP Only)
When Multi Level Precedence and Preemption is used, this is the Precedence
Level for the participant
For more information see MLPP (Multi Level Precedence and Preemption).
AGC
The Audio Gain Control (AGC) protocol that reduces noises is enabled by
default for the participants.
Clear this check box to disable the AGC feature.
ISDN/PSTN Network
Service
Enables you to select the ISDN/PSTN network service.
6 To add general information about the participant, such as e-mail, company name, and so on, click
the Information tab and type the necessary details in the Info 1-4 fields. Text in the info fields can
be added in Unicode format (length: 31 characters).
7 Click OK.
The new participant is added to the selected group in the address book.
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Substituting E.164 Number in Dial String
Between the time a conference is scheduled and when it becomes active, the IP of an endpoint may change,
especially in an environment that uses DHCP. The MCU can be set to ignore the IP address of a participant
when the conference starts. Instead, the alternative E.164 number will be used.
The flag, REMOVE_IP_IF_NUMBER_EXISTS controls this option. This flag must be manually added to
change its value. The values of this flag are:
● YES (default) - The IP address of an endpoint will be ignored if an E.164 number (or other) exists.
● NO - The IP address of an endpoint will be used.
Guidelines for Substituting E.164 Number in Dial String
● When this feature is enabled, the IP address field of participants in scheduled conferences and
conference templates will be empty.
● In order for the MCU to ignore the IP of H.323 participants, the following requirements must be met:
 A gatekeeper must be defined.
 The alias of the participant must be defined.
 The alias type must be defined (not set to None).
● If an H.323 gatekeeper is defined but is not connected, the MCU will fail to connect to H.323 dial-out
participants.
● In order for the MCU to ignore the IP of SIP participants, the following requirements must be met:
 A SIP proxy must be defined.
 The SIP address must be defined.
● If a SIP proxy is defined but is not connected, the MCU will fail to connect to SIP dial-out participants.
Adding a Participant from an Ongoing Conference to the Address
Book
You can add a participant to the Address Book directly from an ongoing conference.
When adding a participant to the address book from a new conference, Participants list of an
ongoing conference or Conference Template, the participant is always added to the “Main” group.
To add a participant from the conference to the Address Book:
1 During an ongoing conference, select the participant in the Participant pane, and either click the
Add Participant to Address Book button ( ), or right-click and select Add Participant to
Address Book.
The participant is added to the Address Book.
Alternatively, you could:
a Double-click the participant’s icon, or right-click the participant icon and select Participant
Properties.
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The Participant Properties window opens.
b Click the Add to Address Book button.
If the participant name is already listed in the All Participants list, an error message is displayed. In
such a case, change the name of the participant before adding the participant to the address book.
Modifying Participants in the Address Book
When required, you can modify the participant’s properties.
To modify participant properties in the Address Book:
1 In the Address Book - Navigation pane, select the group to where the participant to modify is
listed.
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2 In the Address Book - List pane, double-click the participant’s icon.
The Participant’s Properties window is displayed.
3 Modify the necessary properties in the window, such as dialing direction, communication protocol
type, and so on. You can modify any property in any of the three tabs: General, Advanced and Info.
4 Click OK.
The changes to the participant’s properties are updated.
Deleting Participants from the Address Book
To delete participants from the Address Book:
1 In the Address Book - Navigation pane, select the group where the participant to delete is listed.
2 In the Address Book - List pane, either select the participant to delete, and then select the Delete
Participant ( ) button, or right-click the participant icon and then select the Delete Participant
option.
3 A confirmation message is displayed depending on the participant’s assignment to groups in the
address book:
a When the participant belongs to only one group: click Yes to permanently delete the participant
from the address book.
b When the participant belongs to multiple groups, a message is displayed requesting whether to
delete the participant from the Address Book or from the current selected group. Select:
 Current group to delete the participant from the selected group
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 Address Book to permanently delete the participant from the address book (all groups).
Click OK to perform the delete operation, or Cancel to exit the delete operation.
Copying or Moving a Participant
You can copy or move a participant from one group to another group using the Copy, Cut, and Paste options.
A participant can belong to multiple groups. However, there is only one entity per participant. Groups that
contain the same participants refer to the same definition of the participant entity. Alternatively, you can drag
a participant from one location in the Address Book to another location, moving the participant to its new
location using the drag-and-drop operation.
The cut and copy actions are not available when selecting multiple participants.
To copy or move a participant to another group:
1 In the Address Book - Navigation pane, select the group from where to copy the participant.
2 In the Address Book - List pane, select the participant you want to copy.
3 Right-click the selected participant, and select one of the following functions from the drop-down
menu:
Copy / Move Participant
Function
Description
Copy Participant
Copies the participant to be pasted into an additional group.
Cut Participant
Moves the participant from the current group to a different group. Alternatively,
you can move a participant to another location by dragging the participant to the
new location.
4 In the Address Book - Navigation pane, navigate and select the group in which you want to paste
the participant.
5 Right-click the selected group, and click one of the following Paste functions from the drop-down
menu:
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Paste Participant
Function
Description
Paste Participant
Creates a link to the participant entity in the pasted location.
Paste Participant as New
Pastes as a new participant into the selected group. This paste action adds
“Copy” to the end of the participant name.
To drag a participant from an address book group to another group:
1 Select the participant or participants you want to move.
2 Click and hold the left mouse button and drag the selection to the new group.
The participants are moved to the new address book group.
Searching the Address Book
You can search the Address Book for a participant’s name or a group name only on the currently selected
group/level.
To search for participants or groups in the current selected level:
1 In the Address Book - Navigation pane, select the group/level within to run the search.
2 In the Address Book toolbar, activate the search option by clicking the Find field.
The field clears and a cursor appears indicating that the field is active.
3 Type all or part of the participant’s name or group name and click the search button.
The closest matching participant entries are displayed and the Active Filter indicator turns on.
Filtering the Address Book
The entries in an address book group can be filtered to display only the entries (participants or groups) that
meet criteria that you specify and hides entries that you do not want displayed. It enables you to select and
work with a subset of Address Book entries.
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You can filter by more than one column, by adding additional filters (columns).
The filter applies to the displayed group. If All Participants option is selected, it applies to all the listed
participants.
Filtering can be done using:
● A predefined pattern
● Customized pattern
When you use the Find dialog box to search filtered data, only the data that is displayed is searched; data
that is not displayed is not searched. To search all the data, clear all filters.
Filtering Address Book Data Using a Predefined Pattern
To filter the data in an address book group:
1 In the Address Book - Navigation pane, select the group to filter.
2 In the Address Book - List pane, in the column that you want to use for filtering, click the filter ( )
button.
A drop-down menu is displayed containing all the matching patterns that can be applied to the
selected field.
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3 Click the matching pattern to be applied.
The filtered list is displayed with a filter indicator ( ) displayed in the selected column heading.
Example: If the user selects 172.21.41.104 as the matching pattern, the filtered group in the Address
Book is displayed as follows:
Filtering Address Book Data Using a Custom Pattern
To filter the data in an address book group:
1 In the Address Book - Navigation pane, select the group to filter.
2 In the Address Book - List pane, in the column that you want to use for filtering, click the filter ( )
button.
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3 Select the (Custom) option from the drop-down list.
The Custom Filtering dialog box opens.
4 In the Condition - Column text matches field, enter the filtering pattern.
For example, to list only endpoints that include the numerals 41 in their name, enter 41.
5 To add filtering patterns to further filter the list or fine tune your search, click Add Condition.
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6 To clear a filtering pattern, click Clear Condition.
The filtered list is displayed with an active filter (blue) indicator ( ) displayed in the selected column
heading.
For example, if the filtering pattern is 41, the participants list includes all the endpoints that contain
the numerals 41 in their name.
Clearing the Filter
To clear the filter and display all entries:
1 In the filtered Address Book column heading, click the Active Filter indicator.
The pattern matching options menu is displayed.
2 Click (All).
The filter is deactivated and all the group/level entries are displayed.
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Obtaining the Display Name from the Address Book
The MCU can be configured to replace the name of the dial-in participant as defined in the endpoint (site
name) with the name defined in the Address Book.
In this process, the system retrieves the data (name, alias, number or IP address) of the dial-in participant
and compares it first with the conference defined dial-in participants and if the endpoint is not found, it then
searches for the endpoint with entries in the address book. After a match is found, the system displays the
participant name as defined in the address book instead of the site name, in both the video layout and the
Collaboration Server Web Client/Manager.
The system compares the following endpoint data with the address book entries:
● For H.323 participants, the system compares the IP address, Alias, or H.323 number.
● For SIP participants, the system compares the IP address or the SIP URI.
Guidelines for Obtaining the Display Name from the Address Book
● Only Users with Administrator and Operator Authorization Levels are allowed to enable and disable
the Obtain Display Name from Address Book feature.
● This feature is supported for IPv4 participants only.
Enabling and Disabling the Obtain Display Name from Address Book Feature
The Obtain Display Name from Address Book option can be enabled for all participants connecting to the
MCU if the name of the participants are defined in the Address Book.
To enable or disable the Obtain Display Name from Address Book option:
1 On the Collaboration Server main menu bar, select Setup > Customize Display Settings >
Ongoing Conferences.
The Ongoing Conferences dialog box is displayed.
2 Select the Obtain display name from address book check box to enable the feature or clear the
check box to disable the feature.
3 Click OK.
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Importing and Exporting Address Books
Address Books are proprietary Polycom data files that can only be distributed among Collaboration Server
units. The Address Books are exported in XML format, which are editable offline. If no name is assigned to
the exported Address Book, the default file name is:
EMA.DataObjects.OfflineTemplates.AddressbookContent_.xml
Exporting an Address Book
To Export an Address Book:
1 In the Address Book pane, click the Export Address Book (
in the pane, and click Export Address Book.
) button, or right-click an empty area
The Export Address Book dialog box is displayed.
2 Enter the desired path, or click the Browse button.
3 In the Save Address Book dialog box, select the directory to save the file. You may also rename
the file in the File Name field.
4 Click Save.
You will return to the Export File dialog box.
5 Click OK.
The exported Address Book is saved in the selected folder in XML format.
Importing an Address Book
To Import and Address Book:
1 In the Address Book pane, click the Import Address Book (
area in the pane, and then click Import Address Book.
) button, or right-click an empty
The Import Address Book dialog box is displayed.
2 Enter the path from which to import the Address Book, or click the Browse button.
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3 In the Open dialog box navigate to the desired Address Book file (in XML format) to import.
When importing an Address Book, participants with exact names in the current Address Book will be
overwritten by participants defined in the imported Address Book.
4 Click Open.
You will return to the Import File dialog box.
5 Click OK.
The Address Book is imported and a confirmation message is displayed at the end of the process.
6 Click Close.
Upgrading and Downgrading Considerations
When upgrading to a multi-level address book version from a single-level address book, the following
factors have to be taken into consideration:
● The system automatically creates a new address book with a different name and modifies the new
address book to a multi-level hierarchical address book.
● By default, the address book contains two levels:
 The top level (root) named Main.
 Second level - All address book groups from the single-level address book are placed under the
Main group with their associated participants.
● Participants that were not previously associated with any group in the Address Book are placed in the
Main group.
● All participants in the Address Book appear in the All Participants group.
● During the upgrade process, the single-level address book file is save in the system to enable a future
the downgrade of the version to a previous, single-level address book version (if required).
When downgrading from a multi-level address book version to a single-level address book version, the
multi-level address book is replaced during the downgrade process by the single-level address book that
was saved during the upgrade process.
Integrating the Global Address Book (GAB) with the
Collaboration Server
The RealPresence Resource Manager includes a Global Address Book (GAB) with all registered endpoints.
This address book can be used by the RealPresence Collaboration Server users to add participants to
conferences.
Integration with Resource Manager
● The RealPresence Collaboration Server can use only one address book at a time.
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● The Collaboration Server uses the RealPresence Resource Manager address book in read-only
mode. You can only add or modify the address book entries from the RealPresence Resource
Manager.
To Integrate the RealPresence Resource Manager Global Address Book (GAB) with the
Collaboration Server:
1 In the RealPresence Resource Manager, manually add the Polycom Collaboration Server system to
the RealPresence Resource Manager as directed in the RealPresence Resource Manager
Operations Guide.
2 In the RealPresence Resource Manager, add a user or use an existing user for Collaboration Server
login as directed in the RealPresence Resource Manager Operations Guide.
Write down the User Name and Password as they will be used later to define the Collaboration Server
connection to the RealPresence Resource Manager Address Book.
RealPresence Collaboration Server Side
1 On the RealPresence Collaboration Server menu, click Setup > System Configuration.
The System Flags - MCMS_PARAMETERS_USER dialog box opens.
2 Modify the values of the flags in the table below.
For more information, see Modifying System Flags.
3 Click OK to complete the definitions.
4 When prompted, click Yes to reset the MCU and implement the changes to the system
configuration.
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Scheduling Reservations
The Reservations option enables users to schedule conferences. These conferences can be launched
immediately or become ongoing, at a specified time on a specified date.
Scheduling a conference reservation requires definition of conference parameters such as the date and time
at which the conference is to start, the participants and the duration of the conference.
Scheduled conferences (Reservations) can occur once or repeatedly, and the recurrence pattern can vary.
By default, the Scheduler is enabled by a System Flag. The flag prevents potential scheduling conflicts from
occurring as a result of system calls from external scheduling applications such as RealPresence Resource
Manager, ReadiManager®, SE200, and others via the API.
If an external scheduling application is used, the flag INTERNAL_SCHEDULER must be manually added
to the System Configuration and its value must be set to NO.
For more information see Modifying System Flags.
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
Note: Using Reservations with RealPresence DMA
In the Polycom® RealPresence® Web Suite solution, Reservations are defined in the RealPresence
DMA system component and should not be defined directly in the RealPresence Collaboration Server
(RMX) 1800/2000/4000 component.
System resources are calculated according to the Collaboration Server’s license. For more information, see
Forcing Video Resource Allocation to CIF Resolution.
System resource availability is partially checked when reservations are created:
● If a conference duration extension request is received from an ongoing conference, the request is
rejected if it would cause a resource conflict.
● If several reservations are scheduled to be activated at the same time and there are not enough
resources for all participants to be connected:
 The conferences are activated.
 Participants are connected to all the ongoing conferences until all system resources are used up.
If sufficient resources are not available in the system and a scheduled Reservation cannot be activated, the
Reservation is deleted from the schedule.
In RealPresence Collaboration Server (RMX) 1800/2000/4000 resources for Reservations are also
calculated using the Reserve Resources for Audio/Video Participants fields of the New Reservation dialog
box.
Resources are reserved for participants at the highest video resolution supported by the Line Rate specified
in the conference Profile and up to the maximum system video resolution specified by the Resolution
Configuration dialog box.
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When a new Reservation is created in the Reservation Calendar, the effect of the new Reservation
(including its recurrences) on available resources is checked. If resource deficiencies are found an error
message is displayed.
Defined dial-in or dial-out participants, Meeting Rooms, Entry Queues and new connections to Ongoing
conferences are not included in the resources calculation.
A Reservation that has been activated and becomes an ongoing conference is deleted from the Reservation
Calendar list.
The maximum number of concurrent reservations is 80. Reservations with durations that overlap (for any
amount of time) are considered to be concurrent.
System resource availability is partially checked when reservations are created:
● If a conference duration extension request is received from an ongoing conference, the request is
rejected if it would cause a resource conflict.
● If several reservations are scheduled to be activated at the same time and there are not enough
resources for all participants to be connected:
 The conferences are activated.
 Participants are connected to all the ongoing conferences until all system resources are used up.
A scheduled Reservation cannot be activated and is deleted from the schedule if:
● An Ongoing conference has the same Numeric ID.
● Sufficient resources are not available in the system.
If a problem prevents a Reservation from being activated at its schedule time, the Reservation will not be
activated at all. This applies even if the problem is resolved during the Reservation’s scheduled time slot.
A Profile that is assigned to a Reservation cannot be deleted.
Reservations are backed up and restored during Setup > Software Management >Backup / Restore
Configuration operations. For more information see Software Management.
All existing reservations are erased by the Standard Restore option of the Administration > Tools >
Restore Factory Defaults procedure.
Reservations can also be scheduled from Conference Templates. For more information see Scheduling a
Reservation From a Conference Template.
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Using the Reservation Calendar
To open the Reservation Calendar:
● In the RMX Management pane, click the Reservation Calendar button (
).
Toolbar Buttons
The toolbar buttons functions are described in the table below.
Reservations – Toolbar Buttons
Button
Description
New Reservation
Create a new reservation. The date and time of the new reservation is set according to
the highlighted blocks on the Reservation Calendar.
Click to delete the selected reservation.
Delete Reservation
Back
Next
Today
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Click to show the previous day or week, depending on whether Show Day or Show
Week is the selected.
Click to show the next day or week, depending on whether Show Day or Show Week is
the selected.
Click to show the current date in the Reservation Calendar in either Show Day or
Show Week view.
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Button
Show Week
Description
Change the calendar view to weekly display, showing a calendar week: Sunday
through Saturday
Click this button to show the day containing the selected time slot.
Show Day
Click to change to List View and display a list of all reservations.
Reservations List
Used to search for reservations by Display Name. (Available in Reservations List view
only).
Reservations Views
The Reservation Calendar list has the following views available:
● Week
● Day
● Today
● List
In all views the Main Window List Pane header displays the total number of reservations in the system.
Week View
By default the Reservation Calendar is displayed in Week view with the current date highlighted in orange.
Day View
A single day is displayed.
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Today View
The current date (Today), highlighted in orange, can be viewed in both Week View and Day View.
List View
List View does not have a calendar based format.
All Reservations are listed by:
•
Display Name
•
End Time
•
ID
•
Status
•
Internal ID
•
Conference Password
•
Start Time
•
Profile
The Reservations can be sorted, searched and browsed by any of the listed fields.
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Changing the Calendar View
To change between Week and Day views:
1 In Week View: In the Reservation Calendar toolbar, click Show Day (
) to change to Day View.
or
In Day View: In the Reservation Calendar toolbar, click Show Week (
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To view Today (the current date):
● In Week View or Day View, in the Reservation Calendar toolbar, click the Today (
the current date displayed within the selected view.
) button to have
To change to List View:
1 In the Reservation Calendar toolbar, click, the Reservations List (
) button.
The Reservations List is displayed.
2 To sort the data by any field (column heading), click on the column heading.
A
or
symbol is displayed in the column heading indicating that the list is sorted by this field, as
well as the sort order.
3 To toggle the column’s sort order, click on the column heading.
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To return to Calendar View:
● In the Reservation Calendar toolbar, click any of the buttons (Show Week/Show Day/Today) to
return to the required Reservation Calendar view.
Scheduling Conferences Using the Reservation
Calendar
Creating a New Reservation
There are three methods of creating a new reservation:
● Method I – Creating a reservation with default duration of 1 hour
● Method II – Creating a reservation with default duration of ½ hour
● Method III – Interactively define the reservation duration
Each method requires the selection of a starting time slot in the Reservation Calendar. The default time slot
is the current half-hour period of local time.
In all views, if the New Reservation (
) button is clicked without selecting a starting time slot, or if a time
slot is selected that is in the past, the Reservation becomes an Ongoing conference immediately and is not
added to the Reservations calendar.
After selecting a starting time slot in the Reservation Calendar you can create a reservation with a default
duration derived from the creation method used or by interactively defining the duration of the reservation.
Method I – To create a reservation with default duration of 1 hour:
» In the Reservation Calendar toolbar, click the New Reservation (
of 1 hour duration.
) button to create a reservation
Method II – To create a reservation with default duration of ½ hour:
» Right-click and select New Reservation to create a reservation of ½ hour default duration.
Method III – To interactively define the duration:
1 In the calendar, click & drag to expand the time slot to select the required Date, Start Time and
Duration for the reservation.
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2 In the Reservation Calendar toolbar, click the New Reservation (
select New Reservation.
) button or right-click and
Example: The following click & drag sequence would select a reservation for Tuesday, August 12,
2008, starting at 01:00 with a duration of 4 hours.
The duration of reservations created by any of the above methods can be modified in the Scheduler
tab of the New Reservation dialog box.
To create a new reservation:
1 Open the Reservation Calendar.
2 Select a starting time slot.
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3 Create the reservation using one of the three methods described above.
The New Reservation – General tab dialog box opens.
All the fields are the same as for the New Conference – General dialog box. For more information,
refer to General Tab.
New Reservation – Reserved Resources
Field
Description
Reserve Resources
for Video Participants
Enter the number of video participants for which the system must
reserve resources.
Default: 0 participants.
Reserve Resources
for Audio Participants
Enter the number of audio participants for which the system must
reserve resources.
Default: 0 participants.
Note: Changes before activation
When a Conference Profile is assigned to a Meeting Room or a Reservation, the Profile’s parameters
are not embedded in the Reservation, and are taken from the Profile when the reservation becomes an
ongoing conference. Therefore, any changes to the Profile parameters between the time the
Reservation or Meeting Room was created and the time that it is activated (and becomes an ongoing
conference) will be applied to the conference.
If the user wants to save the current parameters, a different Profile with these parameters must be
assigned, or a different Profile with the new parameters must be created.
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4
Click the Schedule tab.
5 Adjust the new reservation’s schedule by modifying the fields as described in the table below.
New Reservation – Schedule Tab
Field
Description
Start Time
Select the Start
Time of the
Reservation.
End Time
Select the End
Time of the
Reservation.
Recurring
Meeting
Select this option to set up a Recurring Reservation - a series of Reservations to be repeated
on a regular basis.
To create a recurring reservation, you must define a time period and a recurrence pattern of
how often the Reservation should occur: Daily, Weekly or Monthly.
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•
The Start/End Times of the Reservation are initially taken from the
time slot selected in the Reservation Calendar.
• The Start/End Times can be adjusted by typing in the hours and
minutes fields or by clicking the arrow buttons.
• The Start/End dates can be adjusted by typing in the date field or
by clicking the arrow buttons or using the calendar.
• The start time of all the reservations can be manually adjusted in
one operation. For more information see Adjusting the Start Times
of all Reservations.
• End Time settings are initially calculated as Start Time + Duration.
End Time settings are recalculated if Start Time settings are
changed.
Changes to End Time settings do not affect Start Time settings.
However, the Duration of the Reservation is recalculated.
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New Reservation – Schedule Tab
Field
Description
Recurrence
Pattern
Daily
If Daily is selected, the system automatically selects all the days of
the week. To de-select days (for example, weekends) clear their
check boxes.
Weekly
If Weekly is selected, the system automatically selects the day of the
week for the Reservation from the day selected in the Reservation
Calendar.
You can also define the recurrence interval in weeks. For example, if
you want the reservation to occur every second week, enter 2 in the
Recur every _ week(s) field.
To define a twice-weekly recurring Reservation, select the check box
of the additional day of the week on which the Reservation is to be
scheduled and set the recurrence interval to 1.
Monthly
If Monthly is selected, the system automatically selects the day of the
month as selected in the Reservation Calendar. You are required to
choose a recurrence pattern:
• Day (1-31) of every (1-12) month(s) - Repeats a conference on a
specified day of the month at a specified monthly interval. For
example, if the first Reservation is scheduled for the 6th day of the
current month and the monthly interval is set to 1, the monthly
Reservation will occur on the 6th day of each of the following
months.
• The (first, second,...,last) (Sun-Sat) of x month(s) - Repeats a
Reservation in a particular week, on a specified day of the week at
the specified monthly interval. For example, a recurrent meeting on
the third Monday every second month.
A series of Reservations can be set to end after a specified number of occurrences or by a specific date.
Select one of the following methods of terminating the series of Reservations:
End After
End After: x Occurrences - Ends a recurring series of Reservations after a specific number
(x) of occurrences.
Default: 1
(Leaving the field blank defaults to 1 occurrence.)
End by Date
End By Date: mm/dd/yyyy - Specifies a date for the last occurrence of the recurring series
of Reservations. The End By Date value can be adjusted by typing in the date field or by
clicking the arrow button and using the calendar utility.
Default: Current date.
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6 Click the Participants tab.
The fields are the same as for the New Conference – Participants dialog box. For more information,
see Participants Tab.
Participant properties are embedded in the conferencing entity and therefore, if the participant
properties are modified in the Address Book (or Meeting Rooms) after the Reservation has been
created they are not applied to the participant when the Reservation is activated.
7 You can add participants from the Participants Address Book.
For more information see Adding Participants from the Address Book.
Between the time a conference is scheduled and when it becomes active, the IP of an endpoint may
change, especially in an environment that uses DHCP. The MCU can be set to ignore the IP address of
a participant when the conference starts. Instead, the alternative E.164 number will be used instead of
the IP address. For more information see Substituting E.164 Number in Dial String.
8 Optional. Add information to the reservation.
Information entered in the Information tab is written to the Call Detail Record (CDR) when the
reservation is activated. Changes made to this information before it becomes an ongoing conference
will be saved to the CDR.
For more information see Information Tab.
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9 Click OK.
The New Reservation is created and is displayed in the Reservation Calendar.
If you create a recurring reservation all occurrences have the same ID. A recurring Reservation is
assigned the same ISDN/PSTN dial-in number for all recurrences.
If a dial in number conflict occurs prior to the conference’s start time, an alert is displayed: ISDN
dial-in number is already assigned to another conferencing entity and the
conference cannot start.
The series number (_0000n) of each reservation is appended to its Display Name.
Example:
Conference Template name: Sales
Display Name for single scheduled occurrence: Sales
If 3 recurrences of the reservation are created:
Display Name for occurrence 1: Sales_00001
Display Name for occurrence 2: Sales_00002
Display Name for occurrence 3: Sales_00003
Managing Reservations
Reservations can be accessed and managed via all the views of the Reservations List.
Guidelines
● The Recurrence Pattern fields in the Schedule tab that are used to create multiple occurrences of
a Reservation are only displayed when the Reservation and its multiple occurrences are initially
created.
● As with single occurrence Reservations, only the Duration, Start Time and End Time parameters of
multiple occurrence reservations can be modified after the Reservation has been created.
● A single occurrence Reservation cannot be modified to become a multiple occurrence reservation.
● Reservations can only be modified one at a time and not as a group.
● If Reservations were created as a recurring series, the system gives the option to delete them
individually, or all as series.
Viewing and Modifying Reservations
Reservations can be viewed and modified by using the Week and Day views of the Reservations Calendar
or by using the Reservation Properties dialog box.
Using the Week and Day views of the Reservations Calendar
In the Week and Day views each Reservation is represented by a shaded square on the Reservation
Calendar. Clicking on a Reservation selects the Reservation. A dark blue border is displayed around the
edges of the Reservation indicating that it has been selected.
The Start Time of the Reservation is represented by the top edge of the square while the End Time is
represented by the bottom edge.
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The cursor changes to a vertical double arrow ( ) when it is moved over the top and bottom sides of the
square.
To move the Reservation to another time slot:
1 Select the Reservation.
2 Hold the mouse button down and drag the Reservation to the desired time slot.
3 Release the mouse button.
To change the Reservation’s Start time:
1 Select the Reservation.
2 Move the mouse over the top edge of the Reservation’s square.
3 When the cursor changes to a vertical double arrow ( ) hold the mouse button down and drag the
edge to the desired Start Time.
4 Release the mouse button.
To change the Reservation’s End time:
1 Select the Reservation.
2 Move the mouse over the bottom edge of the Reservation’s square.
3 When the cursor changes to a vertical double arrow ( ) hold the mouse button down and drag the
edge to the desired End Time.
4 Release the mouse button.
To View or Modify Reservations using the Reservation Properties dialog box:
1 In the Reservations List, navigate to the reservation (or its recurrences) you want to view, using the
Show Day, Show Week, Today, Back, Next or List buttons.
2 Double-click, or right-click and select Reservation Properties, to select the reservation to be
viewed or modified.
The Reservation Properties – General dialog box opens.
3 Select the tab(s) of the properties you want to view or modify.
4 Optional. Modify the Reservation Properties.
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5 Click OK.
The dialog box closes and modifications (if any) are saved.
Adjusting the Start Times of all Reservations
When utilizing GMT offset (for example, Daylight Saving Time change), the start time of the reoccurring
reservations scheduled before the Collaboration Server time change are not updated accordingly (although
their start times appear correctly in the Reservations list, when checking the reservation properties the start
time is incorrect).
Following the Collaboration Server time change, the start time of all reoccurring reservations must be
manually adjusted in one operation.
Using this option, the start times of all reservations currently scheduled on the Collaboration Server are
adjusted with the same offset.
To adjust the reoccurring reservations start time after the GMT Offset has been changed
for Daylight Saving Time (DST) or a physical move:.
Note: Adjustment of Reservation Time
Adjustment of Reservation Time should only be performed after adjustment of Collaboration Server
Time is completed as a separate procedure.
1 On the Collaboration Server menu, click Setup > RMX Time.
The RMX Time dialog box opens.
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2 Click the Adjust Reservations Time button.
The Adjust Reservations Time dialog box opens.
3 Click the arrows of the Offset - Hours box to indicate the number of hours to add or subtract from
the current start time; a positive value indicates adding time, while minus (-) indicates subtracting
time.
4 Click the arrows of the Offset - minutes box to indicate the number of minutes to add or subtract
from the current start time of the reservations. Increments or decrements are by 15 minutes.
For example, to subtract 30 minutes from the start time of all the reservation, enter 0 in the hours
box, and -30 in the minutes box.
To add one hour and 30 minutes to the start time, enter 1 in the hours box and 30 in the minutes box.
5 Click the Adjust button to apply the change to all the reoccurring reservations currently scheduled
on the Collaboration Server.
Note: Internal communication error message
When adjusting the start time of 1000 - 2000 reservations, an Internal communication error
message may appear. Ignore this message as the process completes successfully.
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Deleting Reservations
To delete a single reservation:
1 In the Reservations List, navigate to the reservation you want to delete, using the Show Day, Show
Week, Today, Back, Next or List buttons.
2 Click to select the reservation to be deleted.
3 Click the Delete Reservation (
) button.
or
Place the mouse pointer within the Reservation block, right-click and select Delete Reservation.
4 Click OK in the confirmation dialog box.
The Reservation is deleted.
To delete all recurrences of a reservation:
1 In the Reservations List, navigate to the Reservation or any of its recurrences, using the Show Day,
Show Week, Today, Back, Next or List buttons.
2 Click the Delete Reservation (
) button.
or
Place the mouse pointer within the Reservation or any of its recurrences, right-click and select Delete
Reservation.
A confirmation dialog box is displayed.
3 Select Delete the series.
4 Click OK.
All occurrences of the Reservation are deleted.
Searching for Reservations using Quick Search
Quick Search is available only in List View. It enables you to search for Reservations by Display Name.
To search for reservations:
1 In the Reservation Calendar toolbar, click in the Quick Search field.
The field clears and a cursor is displayed indicating that the field is active.
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2 Type all or part of the reservation’s Display Name into the field and click Search.
The closest matching Reservation entries are displayed.
3 To view or modify the Reservation:
Double- click the Reservation’s entry in the list to open the Reservations Properties dialog box.
or
Right -click the Reservation’s entry in the list and select a menu option to view, modify or delete the
Reservation.
To clear the search and display all reservations:
1 Clear the Quick Search field.
2 Click Search.
All Reservations are displayed.
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Operator Assistance & Participant Move
In the Polycom® RealPresence® Web Suite solution, the Conference Profiles are defined in the
RealPresence DMA system component and should not be defined directly in the RealPresence
Collaboration Server (RMX) 1800/2000/4000 component.
Operator conferences and participant move are supported in AVC CP Conferencing Mode only.
Users (operators) assistance to participants is available when:
● Participants have requested individual help (using *0 DTMF code) during the conference.
● Participants have requested help for the conference (using 00 DTMF code) during the conference.
● Participants have problems connecting to conferences, for example, when they enter the wrong
conference ID or password.
In addition, the user (operator) can join the ongoing conference and assist all conference participants.
Operator assistance is available only when an Operator conference is running on the MCU.
The Operator conference offers additional conference management capabilities to the Collaboration Server
users, enabling them to attend to participants with special requirements and acquire participant details for
billing and statistics. This service is designed usually for large conferences that require the personal touch.
Operator Conferences
An Operator conference is a special conference that enables the Collaboration Server user acting as an
operator to assist participants without disturbing the ongoing conferences and without being heard by other
conference participants. The operator can move a participant from the Entry Queue or ongoing conference
to a private, one-on-one conversation in the Operator conference.
In attended mode, the Collaboration Server user (operator) can perform one of the following actions:
● Participants connected to the Entry Queue who fail to enter the correct destination ID or conference
password can be moved by the user to the Operator conference for assistance.
● After a short conversation, the operator can move the participant from the Operator conference to the
appropriate destination conference (Home conference).
● The operator can connect participants belonging to the same destination conference to their
conference simultaneously by selecting the appropriate participants and moving them to the Home
conference (interactively or using the right-click menu).
● The operator can move one or several participants from an ongoing conference to the Operator
conference for a private conversation.
● The operator can move participants between ongoing Continuous Presence conferences.
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Operator Conference Guidelines
● An Operator conference can only run in Continuous Presence mode.
● Operator conference is defined in the Conference Profile. When enabled in Conference Profile, High
Definition Video Switching option is disabled.
● An Operator conference can only be created by a User with Operator or Administrator Authorization
level.
● Operator conference name is derived from the User Login Name and it cannot be modified.
● Only one Operator conference per User Login Name can be created.
● When created, the Operator conference must include one and only one participant - the Operator
participant.
● Only a defined dial-out participant can be added to an Operator conference as an Operator participant
● Once running, the Collaboration Server user can add new participants or move participants from
other conferences to this conference. The maximum number of participants in an Operator
conference is the same as in standard conferences.
● Special icons are used to indicate an Operator conference in the Ongoing Conferences list and the
operator participant in the Participants list.
● An Operator conference cannot be defined as a Reservation.
● An Operator conference can be saved to a Conference Template. An ongoing Operator conference
can be started from a Conference Template.
● The Operator participant cannot be deleted from the Operator conference or from any other
conference to which she/he was moved to, but it can be disconnected from the conference.
● When deleting or terminating the Operator conference, the operator participant is automatically
disconnected from the MCU, even if participating in a conference other than the Operator conference.
● Participants in Telepresence conferences cannot be moved from their conference, but an operator
can join their conference and help them if assistance is required.
● Moving participants from/to an Operator conference follows the same guidelines as moving
participants between conferences. For move guidelines, see Move Guidelines.
● When a participant is moved from the Entry Queue to the Operator conference, the option to move
back to the source (Home) conference is disabled as the Entry Queue is not considered as a source
conference.
● The conference chairperson cannot be moved to the Operator conference following the individual
help request if the Auto Terminate When Chairperson Exits option is enabled, to prevent the
conference from automatically ending prematurely. In such a case, the assistance request is treated
by the system as a conference assistance request, and the operator can join the conference.
Defining the Components Enabling Operator Assistance
To enable operator assistance for conferences, the following conferencing entities must be adjusted or
created:
● IVR Service (Entry Queue and Conference) in which Operator Assistance options are enabled.
● A Conference Profile with the Operator Conference option enabled.
● An active Operator conference with a connected Operator participant.
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To define a Conference IVR Service with Operator Assistance Options
1 In the RMX Management pane, expand the Rarely Used list.
2 Click the IVR Services (
) entry.
3 On the IVR Services toolbar, click the New Conference IVR Service (
) button.
The New Conference IVR Service - Global dialog box opens.
4 Enter the Conference IVR Service Name.
5 Define the Conference IVR Service - Global parameters. For more information, see Conference
IVR Service Properties - Global Parameters.
6 Click the Welcome tab.
The New Conference IVR Service - Welcome dialog box opens.
7 Define the system behavior when the participant enters the Conference IVR queue. For more
information, see Defining a New Conference IVR Service.
8 Click the Conference Chairperson tab.
The New Conference IVR Service - Conference Chairperson dialog box opens.
9 If required, enable the chairperson functionality and select the various voice messages and options
for the chairperson connection. For more information, see New Conference IVR Service Properties Conference Chairperson Options and Messages.
10 Click the Conference Password tab.
The New Conference IVR Service - Conference Password dialog box opens.
11 If required, enable the request for conference password before moving the participant from the
conference IVR queue to the conference and set the MCU behavior for password request for Dial-in
and Dial-out participant connections. For more information, see New Conference IVR Service
Properties - Conference Password Parameters.
12 Select the various audio messages that will be played in each case. For more information, see For
more information, see New Conference IVR Service Properties - Conference Password Parameters.
13 Click the General tab.
The New Conference IVR Service - General dialog box opens.
14 Select the messages that will be played during the conference. For more information, see
Conference IVR Service Properties - General Voice Messages.
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15 Click the Roll Call/Notifications tab.
The New Conference IVR Service - Roll Call dialog box opens.
16 Enable the Roll Call feature and assign the appropriate audio file to each message type. For more
information, see Conference IVR Service Properties - Roll Call Messages.
17 Click the Video Services tab.
The New Conference IVR Service - Video Services dialog box opens.
18 Define the Video Services parameters. For more information, see New Conference IVR Service
Properties - Video Services Parameters.
19 Click the DTMF Codes tab.
The New Conference IVR Service - DTMF Codes dialog box opens.
The default DTMF codes for the various functions that can be performed during the conference by all
participants or by the chairperson are listed. For the full list of the available DTMF codes, see New
Conference IVR Service Properties - DTMF Codes.
20 If required, modify the default DTMF codes and the permissions for various operations including
Operator Assistance options:
 *0 for individual help - the participant requested help for himself or herself. In such a case, the
participant requesting help is moved to the Operator conference for one-on-one conversation. By
default, all participants can use this code.
 00 for conference help - the conference chairperson (default) can request help for the conference.
In such a case, the operator joins the conference.
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21 Click the Operator Assistance tab.
The Operator Assistance dialog box opens.
22 Select Enable Operator Assistance to enable operator assistance when the participant requires or
requests help during the connection process to the conference or during the conference.
23 In the Operator Assistance Indication Message field, select the audio message to be played
when the participant requests or is waiting for the operator’s assistance.
If the audio file was not uploaded prior to the definition of the IVR Service or if you want to add new
audio files, click Add Message File to upload the appropriate audio file to the Collaboration Server.
24 Click OK to complete the IVR Service definition.
The new Conference IVR Service is added to the IVR Services list.
To define an Entry Queue IVR Service with Operator Assistance Options
1 In the RMX Management pane, click IVR Services (
).
2 In the IVR Services list, click New Entry Queue IVR Service (
).
The New Entry Queue IVR Service - Global dialog box opens.
3 Define the Entry Queue Service Name.
4 Define the Entry Queue IVR Service Global parameters. For more information, see Entry Queue
IVR Service Properties - Global Parameters.
5 Click the Welcome tab.
The New Entry Queue IVR Service - Welcome dialog box opens.
6 Define the system behavior when the participant enters the Entry Queue. This dialog box contains
options that are identical to those in the Conference IVR Service - Welcome Message dialog box.
7 Click the Conference ID tab.
The New Entry Queue IVR Service - Conference ID dialog box opens.
8 Select the required voice messages. For more information, see Entry Queue IVR Service Properties
- Conference ID.
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9 Click the Video Services tab.
The New Entry Queue IVR Service - Video Services dialog box opens.
10 In the Video Welcome Slide list, select the video slide that will be displayed to participants
connecting to the Entry Queue. The slide list includes the video slides that were previously uploaded
to the MCU memory.
11 Click the Operator Assistance tab.
The Operator Assistance dialog box opens.
12 Select Enable Operator Assistance to enable operator assistance when the participant requires or
requests help during the connection process.
13 In the Operator Assistance Indication Message field, select the audio message to be played
when the participant requests or is waiting for operator’s assistance.
If the audio file was not uploaded prior to the definition of the IVR Service or if you want to add new
audio files, click Add Message File to upload the appropriate audio file to the Collaboration Server.
14 Click OK to complete the Entry Queue IVR Service definition.
The new Entry Queue IVR Service is added to the IVR Services list.
To define a Conference Profile for an Operator Conference
1 In the RMX Management pane, click Conference Profiles.
2 In the Conference Profiles pane, click New Profile.
The New Profile – General dialog box opens.
3 Define the Profile name and, if required, the Profile general parameters.
For more details, see New AVC CP Profile - General Parameters.
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4 Click the Operator Conference check box.
5 Click the Advanced tab.
The New Profile – Advanced dialog box opens.
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6 Define the Profile - Advanced parameters. For more details, see New AVC CP Profile - Advanced
Parameters.
Note that when Operator Conference is selected, the Auto Terminate selection is automatically
cleared and disabled and the Operator conference cannot automatically end unless it is terminated
by the Collaboration Server User.
7 Click the Video Quality tab.
The New Profile – Video Quality dialog box opens.
8 Define the Video Quality parameters. For more details, see New AVC CP Profile - Video Quality
Parameters.
9 Click the Video Settings tab.
The New Profile - Video Settings dialog box opens.
10 Define the video display mode and layout. For more details, see New AVC CP Profile - Video
Settings Parameters.
11 Define the remaining Profile parameters. For more details, see Defining AVC CP Conferencing
Profiles.
12 Click OK to complete the Profile definition.
A new Profile is created and added to the Conference Profiles list.
Starting an Ongoing Operator Conference
To start a conference from the Conference pane:
1 In the Conferences pane, click New Conference ( ).
The New Conference – General dialog box opens.
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2 In the Profile field, select a Profile in which the Operator Conference option is selected.
Upon selection of the Operator Conference Profile, the Display Name is automatically taken from the
Collaboration Server User Login Name. This name cannot be modified.
Only one Operator conference can be created for each User Login name.
3 Define the following parameters:
New Conference – General Options
Field
Description
Duration
Define the duration of the conference in hours using the format HH:MM (default
01:00).
Notes:
• The Operator conference is automatically extended up to a maximum of 168
hours. Therefore, the default duration can be used.
• This field is displayed in all tabs.
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New Conference – General Options
Field
Description
Routing Name
The name with which ongoing conferences, Meeting Rooms, Entry Queues and
SIP Factories register with various devices on the network such as gatekeepers
and SIP servers. This name must be defined using ASCII characters.
Comma, colon and semicolon characters cannot be used in the Routing Name.
The Routing Name can be defined by the user or automatically generated by the
system if no Routing Name is entered as follows:
• If ASCII characters are entered as the Display Name, it is used also as the
Routing Name
• If a combination of Unicode and ASCII characters (or full Unicode text) is entered
as the Display Name, the ID (such as Conference ID) is used as the Routing
Name.
If the same name is already used by another conference, Meeting Room or Entry
Queue, the Collaboration Server displays an error message and requests that you
to enter a different name.
Profile
Select an operator profile from the Profile drop-down list.
ID
Enter the unique-per-MCU conference ID. If left blank, the MCU automatically
assigns a number once the conference is launched.
This ID must be communicated to conference participants to enable them to dial in
to the conference.
Conference
Password
Leave this field empty when defining an Operator conference.
Chairperson
Password
Leave this field empty when defining an Operator conference.
Reserve Resources
for Video Participants
Enter the number of video participants for which the system must reserve
resources.
Default: 0 participants.
When defining an Operator conference it is recommended to reserve resources for
at least 2 video participants (for the operator and one additional participant - who
will be moved to the Operator conference for assistance).
Note: This option is not supported with Collaboration Server 1800.
Reserve Resources
for Voice Participants
Enter the number of audio participants for which the system must reserve
resources.
Default: 0 participants.
When defining an Operator conference and the operator is expected to help voice
participants, it is recommended to reserve resources for at least 2 video participants
(for the operator and one additional participant - who will be moved to the Operator
conference for assistance).
Note: This option is not supported with Collaboration Server 1800.
Maximum Number of
Participants
Enter the maximum number of participants that can connect to an Operator
conference (you can have more than two), or leave the default selection
(Automatic).
Maximum number of participants that can connect to an Operator conference:
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New Conference – General Options
Field
Description
Enable ISDN/PSTN
Dial-in
Select this check box if you want ISDN and PSTN participants to be able to connect
directly to the Operator conference. This may be useful if participants are having
problems connecting to their conference and you want to identify the problem or
help them connect to their destination conference.
ISDN/PSTN Network
Service and Dial-in
Number
If you have enable the option for ISDN/PSTN direct dial-in to the Operator
conference, assign the ISDN/PSTN Network Service and a dial-in number to be
used by the participants, or leave these fields blank to let the system select the
default Network Service and assign the dial-in Number.
Note: The dial-in number must be unique and it cannot be used by any other
conferencing entity.
4 Click the Participants tab.
The New Conference - Participants dialog box opens.
You must define or add the Operator participant to the Operator conference.
This participant must be defined as a dial-out participant.
Define the parameters of the endpoint that will be used by the Collaboration Server User to connect
to the Operator conference and to other conference to assist participants.
For more information see Participants Tab.
5 To insert general information, select the Information tab.
The Information dialog box opens.
6 Enter the required information. For more information, see Information Tab.
7 Click OK.
The new Operator conference is added to the ongoing Conferences list with a special icon
The Operator participant is displayed in the Participants list with an Operator participant icon
and the system automatically dials out to the Operator participant.
.
,
Saving an Operator Conference to a Template
The Operator conference that is ongoing can be saved as a template.
To save an ongoing Operator conference as a template:
1 In the Conferences list, select the Operator conference you want to save as a Template.
2 Click the Save Conference to Template (
) button.
or
Right-click and select Save Conference to Template.
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The conference is saved to a template whose name is taken from the ongoing conference Display
Name (the Login name of the Collaboration Server User). The Template is displayed with the
Operator Conference icon.
Starting an Operator Conference from a Template
An ongoing Operator conference can be started from an Operator Template saved in the Conference
Templates list.
To start an ongoing Operator conference from an Operator Template:
1 In the Conference Templates list, select the Operator Template to start as an ongoing Operator
conference.
•
•
You can only start an Operator conference from a template whose name is identical to your Login
Name. For example, if your Login name is Polycom, you can only start an Operator conference from
a template whose name is Polycom.
If an ongoing Operator conference with the same name or any other conference with the same ID
is already running, you cannot start another Operator conference with the same login name.
2 Click Start Conference from Template (
).
or
Right-click and select Start Conference from Template.
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The conference is started.
The name of the ongoing conference in the Conferences list is taken from the Conference Template
Display Name.
Monitoring Operator Conferences and Participants Requiring
Assistance
Operator conferences are monitored in the same way as standard ongoing conferences.
Each Operator conference includes at least one participant - the Operator.
You can view the properties of the Operator conference by double-clicking the conference entry in the
Conferences list or by right-clicking the conference entry and selecting Conference Properties.For more
information, see Participant Level Monitoring.
Requesting Help
A participant can request help using the appropriate DTMF code from his/her touch tone telephone or the
endpoint’s DTMF input device. The participant can request Individual Assistance (default DTMF code *0) or
Conference Assistance (default DTMF code 00).
Participants in Entry Queues who failed to enter the correct destination conference ID or the conference
password will wait for operator assistance (provided that an Operator conference is active).
When requiring or requesting operator assistance, the Collaboration Server management application
displays the following:
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● The participant’s connection Status changes, reflecting the help request. For more information, see
Participants List Status Column Icons and Indications.
● The conference status changes and it is displayed with the exclamation point icon and the status
Awaiting Operator.
● The appropriate voice message is played to the relevant participants indicating that assistance will
be provided shortly.
The following icons and statuses are displayed in the Participant Status column:
Participants List Status Column Icons and Indications
Icon
Status Indication
Description
Awaiting Individual
Assistance
The participant has requested the operator’s
assistance for himself/herself.
Awaiting Conference
Assistance
The participant has requested the operator’s
assistance for the conference. Usually this means
that the operator is requested to join the conference.
When the Operator moves the participant to the Operator conference for individual assistance the
participant Status indications are cleared.
Participant Alerts List
The Participant Alerts list contains all the participants who are currently waiting for operator assistance.
Participants are automatically added to the Participants Alerts list in the following circumstances:
● The participant fails to connect to the conference by entering the wrong conference ID or conference
password and waits for the operator’s assistance
● The participant requests Operator’s Assistance during the ongoing conference
This list is used as reference only. Participants can be assisted and moved to the Operator conference or
the destination conference only from the Participants list of the Entry Queues or ongoing conference where
they are awaiting assistance.
The participants are automatically removed from the Participant Alerts list when moved to any conference
(including the Operator conference).
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Audible Alarms
In addition to the visual cues used to detect events occurring on the Collaboration Server, an audible alarm
can be activated and played when participants request Operator Assistance.
Using Audible Alarms
The Audible Alarm functionality for Operator Assistance requests is enabled for each MCU in either the
Collaboration Server Web Client or RMX Manager.
The Audible Alarm played when Operator Assistance is requested is enabled and selected in the Setup >
Audible Alarm > User Customization. When the Audible Alarm is activated, the *.wav file selected in the
User Customization is played, and it is repeated according to the number of repetitions defined in the User
Customization.
If more than one Collaboration Server is monitored in the RMX Manager, the Audible Alarm must be enabled
separately for each Collaboration Server installed in the site/configuration. A different *.wav file can be
selected for each MCU.
When multiple Audible Alarms are activated in different conferences or by multiple MCUs, the Audible
Alarms are synchronized and played one after the other. It is important to note that when Stop Repeating
Alarm is selected from the toolbar from the Collaboration Server Web Client or RMX Manager, all activated
Audible Alarms are immediately halted.
For more details on Audible alarms and their configuration, see Audible Alarms.
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Conference Templates enable administrators and operators to create, save, schedule and activate identical
conferences.
A Conference Template:
● Saves the conference Profile.
● Saves all participant parameters including their Personal Layout and Video Forcing settings.
● Simplifies the setting up Telepresence conferences where precise participant layout and video forcing
settings are crucial.
Trying to start a Conference Template that exceeds the allowed maximum number of participants will result
in participants being disconnected due to resource deficiency.
If the Profile assigned to a conference is deleted while the conference is ongoing the conference cannot be
saved as a template.
A Profile assigned to a Conference Template cannot be deleted. The system does not permit such a
deletion.
Profile parameters are not embedded in the Conference Template, and are taken from the Profile when the
Conference Template becomes an ongoing conference. Therefore, any changes to the Profile parameters
between the time the Conference Template was created and the time that it is activated (and becomes an
ongoing conference) will be applied to the conference.
Only defined participants can be saved to the Conference Template. Before saving a conference to a
template ensure that all undefined participants have disconnected.
Undefined participants are not saved in Conference Templates.
Participant properties are embedded in the Conference Template and therefore, if the participant properties
are modified in the Address Book after the Conference Template has been created they are not applied to
the participant whether the Template becomes an ongoing conference or not.
The Conference Template display name, routing name or ID can be the same as an Ongoing Conference,
reservation, Meeting Room or Entry Queue as it is not active. However, an ongoing conference cannot be
launched from the Conference Template if an ongoing conference, Meeting Room or Entry Queue already
has the same name or ID. Therefore, it is recommended to modify the template ID, display name, routing
name to be unique.
A Reservation that has become an ongoing conference can be saved as Conference Template.
SIP Factories and Entry Queues cannot be saved as Conference Templates.
The conference specified in the Conference Template can be designated as a Permanent Conference. For
more information see Permanent Conference.
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
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Using Conference Templates
The Conference Templates list is initially displayed as a closed tab in the Collaboration Server Web Client
main window. The number of saved Conference Templates is indicated on the tab.
Clicking the tab opens the Conference Templates list.
The Conference Templates are listed by Conference Template Display Name and ID and can be sorted by
either field. The list can be customized by re-sizing the pane, adjusting the column widths or changing the
order of the column headings.
For more information see Customizing the Main Screen.
Clicking the anchor pin ( ) button hides the Conference Templates list as a closed tab.
Toolbar Buttons
The Conference Template toolbar includes the following buttons:
Conference Templates – Toolbar Buttons
Button
Description
New Conference Template
Creates a new Conference Template.
Delete Conference Template
Deletes the Conference Template(s) that are selected in the list.
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Conference Templates – Toolbar Buttons
Button
Start Conference from Template
Schedule Reservation from Template
Description
Starts an ongoing conference from the Conference Template that has
an identical name, ID parameters and participants as the template.
Creates a conference Reservation from the Conference Template
with the same name, ID, parameters and participants as the
Template.
Opens the Scheduler dialog box enabling you to modify the fields
required to create a single or recurring Reservation based on the
template. For more information see Scheduling Reservations.
The Conferences List toolbar includes the following button:
Conferences List – Toolbar Button
Button
Save Conference to Template
Description
Saves the selected ongoing conference as a Conference Template.
Creating a New Conference Template
There are two methods to create a Conference Template:
● From scratch - defining the conference parameters and participants
● Saving an ongoing conference as Template
Creating a new Conference Template from Scratch
To create a new Conference Template:
1 In the Collaboration Server main screen, click the Conference Templates pane.
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2 Click the New Conference Template (
) button.
The New Conference Template - General dialog box opens.
The fields of the New Template – General dialog box are identical to those of the New Conference
– General dialog box. For more information, see General Tab.
3 Modify the fields of the General dialog box.
Note: Unique dial-in number
A unique dial-in number must be assigned to each conferencing entity. However, Conference
Templates can be assigned dial-in numbers that are already assigned to other conferencing entities,
but when the template is used to start an ongoing conference or schedule a reservation, it will not start
if another ongoing conference, Meeting Room, or Entry Queue or Gateway Profile is using this
number.
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4 Click the Participants tab.
The New Template – Participants dialog box opens.
Specify the Display Name or Duration.
To have a permanent conference, select the Permanent Conference check box.
The fields of the New Template – Participants dialog box are the same as those of the New
Conference – Participant dialog box.
5 You can add participants to the template from the Address Book as desired.
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6 Click the New button.
The New Participant – General dialog box opens.
The New Template – Participant dialog box remains open in the background.
For a full description of the General tab fields see Adding a New participant to the Address Book
Directly.
7 Modify the fields of the General dialog box.
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8 Click the Advanced tab.
The New Participant – Advanced dialog box opens.
9 Modify the fields of the Advanced dialog box.
10 Click the Media Sources tab.
The Media Sources dialog box opens.
The Media Sources dialog box enables you to set up and save, Display Name, Duration and
Personal Layout and Video Forcing settings for each participant. This is especially important when
setting up Telepresence conferences.
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11 Specify the Display Name and Duration.
To have a permanent conference, select the Permanent Conference check box.
12 Modify the Personal Layout and Video Forcing settings for the participant, and select Override
layout from Profile if needed.
For Configuring Personal Layout and Video Forcing functions, see configurations on Video Settings
tab in Defining AVC-Based Conference Profiles.
13 To add any optional information, click the Information tab.
The New Participant – Information dialog box opens.
For more information, see Information Tab.
14 Click the OK button.
The participant you have defined is added to the Participants List.
The New Participant dialog box closes and you are returned to the New Template – Participant
dialog box (which has remained open since step 6).
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15 To add any optional information, in the New Conference Template dialog box, click the Information
tab.
The New Conference Template – Information dialog box opens.
For more information, see Information Tab.
16 Click the OK button.
The New Conference Template is created and its name is added to the Conference Templates list.
Saving an Ongoing or AVC-CP Operator Conference as a Template
Any ongoing or AVC-based CP Operator Conference can be saved as a template.
To save an ongoing or AVC-based CP Operator Conference as a template:
1 In the Conferences List, select the conference or Operator Conference to be saved as a Template.
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2 Click the Save Conference to Template (
) button.
or
Right-click and select Save Conference to Template.
The conference is saved to a template whose name is taken from the ongoing conference Display
Name (the Login name of the Collaboration Server User). The Template is displayed with the
Operator Conference icon.
Starting an Ongoing Conference From a Template
Note: Message Overlay text
Conference Templates saved from an ongoing conference does not include Message Overlay text
messages.
An ongoing conference can be started from any Template saved in the Conference Templates list. In
SVC-based templates, only defined dial-in participants may be part of the conference.
To start an ongoing conference from a Template:
1 In the Conference Templates list, select the Template you want to start as an ongoing conference.
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2 Click the Start Conference from Template (
) button.
or
Right-click and select Start Conference from Template.
The conference is started.
Note: Dial-in number already assigned to ongoing conference
If a Conference Template is assigned a dial-in number that is already assigned to an ongoing
conference, Meeting Room, or Entry Queue or Gateway Profile, when the template is used to start an
ongoing conference or schedule a reservation it will not start. However, the same number can be
assigned to several conference templates provided they are not used to start an ongoing conference
at the same time. If a dial in number conflict occurs prior to the conference’s start time, an alert is
displayed: ISDN dial-in number is already assigned to another conferencing
entity and the conference cannot start.
The name of the ongoing conference in the Conferences list is taken from the Conference Template
Display Name.
Participants that are connected to other ongoing conferences when the template becomes an
ongoing conference are not connected.
Note: Duplicate Display Name, Routing Name, ID
If an ongoing conference, Meeting Room or Entry Queue with the same Display Name, Routing Name
or ID already exists in the system, the conference will not be started.
Starting an Operator Conference from a Template (AVC Conferencing)
An ongoing Operator conference can be started from an Operator Template saved in the Conference
Templates list.
To start an ongoing Operator conference from an Operator Template:
1 In the Conference Templates list, select the Operator Template to start as an ongoing Operator
conference.
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Note: Login Name
You can only start an Operator conference from a template whose name is identical to your Login
Name. For example, if your Login name is Polycom, you can only start an Operator conference from a
template whose name is Polycom.
If an ongoing Operator conference with the same name or any other conference with the same ID is
already running, you cannot start another Operator conference with the same login name.
2 Click the Start Conference from Template (
) button.
or
Right-click and select Start Conference from Template.
The conference is started.
The name of the ongoing conference in the Conferences list is taken from the Conference Template
Display Name.
Scheduling a Reservation From a Conference Template
A Conference Template can be used to schedule a single or recurring Reservation.
To schedule a Reservation from a Conference Template:
1 In the Conference Templates list, select the Conference Template you want to schedule as a
Reservation.
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2 Click the Schedule Reservation from Template (
) button.
or
Right-click and select Schedule Reservation from Template.
The Reservation Properties dialog box is displayed.
The Display Name of the Reservation is taken from the Conference Template Display Name.
For a full description of the Reservation Properties fields see Creating a New Reservation.
3 Modify the fields of the Reservation Properties.
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4 Click the OK button.
A Reservation is created based on the Conference Template. The Reservation can be viewed and
modified along with all other Reservations using the Reservations - Calendar View and Reservations
List.
If you create a recurring reservation all occurrences have the same ID. A recurring Reservation is
assigned the same ISDN/PSTN dial-in number for all recurrences.
If a dial-in number conflict occurs prior to the conference’s start time, an alert is displayed: ISDN
dial-in number is already assigned to another conferencing entity and the
conference cannot start.
The series number (_0000n) of each reservation is appended to its Display Name.
Example:
Conference Template name: Sales
Display Name for single scheduled occurrence: Sales
If 3 recurrences of the reservation are created:
Display Name for occurrence 1: Sales_00001
Display Name for occurrence 2: Sales_00002
Display Name for occurrence 3: Sales_00003
Deleting a Conference Template
One or several Conference Templates can be deleted at a time.
To delete Conference Templates:
1 In the Conference Templates list, select the Template(s) you want to delete.
2 Click the Delete Conference Template (
) button.
or
Right-click and select Delete Conference Template.
A confirmation dialog box is displayed.
3 Click the OK button to delete the Conference Template(s).
Exporting and Importing Conference Templates
Conference Templates can be exported from one MCU and imported to multiple MCUs in your environment.
Additionally, you can export Conference Templates and their associated Conference Profiles
simultaneously. Using this option can save configuration time and ensures that identical settings are used
for conferences running on different MCUs. This is especially important in environments using cascading
conferences that are running on different MCUs.
● Administrators can export and import Conference Templates. Operators are only allowed to export
Conference Templates.
● You can select a single, multiple or all Conference Templates to be exported.
● Both Conference Templates and their associated Conference Profiles can be exported and imported
simultaneously when enabling the Export includes conference profiles or Import includes conference
profiles options.
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● Exporting and importing Conference Templates only can be used when you want to export and import
individual Conference Templates without their associated Conference Profiles. This option enables
you to import Conference Templates when Conference Profiles already exist on an MCU.
Exporting Conference Templates
Conference Templates are exported to a single XML file that can be used to import the Conference
Templates on multiple MCUs.
Using the Export Conference Templates option, you can:
● Export all Conference Templates from an MCU
● Export selected Conference Templates
Exporting All Conference Templates from an MCU
To export all Conference Templates from an MCU:
1 In the Collaboration Server Web Client main window, click the Conference Templates tab.
The Conference Templates list pane is displayed.
2 Click the Export Conference Templates
button, or right-click the Conference Templates list,
and then select Export Conference Templates.
The Conference Templates - Export dialog box is displayed.
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3 In the Export Path field, type the path name to the location where you want to save the exported file
or click Browse to select the desired path.
4 If you wish to clear the Export includes conference profiles check box when you only want to
export Conference Templates.
When this check box is cleared, the Conference Templates - Export dialog box is displayed without
the Profiles file name field.
5 In the Templates file name field, type the file name prefix. The file name suffix
(_confTemplates.xml) is predefined by the system. For example, if you type Templates01, the
exported file name is defined as Templates01_confTemplates.xml.
The system automatically defines the Profiles file name field with the same file name prefix as the
Templates file name field. For example, if you type Templates01 in the Templates file name field, the
exported profiles file name is defined as Templates01_confProfiles.xml.
6 Click OK to export the Conference Templates and Conference Profiles to a file.
Exporting Selected Conference Templates
You can export a single Conference Template or multiple Conference Templates to other MCUs in your
environment.
To export selected Conference Templates:
1 In the Conference Templates list, select the templates you want to export.
2 Right-click the Conference Templates to be exported, and then click Export Selected Conference
Templates.
The Conference Templates - Export dialog box is displayed.
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3 In the Export Path field, type the path name to the location where you want to save the exported file
or click Browse to select the desired path.
4 To export Conference Templates, clear the Export includes conference profiles check box.
When this check box is cleared, the Conference Templates - Export dialog box is displayed without
the Profiles file name field.
5 In the Templates file name field, type the file name prefix. The file name suffix
(_confTemplates.xml) is predefined by the system. For example, if you type, Templates01, the
exported file name is defined as Templates01_confTemplates.xml.
The system automatically defines the Profiles file name field with the same file name prefix as the
Templates file name field. For example, if you type Templates01 in the Templates file name field, the
exported profiles file name is defined as Templates01_confProfiles.xml.
6 Click OK to export the Conference Templates and Conference Profiles to a file.
Importing Conference Templates
You can import Conference Templates and Conference Profiles from one MCU to multiple MCUs in your
environment.
To import Conference Templates:
1 In the Collaboration Server Web Client main window, click the Conference Templates tab.
The Conference Templates are displayed.
2 Click the Import Conference Templates
button, or right-click the Conference Templates pane,
and then click Import Conference Templates.
The Conference Templates - Import dialog box is displayed.
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3 To import Conference Templates, clear the Import includes conference profiles check box.
When this check box is cleared, the Conference Templates - Import dialog box is displayed without
the Profiles file name field.
4 In the Import Path field, click Browse to navigate to the path and file name of the Conference
Templates you want to import.
When clicking the exported templates file you want to import, the system automatically displays the
appropriate files in the Templates file name field and the Profiles file name field (when the Import
includes conference profiles check box is selected).
5 Click OK to import the Conference Templates and their associated Conference Profiles, if selected.
Conference Templates are not imported when:
 A Conference Template already exists
 An associated Conference Profile is not defined in the Conference Profiles list
When one or more Conference Templates are not imported, a Message Alert window is displayed with
the templates that were not imported.
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6 Click Cancel to exit the Message Alerts window.
The imported Conference Templates are added to the Conference Templates list. When the Import
includes conference profiles check box is selected, the imported Conference Profiles are added to
the Conference Profiles list.
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Start a Conference
Note: Virtual Platform Setup
When using RealPresence Platform, the conferencing parameters should be defined in the
RealPresence DMA component, and not directly in the Collaboration Server Virtual Edition.
There are several ways to start a conference:
● Clicking the New Conference button in the Conferences pane. For more information, see Start an
AVC CP Conference from the Conferences Pane and Starting a New SVC Conference.
● Dialing in to a Meeting Room.
A Meeting Room is a conference that is saved on the MCU. It remains in passive mode until it is
activated by the first participant, or the meeting organizer, dialing in.
For more information, see Meeting Rooms.
● Dialing in to an Ad Hoc Entry Queue which is used as the access point to the MCU. This option is
valid to AVC participants dialing into an AVC-based or a Mixed CP and SVC Entry Queue.
For more information, see Entry Queues, Ad Hoc Conferences and SIP Factories.
● Start a Reservation (AVC CP):
 If the Start Time of the Reservation is past due the conference becomes ongoing immediately.
 If the Start Time of the Reservation is in the future the conference becomes ongoing, at the
specified time on the specified date.
Changes made to this information once the conference is running are not saved to the CDR.
● Start any Conference Template saved in the Conference Templates list.
For more information, see Starting an Ongoing Conference From a Template.
● Clicking on the link included in a Microsoft Outlook Polycom Meeting Invitation or by manually dialing
the numbers displayed in the invitation using the endpoint's numeric input device.
This option is valid to AVC participants only.
Attendees that have received an invitation to a Polycom Meeting through the Polycom Conferencing
Add-in for Microsoft Outlook can start a conference by being the first invited attendee to click on a link
in the Meeting Invitation displayed on his/her workstation or calendaring enabled endpoint, or
manually dial in to the meeting using the Polycom Conference information included in the Meeting
Invitation.
For more information see Polycom Conferencing for Microsoft Outlook®.
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Start an AVC CP Conference from the Conferences Pane
To start an AVC CP conference from the Conference pane:
1 In the Conferences pane, click New Conference ( ).
The New Conference – General dialog box opens.
2 Set general information, such as meeting duration, meeting profile, and passwords. For more
information, see General Tab.
3 Click the Participants tab to add participants from the Participants Address Book or to define
participants (mainly dial-out participants). For more information, see Participants Tab.
4 If all participants are undefined, dial-in, and no additional information is required for the new
conference, click OK.
The system displays the conference default Name, Duration and the default Profile, which contains
the conference parameters and media settings.
The system automatically allocates the conference ID when the conference starts.
In most cases, the default conference ID can be used and you can just click OK to launch the
conference. If required, you can enter a conference ID before clicking OK to launch the conference.
If you are the meeting chairperson or organizer using the Collaboration Server Web Client to start
your own meeting, you need to communicate the default conference ID (or the one you created) to
the other conference participants so that they can dial in.
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General Tab
You can define parameters such as your display name, conference profile, conference password, and
duration in the General tab.
The following table describes the information displayed in the General Tab.
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New Conference - General tab parameters
Field
Description
Display Name
The Display Name is the conferencing entity name in native language character sets to
be displayed in the RP Collaboration Server Web Client.
In conferences, Meeting Rooms, Entry Queues and SIP factories the system
automatically generates an ASCII name for the Display Name field that can be
modified using Unicode encoding.
• English text uses ASCII encoding and can contain the most characters (length varies
according to the field).
• European and Latin text length is approximately half the length of the maximum.
• Asian text length is approximately one third of the length of the maximum.
The maximum length of text fields also varies according to the mixture of character
sets (Unicode and ASCII).
Maximum field length in ASCII is 80 characters.
If the same name is already used by another conference, Meeting Room or Entry
Queue, the Collaboration Server displays an error message requesting you to enter a
different name.
Note: This field is displayed in all tabs.
Duration
Define the duration of the conference in hours using the format HH:MM (default 01:00).
Note: This field is displayed in all tabs.
Permanent Conference
Click this check box to make this conference a Permanent Conference: an ongoing
conference with no pre-determined End Time, continuing until it is terminated by an
administrator, operator or chairperson. For more information see Audio Algorithm
Support.
Note: This field is displayed in all tabs.
Routing Name
Routing Name is the name with which ongoing conferences, Meeting Rooms, Entry
Queues and SIP Factories register with various devices on the network such as
gatekeepers and SIP servers. This name must be defined using ASCII characters.
Comma, colon and semicolon characters cannot be used in the Routing Name.
The Routing Name can be defined by the user or automatically generated by the
system if no Routing Name is entered as follows:
• If ASCII characters are entered as the Display Name, it is used also as the Routing
Name
• If a combination of Unicode and ASCII characters (or full Unicode text) is entered as
the Display Name, the ID (such as Conference ID) is used as the Routing Name.
If the same name is already used by another conference, Meeting Room or Entry
Queue, the Collaboration Server displays an error message and requests that you
enter a different name.
Profile
The system displays the name of the default Conference Profile. Select the required
Profile from the list.
The Conference Profile includes the conferencing mode, conference line rate, media
settings and general settings.
For a detailed description of Conference Profiles, see Using Conference Profiles.
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New Conference - General tab parameters
Field
Description
ID
Enter the unique-per-MCU conference ID. If left blank, the MCU automatically assigns
a number once the conference is launched.
This ID must be communicated to conference participants to enable them to dial in to
the conference.
Note: If setting the Conference ID to the digits that are used for MCU prefix in
Gatekeeper (for example gatekeeper prefix is set to 10 and the conference ID is 1001),
the system will not be able to dial to the destination conference as the prefix digits are
truncated from the conference ID, preventing the system from locating it.
Note: If SIP Factories are used do not use the number 7001 as an ID. 7001 is the
default SIP Factory ID.
Conference Password
Enter a password to be used by participants to access the conference. If left blank, no
password is assigned to the conference.
This password is valid only in conferences that are configured to prompt for a
conference password.
This field is numeric and has a default length of 4 characters. The administrator can
modify it in the Setup >System Configuration settings. For more information, see
System Configuration Flags.
The Collaboration Server can be configured to automatically generate conference (and
chairperson) passwords when these fields are left blank. For more information, see
Automatic Password Generation Flags.
Chairperson Password
Enter a password to be used by the Collaboration Server to identify the Chairperson
and grant him/her additional privileges. If left blank, no chairperson password is
assigned to the conference. This password is valid only in conferences that are
configured to prompt for a chairperson password.
This field is numeric and has a default length of 4 characters. The administrator can
modify it in the Setup > System Configuration settings. For more information, see
System Configuration Flags.
The RealPresence Collaboration Server can be configured to automatically generate
chairperson (and conference) passwords when these fields are left blank. For more
information, see Automatic Password Generation Flags.
Reserve Resources for
Video Participants
Enter the number of video participants for which the system must reserve resources.
Default: 0 participants.
Maximum participants:
• MPMRx-D / RMX1800: 100
• MPMRx-S: 30
Reserve Resources for
Audio Participants
Enter the number of audio participants for which the system must reserve resources.
Default: 0 participants.
Maximum participants:
• MPMRx-D / RMX1800: 300
• MPMRx-S: 90
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New Conference - General tab parameters
Field
Description
Maximum Number of
Participants
Indicate the total number of participants that can be connected to the conference. The
automatic setting indicates that the maximum number of participants that can be
connected to the conference is determined according to resource availability.
Note: If a number is specified, it should be large enough to accommodate the
participants specified in the Reserve Resources for Video/Voice Participants fields.
Enable ISDN/PSTN
Dial-in
Select this check box if you want ISDN and PSTN participants to be able to connect
directly to the conference.
ISDN/PSTN Network
Service
The default Network Service is automatically selected. A different ISDN/PSTN Network
Service can be selected from the Network Services list.
Dial-in Number (1)
Leave this field blank to let the system automatically assign a number from the dial-in
range defined for the selected ISDN/PSTN Network Service. To manually define a
dial-in number, enter a unique number from the dial-in number range defined for the
selected Network Service. This number cannot be assigned to another
Conference/Reservation/Meeting Room/Gateway Profile
Dial-in Number (2)
By default, the second dial-in number is not defined. To define a second dial-in number,
enter a required number from the dial-in number range defined for the selected
Network Service.
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Participants Tab
The Participants tab is used to add participants to the conference from the Address Book or by defining
them. Defined dial-out participants are connected to the conference automatically when the conference is
launched, unless the Dial Out Manually option is selected.
Note:
This procedure is optional.
When defining a new conference, the Participants List is empty.
The following table describes the information displayed in the Participants List and the operations that can
be performed.
New Conference – Participants Tab
Column / Button
Description
Participants List
Name
A Unicode field that displays the participant’s name and an icon representing the
endpoint type: Voice or Video.
IP Address/Phone
Indicates the IP address or phone number of the participant’s endpoint.
• For dial-out connection, displays the IP address or phone number of the
endpoint called by the RealPresence Collaboration Server.
• For dial-in connection, displays the participant’s IP address or phone number
used to identify and route the participant to the appropriate conference.
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Column / Button
Description
Alias Name/SIP
Address (IP Only)
Displays the alias name of an H.323 endpoint or the SIP URL.
Network
The network communication protocol used by the endpoint to connect to the
conference: IP (H.323 or SIP) or ISDN/PSTN.
Dialing Direction
Select the dialing direction:
• Dial-in – The participant dials in to the conference. This field applies to IP
participants only.
• Dial-out – The MCU dials out to the participant.
Note: Dial-out is forced when defining an ISDN/PSTN participant.
Encryption
Displays whether the endpoint uses encryption for its media.
The default setting is Auto, indicating that the endpoint must connect according to
the conference encryption setting.
Lecturer
This option is used to activate the Lecture Mode. Select the participant you want to
designate as Lecturer from the list of conference participants.
The Lecturer can be selected once the conference is running and participants are
connected.
Dial Out Manually
Select this option to designate a Collaboration Server User-controlled dial-out
conference connection. When checked, the user must connect each of the dial-out
participants who remain on standby until they are connected to the conference.This
option is disabled in SVC conferences.
Buttons
New
Click to define a new participant.
For more information, see Polycom® RealPresence Collaboration Server (RMX)
1800/2000/4000 Administrator’s Guide, Adding a New participant to the Address
Book Directly.
Remove
Click to remove the selected participant from the conference.
Add from Address
Book
Click to add a participant from the Address Book to the conference.
Participants can be added to the conference in the following methods:
● Defining a new participant during the definition of the conference (clicking the New button).
● Adding pre-defined participants from the Address Book by either selecting the participants from the
list or dragging and dropping the participants from the Address Book to the Participants list.
● Dial-in participants can connect to the conference after it was started (without using the New
Conference - Participants dialog box).
● Once the conference has started, participants can be added to a conference directly from the
Participants Address Book without having to use the New Conference – Participants tab.
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To add participants from the Address Book:
1 In the Participants List, click the Add from Address Book button to open the Participants Address
Book.
The All Participants list opens.
2 In the Participants Address Book, select the participants that you want to add to the conference and
click the Add button.
Standard Windows multiple selection techniques can be used in this procedure.
3 The selected participants are assigned to the conference and appear in the Participant List.
4 Select additional Participants or click the Close button to return to the Participants tab.
Information Tab
In the Info fields, you can add general information about the conference, such as contact person name,
company name, billing code, etc.
This information is written to the Call Detail Record (CDR) when the conference is launched.
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Changes made to this information once the conference is running are not saved to the CDR.
Note:
This procedure is optional. The information entered into these fields does not affect the conference.
New Conference – Information Tab
Field
Description
Info1, 2, 3
There are three information fields that allow you to enter general information for
the conference such as company name, contact person etc.
Unicode can be used in these fields.
The maximum length of each field is 80 characters.
Billing
Enter the conference billing code if applicable.
If no participants were defined for the conference or as long as no participants are connected, the indication
Empty and a warning icon ( ) appear in the Status column in the Conferences pane.
The status changes when participants connect to the conference.
If no participant connects within the time specified in the Conference Profiles > Auto Terminate > Before
First Joins field, the conference is automatically terminated by the system.
Media Sources Tab
You can define a new layout, and override the original layout that defined in the Conference Profile.
Furthermore you can define any participant into a specific window in the new layout. For more information,
see Video Forcing (AVC-Based CP and Mixed CP and SVC Conferences).
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Starting a Mixed CP and SVC or SVC Only Conference from the
Conferences Pane
Starting a New SVC Conference
Note: SVC Conferencing Requirements
• A license is required for SVC conferencing.
• In mixed AVC-SVC conferences, participants with both SVC and AVC endpoints can participate in
the same conference.
• In non-virtual MCUs, during mixed SVC & CP conferences, PSTN (Audio Only) calls are also
supported.
To start a SVC conference from the Conference pane:
1 In the Conferences pane, click the New Conference ( ) button.
The New Conference – General dialog box opens.
2 Select a Profile set to SVC or Mixed CP and SVC Conferencing Mode.
3 Set parameters in the General, Information, Media Sources, and Participants tabs.
For more information on parameters, see General Tab, Participants Tab, Information Tab, and Media
Sources Tab.
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Scheduling an AVC-based Reservation
AVC-based Reservations are started by selecting an AVC-based Profiles.
To start a conference from the Reservation Calendar:
1 In the RMX Management pane, click the Reservations entry (
displayed.
2 Click the New Reservation (
). The Reservation Calendar is
) button.
The New Reservation – General tab dialog box is displayed.
This dialog box is identical to the AVC CP New Conference - General dialog box. For a field
description, see General Tab.
3 Optional. Select the Enable ISDN/PSTN Dial-in check box if you want ISDN and PSTN participants
to be able to connect directly to the conference.
4 If Enable ISDN/PSTN Dial-in option is selected, either enter a dial-in number, or leave the Dial-in
Number field blank to let the system automatically assign a number from the dial-in range defined
for the selected ISDN/PSTN Network Service.
5 Click the OK button.
A confirmation box is displayed stating that the Reservation time is past due and that the conference
will become ongoing.
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6 Click OK.
The conference is started. If applicable, an ISDN/PSTN dial-in number was assigned to the
conference either automatically or manually, this number can be viewed in the Conferences pane.
For more information about Reservations, see Scheduling Reservations.
Starting an Ongoing Conference From a Template
An ongoing conference can be started from any Conference Template saved in the Conference Templates
list.
Note: SVC Conferencing Exception
SVC-based Conference Templates cannot be saved with dial-out participants, only dial-in SIP
participants may be defined.
To start an ongoing conference from a template:
1 In the Conference templates list, select the template you want to start as an ongoing conference.
2 Click the Start Conference from Template (
from Template.
) button, or right-click and select Start Conference
The conference is started.
Note: ISDN/PSTN Numbers Assignment
If a CP Conference Template is assigned an ISDN/PSTN dial-in number already assigned to an
ongoing conference, Meeting Room, Entry Queue or Gateway Profile, the template cannot be used to
start an ongoing conference or schedule a reservation. However, the same number can be assigned
to several conference templates provided they are not used to start an ongoing conference
simultaneously. If a dial in number conflict occurs prior to conference’s start time, an alert appears,
indicating ISDN dial-in number is already assigned to another conferencing entity.
The name of the ongoing conference in the Conferences list is taken from the conference template
Display Name.
Participants that are connected to other ongoing conferences when the template becomes an
ongoing conference are not connected.
Note: Name Differing Requirement
Display Name, Routing Name and ID must be unique for ongoing conferences, Meeting Rooms or
Entry Queues.
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For detailed description of Conference Templates, see Conference Templates.
Starting a Meeting from Microsoft Outlook Using Polycom Add-in
Note: Feature Applicability
This feature is applicable for CP Only Conferencing.
Polycom Conferencing for Microsoft Outlook is an add-in that enables users to easily organize and invite
attendees to video enabled meetings via Microsoft Outlook®. For more information see Polycom
Conferencing for Microsoft Outlook®.
Using Polycom Conferencing add-in to Microsoft Outlook, an audio meeting can be created as any other
Outlook meeting. For more information, see Appendix H - Integration Into Microsoft Environments.
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Operations
Conference Operations
The Collaboration Server user can Copy and Paste conferences. When using the Collaboration Server Web
Client, conferences can be copied and pasted on the same MCU. However, when using the RMX Manager,
with its ability to manage multiple MCUs, conferences can be copied and pasted between different MCUs.
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
Copy Conference
The Copy command copies all the conference’s properties including connected participants and makes
these properties available for pasting and starting a new conference. The copied conference remains active
until it terminates or is deleted.
To copy a conference:
» In the Conferences List pane, right-click the conference you want to copy, and select Copy
Conference.
Paste Conference
The Paste Conference command starts the new conference on the same MCU or on a different MCU.
To paste a conference:
» Right-click in the Conferences List pane, and select Paste Conference.
or
If you are using the RMX Manager, and you want to paste the conference to a different MCU:
a In the MCUs list pane, click the MCU in which to create the conference.
b In the Conferences List pane, right-click, and select Paste Conference.
The conference is pasted to the MCU with its Display Name assigned by the system.
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Paste Conference As
The Paste Conference As command allows the system user to create a new conference using the copied
conference properties as a template. It automatically opens the Conference Properties dialog box allowing
the user to modify information in the General, Participants and Information tabs to create the new
conference. When the OK button in the Conference Properties dialog box conference Properties dialog
box is clicked, the new conference is started.
To paste a conference as a new conference:
1 Right-click in the Conferences List pane, and select Paste Conference As.
or
If you are using the RMX Manager and you want to paste the conference to a different MCU:
a In the MCUs list pane, click the MCU in which to create the conference.
b In the Conferences List pane, right-click, and select Paste Conference As.
The Conference Properties dialog box is displayed.
2 Modify the conference information as required.
3 Click OK to paste and start the new conference.
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Participant Operations
Operational Buttons
Various operations can be performed during an ongoing conference, affecting only the selected participants
in the conference. These operations enable you to modify and control the connections and statuses of
participants in ongoing conferences, as described in the following table.
Participant Level Operations
Menu Option
Button
Description
New Participant
Define a new participant.
SVC dial-in participants can be added to SVC Only and Mixed CP and SVC
conferences. SVC dial-out participants cannot be added to any conference.
Add Participant
From Address Book
Open the Address Book to select the participant for the conference.
SVC dial-in participants can be added to SVC Only and Mixed CP and SVC
conferences. SVC dial-out participants cannot be added to any conference.
Connect Participant
Connect a disconnected defined dial-out participant to the conference.
Disconnect
Participant
Disconnect the participant from the conference.
Delete Participant
Delete the selected participants from the conference.
Mute Audio
Mute the audio transmission from the participant to the conference.
The Audio Muted indicator appears in the Participants List and the Unmute
Audio button (
) becomes active.
Unmute Audio
Resume the participant’s audio transmission to the conference.
The Mute Audio button (
) becomes active.
Suspend Video
Suspend the video transmission from the participant to the conference. The
suppressed participant’s video is not transmitted to the conference but the
participant still receives conference video.
The Suspend Video indicator appears in the Participants List and the
Resume Video button (
) becomes active.
Resume Video
Resume the participant’s video transmission to the conference.
The Suspend Video button becomes active (
).
Block Audio
Block the audio transmission from the conference to the participant. When
blocked, the participant can still be heard by the conference.
The Audio Blocked indicator appears in the Participants List and the
Unblock Audio button (
) becomes active.
Unblock Audio
Resume the audio transmission from the conference to the participant.
The Block Audio button (
) becomes active.
Change to
Chairperson
Define the selected participant as the conference leader/chairperson.
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Participant Level Operations
Menu Option
Button
Description
Change to Regular
Participant
Define the chairperson as a regular participant without chairperson privileges.
Change To Content
Token Owner
Initiate Content Broadcast Control to prevent the accidental interruption or
termination of H.239 Content that is being shared by this participant.
Cancel Content
Token Owner
Cancel Content Broadcast Control.
Add Participant to
Address Book
Add selected participant’s details to the Participant Address Book.
Move to Conference
(AVC Only)
Move an AVC participant to another ongoing AVC CP conference. The
destination conference is selected from a displayed list.
View Participant
Sent Video
(AVC Only)
Preview the video sent from the AVC participant to the conference.
View Participant
Received Video
(AVC Only)
Preview the video sent from the conference to the AVC participant.
Copy Participant
Copy the all participant’s parameters in preparation for Pasting into another
conference or back into the current conference.
Cut Participant
Copy the all participant’s parameters and delete the participant from the
current conference. The participant can be pasted into another conference or
back into the current conference.
Paste Participant As
Paste the participant into the selected conference as a new participant with
parameters modified via the Address Book Participant - Properties dialog
box.
Abort H.239 Session
Withdraw the Content Token from the participant back to the MCU for
re-assignment.
Connect to Website
(AVC Only)
Connect directly to the internal website of the participant’s endpoint to perform
administrative, configuration and troubleshooting activities.
AGC
Enable AGC for the participant with weak audio signal during ongoing
conferences.
Notes:
• To implement AGC for the participant audio, the ENABLE_AGC System
Flag in system.cfg must be set to YES. The flag default value is NO.
• Enabling AGC may result in amplification of background noise.
(Auto Gain Control)
(AVC Only)
Participant
Properties
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Copy, Cut and Paste Participant
The Collaboration Server user can Copy, Cut and Paste participants between different conferences running
on the MCU, including his/her current conference. These functions, when used via the RMX Manager, with
its ability to manage multiple MCUs and participants, allow the MCU user to Copy, Cut and Paste
participants between conferences running on different MCUs.
Copy Participant
The Copy command copies all the participant’s properties and makes them available for pasting. The
participant remains connected to his/her current conference.
To copy a participant:
» In the Participants List pane, right-click the participant you want to copy, and select Copy
Participant.
Cut Participant
The Cut command copies all the participant’s properties and makes them available for pasting. The
participant is deleted from his/her current conference.
To cut a participant:
» In the Participants List pane, right-click the participant you want to cut, and select Cut Participant.
Paste Participant
The Paste command connects the copied or cut participant to the selected conference.
If the participant was copied, he/she should be deleted from the conference he/she was copied from, unless
it is required that the participant is connected to two (or more) conferences. (There are endpoints that permit
a participant to be connected to multiple conferences).
To paste a participant:
1 In the Conferences List pane, click the conference into which you want to paste the copied/cut
participant.
2 Right-click in the Participants List pane of the selected conference, and select Paste Participant.
or
If you are using the RMX Manager and you want to paste the participant to a conference on a different
MCU:
a In the MCUs list pane, click the MCU hosting the conference into which you wish to add the
participant.
b In the Conferences list pane, click the conference into which you want to paste the copied/cut
participant.
c Right-click the Participants list pane, and select Paste Participant.
The participant is connected to the conference.
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Paste Participant As
The Paste Participant As command allows the MCU user to create a new participant using the copied
participant’s properties as a template. It automatically opens the Address Book - Participant Properties
dialog box allowing the MCU user to modify the participant’s properties effectively creating a new participant.
When the OK button in the Participant Properties dialog box is clicked, the new participant is connected
to the selected conference.
To paste a participant as a new participant:
1 In the Conferences List pane, click the conference into which you want to paste the copied/cut
participant.
Right-click in the Participants List pane of the selected conference, and select Paste Participant As
or
If you are using the RMX Manager and you want to paste the participant to a conference on another
MCU:
a In the MCUs list pane, click the MCU hosting the conference into which you wish to add the
participant.
b In the Conferences list pane, click the conference into which you want to paste the copied/cut
participant.
c Right-click the Participants List pane, and select Paste Participant As.
The Address Book - Participant Properties dialog box is displayed.
2 Modify the participant information as required. For more information, see Address Book.
Optional. If not already in the Address Book, both the copied/cut participant and the new one can be
added to the Address Book.
3 Click OK to connect the new participant to the selected conference.
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Moving Participants Between Conferences
The Collaboration Server User can move participants between ongoing conferences, including an Operator
conference, and from the Entry Queue to the destination conference if help is required.
When moving between conferences or when a participant is moved from an Entry Queue to a conference
by the Collaboration Server user (after failure to enter the correct destination ID or conference password),
the IVR messages and slide display are skipped.
Move Guidelines
● Move is available only between CP conferences (meaning, not between Video Switching
conferences).
● Move between conferences can be performed without an active Operator conference.
● When moving the conference chairperson from one conference to another, the source conference
will automatically end if the Auto Terminate When Chairperson Exits option is enabled, and that
participant is the only conference chairperson.
● When moving the Operator to any conference (following assistance request), the IVR messages and
slide display are skipped.
● Participants cannot be moved from a Telepresence conference.
● Participants cannot be moved from LPR-enabled conferences to non-LPR conferences. Move from
non-LPR conferences to LPR-enabled conferences is possible.
● Move between encrypted and non-encrypted conferences depends on the
ALLOW_NON_ENCRYPT_PARTY_IN_ENCRYPT_CONF flag setting, as described in the following
table:
Participant Move Capabilities vs. ALLOW_NON_ENCRYPT_PARTY_IN_ENCRYPT_CONF flag setting
Flag Setting
Source Conference/EQ
Encrypted
Destination
Conference Encrypted
Move Enabled?
NO
Yes
Yes
Yes
NO
Yes
No
Yes
NO
No
Yes
No
NO
No
No
Yes
YES
Yes
Yes
Yes
YES
Yes
No
Yes
YES
No
Yes
Yes
YES
No
No
Yes
● When moving dial-out participants who are disconnected to another conference, the system
automatically dials out to connect them to the destination conference.
● Cascaded links cannot be moved between conferences.
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● Participants cannot be moved to a conference if the move will cause the number of participants to
exceed the maximum number of participants allowed for the destination conference.
Moving Participants Options
Collaboration Server users can assist participants by performing the following operations:
● Move a participant to an Operator conference (Attend a participant).
● Move a participant to its Home (destination) conference.
● Move participant from one ongoing conference to another
A move can be performed using the following methods:
● Using the participant right-click menu
● Using drag and drop
To move a participant from the ongoing conference using the right-click menu options:
1 In the Conference List pane, click the conference where there are participants waiting for
Operator’s Assistance to display the list of participants.
2 In the Participants list, right-click the icon of the participant to be moved, and select one of the
following options:
 Move to Operator Conference - to move the participant to the Operator conference.
 Move to Conference - to move the participant to another ongoing AVC CP conference.
When selected, the Move to Conference dialog box opens, letting you select the name of the
destination conference.
 Back to Home Conference - if the participant was moved to another conference or to the
Operator conference, this options moves the participant back to its source conference.
This option is not available if the participant was moved from an Entry Queue to an Operator
conference or the destination conference.
Moving a Participant Interactively
You can drag and drop a participant from an Entry Queue or an ongoing conference to an Operator or
destination (Home) conference.
1 Display the participants list of the Entry Queue or the source conference by clicking its entry in the
Conferences list.
2 In the Participants list, drag the icon of the participant to the Conferences List pane, and drop it on
the Operator Conference icon or another ongoing conference.
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Personal Conference Manager (PCM) (AVC CP
Conferences)
The Personal Conference Manager (PCM) interface enables the conference chairperson to control various
conference features using his/her endpoint’s remote control device.
The following conference operations can be performed:
● Initiate Polycom’s Click&View™ application to change the local screen layout.
● Invite participants to connect to the conference.
● View and control the audio and video of each connected endpoint.
● Camera Control - control the camera of a remote endpoint using Far End Camera Control (FECC).
● Control the camera of a connected endpoint.
● Video Force a specific participant to a specific window of the video layout.
● Initiate and control recording of the conference.
● Disconnect a participant.
● Terminate the conference.
PCM Usage Guidelines
● PCM is supported in H.323 and SIP network environments.
● PCM is only available in AVC CP conferences.
● PCM can only be activated from up to 4 conferences per MPMRx card.
● If all possible PCM sessions are active on a media card and a chairperson of another conference,
hosted on the same media card, attempts to start an additional PCM session, a message is displayed
indicating that the requested PCM session cannot be activated because the maximum number of
concurrent PCM sessions are already activated. FECC is then activated.
● When PCM is active, FECC can only be performed by the chairperson using PCM’s Camera Control
menu.
● FECC is available to all IP participants in the conference with the exception of the chairperson that
has an active PCM session.
● If enabled, Message Overlay is not displayed while PCM is active.
● Normal conference video is resumed after 4 seconds of inactivity within the PCM menu.
● If a chairperson with active PCM session is moved to another conference, the PCM session is
automatically terminated.
● PCM cannot be activated from the chairperson’s endpoint when connected with QCIF resolution.
● The PCM_LANGUAGE System Flag determines the language of the PCM interface.
Possible Values: ENGLISH, CHINESE_SIMPLIFIED, CHINESE_TRADITIONAL, JAPANESE,
GERMAN, FRENCH, SPANISH, KOREAN, PORTUGUESE, ITALIAN, RUSSIAN, NORWEGIAN
Default: Current RMX Web Client language.
● The Start PCM DTMF code for initiating PCM is configured in the DTMF Codes tab of the New
Conference IVR Service or Conference IVR Service Properties dialog box. The default DTMF Code
is ## and can be modified or if required.
The default value of the Permission field is Everyone. It can be modified to Chairperson.
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● Although PCM is successfully initiated on Microsoft Office Communications client or Lync client, since
FECC feature is not available on these clients none of the PCM options are accessible as the only
available keys are #,*,1..9 and not the navigation keys ("LEFT", "RIGHT", etc.).
Modifying the PCM_FECC System Flag
The PCM_FECC System Flag determines whether the DTMF Code, ##, the Arrow Keys (FECC) or both will
activate the PCM interface. In addition, this flag can be used to disable the PCM.
Possible Values: YES / NO
Default: YES.
The table below summarizes the combined effect of the DTMF Code and PCM_FECC System Flag settings
on the system.
PCM_FECC / DTMF Code
PCM_FECC=
DTMF Code
Effect on System
Defined
Both DTMF Codes and Arrow Keys will activate PCM.
Deleted
Only the Arrow Keys can activate PCM.
Once PCM is activated the Arrow Keys lose their FECC functionality.
Defined
Only the DTMF Code can activate PCM.
Once PCM is activated the Arrow Keys lose their FECC functionality.
Deleted
PCM is disabled.
YES
NO
For more information see Modifying System Flags.
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PCM Interface
Initiating PCM
For IP endpoints: PCM is activated by pressing the Far key (
the Remote Control Device.
) followed by any of the Arrow Keys on
For ISDN/PSTN endpoints: PCM is activated by pressing ## on the Remote Control Device.
If, after activation, no keys are pressed on the Remote Control Device within a period of 10 seconds, PCM
is automatically deactivated.
When PCM becomes active, the top level of the PCM Main Menu is displayed on the conference
chairperson’s endpoint:
PCM Main Menu - Level 2 can only viewed and accessed by ISDN endpoints connected to
Collaboration Server 1800/2000/4000 by using DTMF Codes 7 and 8.
Remote Control Device Keys
PCM facilitates user interaction with the RMX using DTMF Codes and the Arrow (FECC) keys of the
endpoint’s remote control device.
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Menu Navigation - Arrow and Zoom Keys
The PCM Menu is navigated using the Remote Device’s Arrow Zoom Keys which are defined as follows:
They are defined as follows:
PCM - Menu Navigation
Arrow Key
Description
Left
•
•
•
•
•
Go up one menu level.
Delete the character to the left of the cursor when entering data.
Exit the PCM menu and return to normal conference video (from PCM the top level menu.)
Select the video window to the left when navigating a video layout.
Initiate PCM from a conference.
Right
•
•
•
•
Go down one menu level.
Confirm current selection in the PCM menu.
Select the video window to the right when navigating a video layout.
Initiate PCM from a conference.
Up
•
•
•
Cycle upward through the displayed menu options.
Select the video window above when navigating a video layout.
Initiate PCM from a conference.
Down
•
•
•
Cycle downward through the displayed menu options.
Select the video window below when navigating a video layout.
Initiate PCM from a conference.
Zoom in (+)
•
•
Confirm current selection in the PCM menu.
Zoom in on the remote camera when using FECC.
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PCM - Menu Navigation (continued)
Arrow Key
Description
Zoom out (-)
•
•
Exit PCM Menu.
Zoom out on the remote camera when using FECC.
DTMF Codes - Numeric Keys
Before using the shortcut number keys 0-9, enable the DTMF function of the endpoint according to
that endpoint’s configurations.
DTMF codes are entered using the Remote Device Numerics Keys and are defined as follows:
PCM - DTMF Codes
Numeric Key
Description
0
•
•
Number input.
Shortcut key to a numbered menu option - when conference video or the PCM menu is
displayed.
•
•
Number input.
Shortcut key to a numbered menu option - when the conference video or the PCM menu is
displayed.
Initiate PCM session on ISDN/PSTN endpoint (1 is the default for the Start PCM DTMF Code).
1-9
•
•
*
•
#
Initiate DTMF mode (with Polycom endpoints) - during a conference or when the PCM menu is
displayed.
Enter a period “.” - while entering an IP address.
Confirm selection and send information to the RMX.
PCM Main Menu - Level 1
Click & View
With the Click&View application, participants can change their Personal Layouts using the Arrow Keys or
DTMF codes entered using the Numeric Keys of their endpoints.
For a full description of Click&View see Defining a New Conference IVR Service.
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Invite Participant
To invite a participant to connect to the conference:
1 Use the Up/Down arrow keys to select Invite Participant in the Main Menu and then press the
Right arrow or # key to confirm your selection.
or
Press the 2 key on the Remote Device.
The Invite Participant sub menu is displayed.
2 Use the Numeric Keys to enter the number of the participant’s endpoint.
3 Press the # key to initiate the call.
4 Optional.
a Use the Down arrow to browse to and change the selection of the dialing method from Auto to
one of the following:
 H.323
 SIP
 VoIP
 ISDN
 PSTN
b Press the # key to initiate the call.
5 Optional.
a Use the Arrow Keys to browse to the Directory button and select the Alphabetic Grouping sub
menu.
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The Local Directory is displayed.
b Use the Numeric Keys to enter the number of the participant’s endpoint.
c Press # or the Zoom In key to initiate the call.
or
d Use the Up/Down arrow keys to select a Participant from the displayed list or use the Up/Down
and Left /Right arrow keys to display other directory listings.
A message, Press # or Zoom In to select, is displayed.
e Press # or the Zoom In key to initiate the call.
Optional. To display all participants:
 Use the Down arrow and the to select the ALL button.
 Use the Zoom In key to display the global directory.
 Use the Up/Down and Left /Right arrow keys to select a Participant.
 Press # or the Zoom In key to initiate the call.
Participants Mute/Status
The chairperson can view and control the audio and video status of each participant's endpoint. The
chairperson can:
● View the audio and video status of participants.
● Mute, block, unmute, unblock participant audio.
● Suspend or allow participant video.
● Mute all (except chairperson and lecturer).
To view and change the Participants Mute/Status:
1 Use the Up/Down arrow keys to browse to Participants Mute/Status entry in the Main Menu and
then press the Right arrow or # key to confirm your selection.
or
Press the 3 key on the Remote Device.
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The Participants Mute/Status sub menu is displayed.
The sub menu displays the Muted, Blocked and Video Suspended status of all participants’
endpoints. ALL can be selected to mute and unmute all participant endpoints with the exception of
the conference chairperson and lecturer.
2 Use the Up/Down/Left/Right arrow keys to select a Mute, Block or Suspend status icon.
3 Use the Zoom In key to toggle the state of the audio and video channels of the participants’
endpoints.
The status icons are summarized in the table below.
Participants Mute/Status - Icons
Status
Icon
Description
Audio output from this endpoint is allowed. Other conference participants can hear the audio
from this endpoint.
The endpoint is muted. Other conference participants will not hear audio from this endpoint.
The audio input to this endpoint is allowed. This endpoint can hear the conference audio.
The audio output to this endpoint is blocked. This endpoint cannot receive the conference
audio.
Video output from the endpoint is allowed. Other conference participants can see video from
this endpoint.
Video output from the endpoint is suspended. Other conference participants cannot see video
from this endpoint.
The endpoint is neither muted nor blocked.
The endpoint is both muted and blocked.
Far End Camera Control
This function is not available to chairpersons using PCM with ISDN endpoints.
The endpoint to be controlled must support Far End Camera Control (FECC).
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Using the arrow keys of the remote control device, the chairperson can control the direction, zoom and focus
of a remote endpoint’s camera.
To control a far camera:
1 Use the Up/Down arrow keys to select Camera Control in the Main Menu and then press the Right
arrow or # key to confirm your selection.
or
Press the 4 key on the Remote Device.
The Camera Control sub menu is displayed.
2 Use the Up/Down/Left/Right arrow keys to select the far camera to be controlled.
A colored frame is displayed around the selected cell in the video layout.
3 Use the Zoom In key to activate FECC.
Following a 10 second delay, the FECC active icon (
selected site.
) is displayed in the video image of the
If no camera control keys are pressed on the Remote Control Device within a period of 5 seconds,
the previous menu (Camera Control) menu is automatically displayed.
4 Use the Up/Down/Left/Right/Zoom In/Zoom Out arrow keys to control the remote camera.
5 Use the Zoom Out key on the remote control to return to the PCM Main Menu.
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Video Forcing
This function is not available to chairpersons using PCM with ISDN endpoints.
Video Forcing enables the chairperson to force the video of a specific participant to a specific window of the
video layout.
To Video Force a participant’s video:
1 Use the Up/Down arrow keys to select Video Force in the Main Menu and then press the Right
arrow or # key to confirm your selection.
or
Press the 5 key on the Remote Device.
The Video Force sub menu is displayed.
2 Use the Up/Down/Left/Right arrow keys to select the window of the video layout that you want the
specific participant to be displayed in.
3 Press the Zoom In key to cycle the display of all participants in the selected video window until the
participant you want to be displayed appears.
4 Press the Right arrow or # key to confirm your selection.
Recording
This function is not available to chairpersons using PCM with ISDN endpoints.
If a Recording Link has been set up in the Conference Profile, the conference chairperson can use the PCM
interface to start, pause, resume and stop recording. If no Recording Link exists, this function appears
grayed out and cannot be selected.
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In x mode aA Recording or Recording Paused indication is displayed in the upper left corner of the video
layout of all conference participants.
To use the recording functions:
1 Use the Up/Down arrow keys to select Recording in the Main Menu and then press the Right
arrow or # key to confirm your selection.
or
Press the 6 key on the Remote Device.
The Recording sub menu is displayed.
To start recording:
» Use the Up/Down arrow keys to select Start Recording and then press the Right arrow or # key to
confirm your selection.
or
Press the 1 key on the Remote Device.
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To pause recording:
» While recording is in progress, use the Up/Down arrow keys to select Pause and then press the
Right arrow or # key to confirm your selection.
or
Press the 2 key on the Remote Device.
To resume recording:
» While recording is paused, use the Up/Down arrow keys to select Resume and then press the Right
arrow or # key to confirm your selection.
or
Press the 2 key on the Remote Device.
To stop recording:
» While recording is in progress, use the Up/Down arrow keys to select Stop Recording and then
press the Right arrow or # key to confirm your selection.
or
Press the 3 key on the Remote Device.
PCM Main Menu - Level 2
Drop Participant
To disconnect a participant from the conference:
1 Use the Up/Down arrow keys to navigate to the second page of the Main Menu.
2 Use the Up/Down arrow keys to select Drop Participant and then press the Right arrow or # key to
confirm your selection.
or
Press the 7 key on the Remote Device.
The Drop Participant sub menu is displayed.
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3 Using the Up/Down arrow keys, select the participant to be disconnected and then press the Right
arrow or # key to confirm your selection.
or
Press the Numeric Key on the Remote Device corresponding to the participant entry.
Terminating a Conference
Although this function can be accessed and viewed by chairpersons using PCM with ISDN
endpoints, conference termination cannot be executed.
To terminate a conference:
1 Use the Up/Down arrow keys to navigate to the second page of the Main Menu
2 Use the Up/Down arrow keys to select Terminate Conference and then press the Right arrow or #
key to confirm your selection.
or
Press the 8 key on the Remote Device.
The Terminate Conference sub menu is displayed.
3 Press the Right arrow key to confirm termination of the conference.
or
Press the Left arrow key to cancel termination of the conference.
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Outlook®
Polycom Conferencing for Microsoft Outlook is supported in AVC CP Conferencing Mode only
Polycom Conferencing for Microsoft Outlook is an add-in that enables users to easily organize and invite
attendees to Video Enabled meetings via Microsoft Outlook®.
Polycom Conferencing for Microsoft Outlook is implemented by installing the Polycom Conferencing Add-in
for Microsoft Outlook on Microsoft Outlook® e-mail clients. It enables meetings to be scheduled with video
endpoints from within Outlook. The add-in also adds a Polycom Conference button in the Meeting tab of the
Microsoft Outlook e-mail client ribbon.
The meeting organizer clicks the Polycom Conference button to add Conference Information to the meeting
invitation.
Attendees call the meeting at the scheduled Start Time using the link or the dial-in number provided in the
meeting invitation.
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A Gathering Slide is displayed to connected participants until the conference starts.
The Gathering Slide displays live video along with information taken from the meeting invitation such as the
subject, meeting organizer, duration, dial-in numbers etc. At the end of the Gathering Phase, the conference
layout is displayed.
For more information see Video Preview (AVC Participants Only).
Setting up the Calendaring Solution
The following steps are performed to set up the Calendaring solution:
a The administrator installs the Polycom Conferencing Add-in for Microsoft for Microsoft Outlook
e-mail clients. For more information, see the Polycom Unified Communications Deployment
Guide for Microsoft Environments.
b The administrator creates an Microsoft Outlook e-mail-account for the Collaboration Server.
If included in the solution, Polycom RealPresence DMA system and calendaring-enabled
endpoints share this e-mail account. For more information, see the Polycom Unified
Communications Deployment Guide for Microsoft Environments.
c The administrator configures the Collaboration Server for Calendaring using the Exchange
Integration Configuration dialog box, providing it with the Microsoft Exchange Server Name, User
Name and Password and optional Primary SMTP Mail box information needed to access the
e-mail account.
To configure the Collaboration Server’s Exchange Integration Configuration:
1 On the Collaboration Server menu, click Setup > Exchange Integration Configuration.
The Exchange Integration Configuration dialog box is displayed.
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There are three options that can be used to configure the Exchange Integration Configuration. The
option you choose will depend on the configuration of the mailbox in the Exchange Server and the
configuration of the Exchange Server itself.
 Option 1 - Use this option if the Exchange Server settings have been left at their default values.
 Option 2 - Use this option if the Primary SMTP Mailbox is not the default mailbox.
 Option 3 - Use this option if the Exchange Server settings have been modified by the
administrator.
Option 1 - Using default Exchange Server settings
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a Define the following fields:
Exchange Integration Configuration - Option 1
Field
Description
Enable Calendaring Service
Select or clear this check box to enable or disable the Calendaring
Service using the Polycom Add-in for Microsoft Outlook. When this
check box is cleared all fields in the dialog box are disabled.
Exchange Server Address
Enter the IP address of the Exchange Server.
User Name
Enter the User Name of the Collaboration Server, as registered in
the Microsoft Exchange Server, that the Collaboration Server uses
to login to its e-mail account.
Field length: Up to 80 characters.
Password
Enter the Password the Collaboration Server uses to login to its
e-mail account as registered in the Microsoft Exchange Server.
Field length: Up to 80 characters.
Domain
Enter the name of the network domain where the Collaboration
Server is installed as defined in the Microsoft Exchange Server.
Primary SMTP Mailbox (Optional)
This field is left empty.
Accept Appointments
Select this check box to enable the Collaboration Server to send
replies to meeting invitations.
Clear this check box when the Collaboration Server is part of a
Unified Conferencing solution that includes a RealPresence DMA
system, as the RealPresence DMA system will send a reply to the
meeting invitation.
b Click OK.
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Option 2 - Using an alternate Primary SMTP Mailbox
a Define the following fields:
Exchange Integration Configuration - Option 2
Field
Description
Enable Calendaring Service
Exchange Server Address
User Name
These fields are defined as for Option 1 above.
Password
Domain
Accept Appointments
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Exchange Integration Configuration - Option 2
Field
Description
Primary SMTP Mailbox (Optional)
Enter the name of the SMTP Mailbox in the Microsoft Exchange
Server to be monitored by the Collaboration Server.
Note: Although several mailboxes can be assigned to each user in
the Microsoft Exchange Server, only the Primary SMTP Mailbox is
monitored. The Primary SMTP Mailbox name does not have to
contain either the Collaboration Server’s User Name or Domain
name.
b Click OK.
Option 3 - Using modified Exchange Server settings
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a Define the following fields:
Exchange Integration Configuration - Option 3
Field
Description
Exchange Server Address
If Exchange Server settings have been modified, enter the full path
to the Microsoft Exchange Server where the Collaboration Server’s
Microsoft Outlook e-mail account is registered, for example if the
EWS folder has been renamed EWD:
https://labexch01/EWD/Exchange.asmx
Note: If a server name is entered, the Collaboration Server and the
Microsoft Exchange Server must be registered to the same
Domain. (The Domain name entered in this dialog box must match
the Local Domain Name entry in the Management Network - DNS
Properties dialog box.)
For more information see Modifying the Default IP Network Service.
Field length: Up to 80 characters.
Enable Calendaring Service
User Name
Password
These fields are defined as for Option 1 above.
Domain
Primary SMTP Mailbox (Optional)
Accept Appointments
b Click the OK button.
If applicable, the Polycom® RealPresence® Media Suite, VMC, RealPresence DMA system, and
calendaring-enabled endpoints are configured with the Exchange Server Name, User Names and
Passwords needed to access their accounts.
For more information see the Polycom Unified Communications Deployment Guide for Microsoft
Environments.
The administrator configures the Collaboration Server to have a default Ad-hoc Entry Queue service
enabled.If ISDN/PSTN participants are included, up to two ISDN/PSTN dial-in numbers must be
configured for the Ad Hoc Entry Queue.
For more information see Defining a New Entry Queue.
Calendaring Guidelines
● The Collaboration Server must have its MCU prefix registered in the gatekeeper.
For more information see Modifying the Default IP Network Service.
● The Collaboration Server must be configured as a Static Route.
For more information see Modifying the Default IP Network Service.
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● The Collaboration Server’s Default Entry Queue must be configured as an Ad Hoc Entry Queue and
must be designated as the Transit Entry Queue.
For more information see the Entry Queues.
● The meeting organizer can enable recording and/or streaming of the meeting.
● If meeting is to be recorded, the Ad Hoc Entry Queue must have recording enabled in its Profile.
For more information see Defining AVC CP Conferencing Profiles.
● Meetings can be single instance or have multiple occurrences.
● Attendees that do not have video devices may be invited to the meeting.
● Attendees using e-mail applications that use the iCalendar format may be invited to meetings via the
Calendaring Service.
● Meeting invitations sent by Polycom Conferencing for Microsoft Outlook can be in a different
language to the Collaboration Server Web Client. The following languages are supported:
 English
 French
 German
 International Spanish
 Korean
 Japanese
 Simplified Chinese
● Collaboration Server resource management is the responsibility of the system administrator:
 Conferences initiated by Polycom Conferencing for Microsoft Outlook are ad hoc and therefore
resources are not reserved in advance.
 Polycom Conferencing for Microsoft Outlook Add-in assumes that sufficient resources are
available and does not check resource availability. Sufficient resources are therefore not
guaranteed.
 A meeting invitation that is automatically accepted by the Collaboration Server is not guaranteed
availability of resources.
 If the Collaboration Server runs out of resources, attendees will not be able to connect to their
conferences.
● By using RealPresence DMA system to load-balance resources between several Collaboration
Servers, resource capacity can be increased, alleviating resource availability problems.
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Creating and Connecting to a Conference
Creating a Conference
Meetings are organized using the Microsoft Outlook client in the normal manner.
If the meeting organizer decides that video participants are to be included in a multipoint video conference,
he/she clicks the Polycom Conference button. Conference Information such as the Meeting ID and
connection information is automatically added to the existing appointment information.
The meeting organizer can add a meeting agenda or personal text to the invitation before it is sent. The
meeting organizer can update or cancel the video enabled meeting in the same manner as for any other
meeting.
When the meeting organizer sends the meeting invitation a meeting record is saved in the Microsoft
Exchange Server, the RealPresence Collaboration Server, RealPresence DMA system, the Polycom®
RealPresence® Media Suite, and calendaring-enabled endpoints.
RealPresence Collaboration Servers, RealPresence DMA system, and calendaring-enabled endpoints poll
the Microsoft Exchange Server to retrieve new meeting records and updates to existing meeting records.
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Microsoft Outlook Field Usage table below summarizes the Collaboration Server’s usage of Microsoft
Outlook data fields included in the meeting invitation.
Microsoft Outlook Field Usage
Microsoft
Outlook Field
Usage by the Collaboration Server / RealPresence DMAsystem
Conference / Meeting Room
Gathering Slide
Subject
Display Name of Conference / Meeting
Room.
Meeting Name.
Start/End Time
Used to calculate the Conference’s Duration.
Record
Enable Recording in the Conference or
Meeting Room Profile.
Display Recording option.
Video Access
Number
Comprised of: <MCU Prefix in
Gatekeeper> <Conference
Numeric ID>.
Note: It is important that MCU Prefix in
Gatekeeper field in the Collaboration
Server’s IP Network Service Gatekeeper tab and the Dial-in prefix
field in the Polycom Conferencing
Add-in for Microsoft Outlook - Video
Network tab contain the same prefix
information.
If Recording and Streaming are enabled
in the Conference Profile, this number
is used as part of the recording file
name.
Displayed as the IP dial in number
in the Access Number section of
the Gathering Slide.
Audio Access
Number
ISDN/PSTN dial-in number.
Up to two numbers are supported.
Displayed as the ISDN/PSTN
dial-in number in the Access
Number section of the Gathering
Slide.
Streaming
recording link
Enables recording of conferences by
the Polycom® RealPresence® Media
Suite using the recording link.
Enables streaming of the recording of
the conference from the Polycom®
RealPresence® Media Suite.
If recording is enabled, a REC
indicator is displayed in the top left
corner of the slide.
Connecting to a Conference
Participants can connect to the conference in the following ways:
● Participants with Polycom CMA/RealPresence Desktop™ or a Microsoft Office Communicator client
running on their PCs can click on a link in the meeting invitation to connect to the meeting.
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● Participants with a HDX or a room system will receive a prompt from the endpoint’s calendaring
system along with a button that can be clicked in order to connect.
Participants with endpoints that are not calendaring-enabled can connect to the meeting by dialing
the meeting number manually.
● Participants outside the office or using PSTN or mobile phones, can use the dial in number in the
meeting invitation to manually dial in to the meeting.
Collaboration Server Standalone Deployment
When using a single Collaboration Server in a standalone deployment, connection is via an Ad Hoc Entry
Queue. The meeting is started when the first participant connects to the Collaboration Server.
When the first participant connects, a conference is created and named according to the information
contained in the dial string. Subsequent participants connecting with the same dial string are routed from
the Ad Hoc Entry Queue to the conference.
After the conference has been created the Conference Name, Organizer, Time, Duration and Password (if
enabled) are retrieved from the conference parameters for display during the Gathering Phase.
Collaboration Server and Polycom RealPresence DMA System Deployment
In a RealPresence DMA system deployment a Virtual Meeting Room is activated when the first participant
connects to the RealPresence DMA system. The RealPresence DMA system receives the dial string to
activate a Virtual Meeting Room on the Collaboration Server.
The RealPresence DMA system uses the Meeting ID contained in the dial-in string to access meeting
information stored in the Exchange Server database.
When the meeting information is found on the Exchange Server, the Conference Name, Organizer, Time,
Duration and Password (if enabled) are retrieved from the Exchange Server database for display during the
Gathering Phase.
If enabled, automatically generated passwords are ignored.
For more information see Automatic Password Generation Flags.
Polycom Solution Support
Polycom Implementation and Maintenance services provide support for Polycom solution components only.
Additional services for supported third-party Unified Communications (UC) environments integrated with
Polycom solutions are available from Polycom Global Services and its certified Partners. These additional
services will help customers successfully design, deploy, optimize and manage Polycom visual
communications within their UC environments.
Professional Services for Microsoft Integration is mandatory for Polycom Conferencing for Microsoft Outlook
and Microsoft Office Communications Server integrations. For additional information and details please see
http://www.polycom.com/services/professional_services/index.html or contact your local Polycom
representative.
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Conference and Participant Monitoring
You can monitor ongoing conferences and perform various operations while conferences are running.
Three levels of monitoring are available with the Collaboration Server:
● General Monitoring - You can monitor the general status of all ongoing conferences and their
participants in the main window.
● Conference Level Monitoring - You can view additional information regarding a specific conference
and modify its parameters if required, using the Conference Properties option.
● Participant Level Monitoring - You can view detailed information on the participant's status, using the
Participant Properties option.
General Monitoring
Users can monitor a conference or keep track of its participants and progress.
All monitoring and operations procedures performed during ongoing conferences can be performed by
either of following two methods:
● Using the buttons in the toolbars.
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● Right-clicking an entity in the Conferences or Participants pane and selecting an operation from the
menu.
Multi-Selection
Using multiple selection, you can monitor and perform simultaneous operations on multiple participants in
multiple conferences.
The selected conferences are displayed as sub-lists in the Participants list pane.
The sub-lists can be expanded and collapsed by clicking the
next to the conference name in the sub-list headings.
and
sublist control buttons that appear
Using the Chairperson Password for Filtering
If you are logged in as a chairperson, the Chairperson Password field is displayed. It enables you to
search for, and display a list of, ongoing conferences for which you have the password.
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To search for an ongoing conference by Chairperson Password:
1 Click in the Chairperson Password field.
2 Enter the password to be searched for.
3 Click the Refresh button.
The Conferences list is refreshed and ongoing conferences with the requested password are
displayed.
You can also click the blinking Participant Alerts indication bar to view participants that require attention. For
more information, see System and Participant Alerts.
Video Switching conferences appear with the HD (
CP and VSW conferences.
) icon in the conferences list to differentiate between
Monitoring is done in the same way as for CP conferences.
Conference Level Monitoring
In addition to the general conference information that is displayed in the Conference list pane, you can view
the details of the conference’s current status and setup parameters, using the Conference Properties dialog
box.
The tabs displayed in the Conference Properties dialog boxes are dependent on the Conferencing Mode,
the participant authorization.
Conference monitoring - Tab list per conferencing mode and user
Admin
Chairperson
Operator
Tab Name
CP
SVC
Mixed
VSW
CP
SVC
Mixed
VSW
CP
SVC
Mixed
VSW
General












Advanced












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Conference monitoring - Tab list per conferencing mode and user
Admin
Chairperson
Operator
Tab Name
CP
SVC
Mixed
VSW
CP
SVC
Mixed
VSW
CP
SVC
Mixed
VSW
Gathering
Settings



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Monitoring the Conference in the Conference List Pane
The Conference List pane displays information about ongoing conferences.
No status indicator display in the Status column means that the conference is running without problems.
One or more of the status indicators may appear in the Status column.
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Conferences – Monitoring Information
Field
Description
Display Name
Displays conference name and type of conference:
•
•
•
•
•
•
– AVC Conference running in CP mode.
– AVC Conference running in Video Switching mode.
– The AVC conference has been secured using the *71 DTMF code. For details,
see Secured Conference Monitoring (AVC CP Only).
– AVC Operator Conference
– Media Relay - SVC-based Conference.
– Mixed CP and SVC Conference
Status
Displays the status of the ongoing conference.
If there is no problem with the participant’s connection no indication is displayed.
If one of the following statuses occur, the appropriate indication is displayed,
proceeded by a warning icon ( ).
• Audio – There is a problem with the participant’s audio.
• Empty – No participants are connected.
• Faulty Connection – Participants are connected, but the connection is problematic.
• Not Full – Not all the defined participants are connected.
• Partially Connected – The connection process is not yet complete; the video
channel has not been connected.
• Single Participant – Only one participant is connected.
• Video – There is a problem with the participant’s video.
• Content Resource Deficiency – Content will not be sent to legacy endpoints.
• Awaiting Operator – A participant has requested operator assistance.
• Connected, Noisy – Participant’s endpoint is requesting too many intras, resulting
in the MCU ceasing to send intras to the endpoint to preserve conference quality for
all other participants.
ID
The Conference ID assigned to the conference.
Start Time
Conference start time.
End Time
The time the conference is expected to end.
Dial in Number(1)
The Conference dial in number for ISDN/PSTN participants.
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Conferences – Monitoring Information (Continued)
Field
Description
SIP Registration
The status of registration with the SIP server:
• Not configured - Registration with the SIP Server was not enabled in the
Conference Profile assigned to this conferencing Entity.
In Multiple Networks configuration, If one service is not configured while others are
configured and registered, the status reflects the registration with the configured
Network Services. The registration status with each SIP Server can be viewed in
the Properties - Network Services dialog box of each conferencing entity.
• Failed - Registration with the SIP Server failed. This may be due to incorrect
definition of the SIP server in the IP Network Service, or the SIP server may be
down, or any other reason the affects the connection between the MCU or the SIP
Server to the network.
• Registered - The conferencing entity is registered with the SIP Server.
• Partially Registered - This status is available only in Multiple Networks
configuration, when the conferencing entity failed to register to all the required
Network Services if more than one Network Service was selected.
Additional information about the conference can be viewed when accessing the conference properties.
Viewing the Properties of CP and Mixed CP and SVC Conferences
To view the parameters of an ongoing CP conference:
1 In the Conference list pane, double-click the conference, or right-click the conference and then click
Conference Properties.
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The Conference Properties - General dialog box with the General tab opens.
The following information is displayed in the General tab:
Conference Properties - General
Field
Description
Display Name
The Display Name is the conference name in native language and Unicode
character sets to be displayed in the Collaboration Server Web Client.
Note: This field is displayed in all tabs.
Duration
The expected duration of the conference using the format HH:MM.
Note: This field is displayed in all tabs.
Permanent Conference
Indicates whether the conference is set as a Permanent Conference, with no
pre-determined End Time. This conference continues until it is terminated by
an administrator, operator or chairperson.
Note: This field is displayed in all tabs.
Routing Name
The ASCII name of the conference. It can be used by H.323 and SIP
participants for dialing in directly to the conference. It is used to register the
conference in the gatekeeper and the SIP server.
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Conference Properties - General
Field
Description
Conferencing Mode
The conferencing mode set for the conference: CP, VSW, SVC Only, or CP
and SVC.
Start Time
The time the conference started.
End Time
The expected conference end time.
Note: This field is not shown when the conference is set as a Permanent
Conference.
Conference Password
A numeric password for participants to access the conference.
Chairperson Password
A numeric password used by participants to identify themselves as the
conference chairperson.
ID
The conference ID.
Profile
The name of the conference Profile from which conference parameters were
taken.
Line Rate
The maximum transfer rate, in kilobytes per second (Kbps) of the call (video
and audio streams).
Video Switching
When selected, the conference is running in a special conferencing mode
which implies that all participants must connect at the same line rate and the
video parameters are set according to the highest common parameters.
Participants with endpoints not supporting the video parameters (such as the
video protocol, resolution and frame rate) selected for the conference will
connect as secondary (audio only). If HD 1080p is selected for the conference,
endpoints that do not support HD 1080p resolution are connected as
Secondary (Audio Only) participants.
Video layout changes are not enabled during a conference.
Note: Video Switching conferencing mode is unavailable to ISDN participants.
For more information, see Video Switching (VSW) Conferencing.
Reserve Resources for
Video Participants
Displays the number of video participants for which the system reserved
resources.
Default: 0 participants.
Reserve Resources for
Audio Participants
Displays the number of audio participants for which the system reserved
resources.
Default: 0 participants.
Max Number of
Participants
Indicates the total number of participants that can be connected to the
conference. The Automatic setting indicates the maximum number of
participants that can be connected to the MCU according to resource
availability.
Enable ISDN/PSTN
Network Service
When selected, ISDN/PSTN participants can dial into the conference.
ISDN/PSTN Network
Service
When the Enable ISDN/PSTN Network Service is selected, displays the default
Network Service.
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Conference Properties - General
Field
Description
Dial-in Number (1)
Displays the conference dial in number.
Dial-in Number (2)
Displays the conference dial in number.
2 Click the Advanced tab.
The Conference Properties - Advanced dialog box opens.
3 The following information is displayed in the Advanced tab:
Conference Properties - Advanced Parameters
Field/Option
Description
Encryption
Indicates whether the conference is encrypted.
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Conference Properties - Advanced Parameters
Field/Option
Description
Packet Loss
Compensation (LPR and
DBA)
Indicates wether Packet Loss Compensation (LPR and DBA) is enabled for the
conference.
Auto Terminate
When selected, indicates that the MCU will automatically terminate the
conference when Before First Joins, At the End-After Last Quits and At the
End - When Last Participant Remains parameters apply.
Auto Redialing
Indicates whether dial-out participants are automatically (when selected) or
manually (when cleared) connected to the conference.
This option is disabled in mixed CP and SVC conferences.
Exclusive Content Mode
When selected, Content is limited to one participant.
Enable FECC
When selected, Far End Camera Control is enabled.
FW NAT Keep Alive
When selected, sends a FW NAT Keep Alive message at specific Intervals for
the RTP, UDP and BFCP channels. The interval specifies how often a FW NAT
Keep Alive message is sent. For more information, see RealPresence
Collaboration Server (RMX) Network Port Usage.
TIP Compatibility
Indicates the TIP Compatibility mode implemented for the conference, when
the environment implements the Collaboration Server and Cisco Telepresence
Systems (CTS) Integration solution.
• None
• Prefer TIP
For more information, see .
MS AV MCU cascade
mode
This field is part of the Microsoft Lync utility and defines the resource usage in
the cascading relationship between Polycom RealPresence Collaboration
Server and Microsoft Lync server.
Microsoft Lync is not supported by Collaboration Server (RMX) 18000-0.
For more information see Video Resource Requirement Selection in Lync 2013
AVMCU Cascade.
4 Click the Gathering Settings tab.
The Conference Properties - Gathering Settings dialog box opens.
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The following information is displayed:
Profile - Gathering Settings
Field/Options
Description
Enable Gathering
Indicates whether the Gathering Phase has been enabled.
Display Language
Indicates the language of the Gathering Slide field headings.
Note: When working with the Polycom Conferencing Add-in for Microsoft Outlook,
the language selected should match the language selected for the conference in
the Polycom Conferencing Add-in for Microsoft Outlook to ensure that the
Gathering Phase slide displays correctly.
Access Number 1
Indicates the ISDN or PSTN number(s) to call to connect to the conference.
Note: The numbers entered must be verified as the actual Access Numbers.
Access Number 2
IP Dial-in Number
Info 1
Additional information to be displayed during the Gathering Phase.
Info 2
Info 3
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5 Click the Video Quality tab.
The Conference Properties - Video Quality dialog box opens.
The following information is displayed:
Conference Properties - Video Quality Parameters
Field/Option
Description
People Video Definition
Video Quality
Indicates the resolution and frame rate that determine the video quality set for
the conference. Possible settings are: Motion or Sharpness. For more
information, see Video Resolutions in AVC CP Conferencing.
Maximum Resolution
Indicates the Maximum Resolution setting for the conference.
Auto (default) - indicates that the Maximum Resolution is selected in the
Resolution Configuration dialog box.
The Maximum Resolution settings for conferences and participants cannot be
changed during an ongoing conference.
Video Clarity
Indicates if Video Clarity is enabled for the conference.
Video Clarity is not support by Collaboration Server (RMX) 1800 with no DSP
cards.
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Conference Properties - Video Quality Parameters
Field/Option
Description
Auto Brightness
Indicates if Auto Brightness is selected for the conference.
Note: When Auto Brightness is enabled, color changes may be observed in
computer-based VGA Content sent by HDX endpoints through the People video
channel.
Auto Brightness is not support by Collaboration Server (RMX) 1800 with no
DSP cards.
Content Video Definition
AS-SIP
When selected, content is shared using the Multiple Resolutions mode and is
not supported in any other Content sharing mode.
For more information, see Enabling AS-SIP Content.
Multiple Content
Resolutions
Indicates if Multiple Content Resolutions mode for content sharing is enabled. In
this mode, content is shared in multiple streams, one for each video protocol:
H.263 and H.264. This allows endpoints with different protocols to connect and
disconnect without having to restart Content sharing in the middle of a
conference. For more information, see Multiple Content Resolutions.
Content Settings
Indicates the Content channel resolution set for the conference. Possible
resolutions are Graphics (default), Hi-res Graphics, Live Video, and
Customized Content Rate.
For more information, see Content Settings.
Content Protocol
Indicates the Content Protocol used for content sharing in Highest Common
Content Sharing Mode.
For more information, see MCU Usage Modes of Content Protocols.
Content Resolution
Indicates the Content Resolution and frame rate according to the selected
Content Sharing Mode (Highest common Content or Multiple Resolution
Contents) and the video protocol. For more information, see H.264 Cascade
Optimized.
Send Content to Legacy
Endpoints
(CP only)
Indicates if the Send Content to Legacy Endpoints is enabled. If enabled,
Content can be sent to H.323/SIP/ISDN endpoints that do not support H.239
Content (legacy endpoints) over the video (people) channel. For more
information see Sending Content to Legacy Endpoints.
6 Click the Video Settings tab to list the video parameters.
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Conference Properties - Video Settings Parameters
Field
Description
Presentation Mode
When checked, indicates that the Presentations Mode is active.
This option is disabled in a mixed CP and SVC conference.
For more information, see Supplemental Conferencing Features.
Lecturer View
Switching
When checked, the Lecturer View Switching enables automatic random
switching between the conference participants in the lecturer video window.
This option is disabled in a mixed CP and SVC conference.
Same Layout
When checked, forces the selected layout on all conference participants, and the
Personal Layout option is disabled.
This option is disabled in a mixed CP and SVC conference.
Auto Layout
When enabled, the system automatically selects the conference layout based on
the number of participants in the conference.
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Conference Properties - Video Settings Parameters
Field
Description
Telepresence Mode
Enabled
Indicates if the conference is running in Telepresence
Mode.
Telepresence Mode
Indicates the Telepresence Mode.
Telepresence Layout
Mode
Indicates the layout of the Telepresence Mode.
Lecturer
Indicates the name of the lecturer (if one is selected). Selecting a lecturer enables
the Lecture Mode.
This option is disabled in a mixed CP and SVC conference.
Auto Scan Interval(s)
The time interval, 10 - 300 seconds, that Auto Scan uses to cycle the display of
participants that are not in the conference layout in the selected cell.
This option is disabled in a mixed CP and SVC conference.
Video Layouts
Indicates the currently selected video layout.
These fields are enabled
if the Collaboration
Server has a
Telepresence license
installed.
See Defining New
Profiles.
7 Click the Audio Settings tab to view the audio setting for the conference.
8 If needed, you can enable or disable the Mute participants except lecturer setting.
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9 CP Only Conferences: Click the Customized Polling tab to view and modify the customized
polling for the conference.
All conference participants are listed in the left pane (All Participants) whereas the participants to be
displayed in the Auto Scan enabled cell are listed in the right pane (Scanning Order).
The dialog box buttons are summarized in the table below.
Customized Polling - Buttons
Button
Description
Add
Select a participant and click this button to add a the participant to the list of
participants to be Auto Scanned.
The participants name is removed from the All Participants pane.
Delete
Select a participant and click this button to delete the participant from the list of
participants to be Auto Scanned.
The participants name is moved back to the All Participants pane.
Add All
Add all participants to the list of participants to be Auto Scanned.
All participants’ names are removed from the All Participants pane.
Delete All
Delete all participant from the list of participants to be Auto Scanned.
All participants’ names are moved back to the All Participants pane.
Up
Select a participant and click this button to move the participant up in the Scanning
Order.
Down
Select a participant and click this button to move the participant down in the
Scanning Order.
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10 Click Apply to confirm and keep the Conference Properties dialog box open.
or
Click OK to confirm and return to the Collaboration Server Web Client main screen.
11 Click the Skins tab to view the skin selected for the conference.
You cannot select another skin during an ongoing conference.
12 Click the IVR tab to view the IVR settings.
13 Click the Information tab to view general information defined for the conference. Changes made to
this information once the conference is running are not saved to the CDR.
14 Click the Recording tab to review the recording settings for the conference.
15 Click the Site Names tab to enable or disable the display of site names during the conference, and
adjust the display properties.
16 Click the Message Overlay tab to send text messages to the conference participants during the
conference, and adjust the display properties of the text messages.
For more information, see Sending Text Messages During a Conference Using Message Overlay.
17 Click the Network Services tab to verify the SIP registration for the conference.
18 Click the Layout Indications tab to view the layout indications settings for the conference.
No changes in layout indications settings is possible for ongoing conferences.
19 Click OK to close the Conference Properties dialog box.
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Viewing the Properties of Ongoing SVC Conferences
To view the parameters of an ongoing SVC conference:
1 In the Conference list pane, double-click the SVC conference or right-click the SVC conference and
then click Conference Properties.
The Conference Properties - General dialog box with the General tab opens.
2 The following information is displayed in the General tab:
Conference Properties - General Parameters
Field
Description
Display Name
The Display Name is the conference name in native language and Unicode
character sets to be displayed in the Collaboration Server Web Client.
Note: This field is displayed in all tabs.
Duration
The expected duration of the conference using the format HH:MM.
Note: This field is displayed in all tabs.
Conferencing Mode
The conferencing mode for the conference.
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Conference Properties - General Parameters
Field
Description
Routing Name
The ASCII name of the conference. It can be used by H.323 and SIP participants
for dialing in directly to the conference. It is used to register the conference in the
gatekeeper and the SIP server.
Start Time
The time the conference started.
End Time
The expected conference end time.
Conference
Password
A numeric password for participants to access the conference.
Chairperson
Password
A numeric password used by participants to identify themselves as the conference
chairperson.
ID
The conference ID.
Profile
The name of the conference Profile from which conference parameters were taken.
Line Rate
The maximum transfer rate, in kilobytes per second (Kbps) of the call (video and
audio streams).
Video Switching
Video Switching is not supported in SVC conferences.
Reserve Resources
for Video Participants
Reserve Resources for Video Participants is not supported in SVC conferences.
Reserve Resources
for Audio Participants
Reserve Resources for Audio Participants is not supported in SVC conferences.
Max Number of
Participants
Indicates the total number of participants that can be connected to the conference.
The Automatic setting indicates the maximum number of participants that can be
connected to the MCU according to resource availability.
Enable ISDN/PSTN
Network Service
ISDN/PSTN participants are not supported in SVC conferences.
ISDN/PSTN Network
Service
ISDN/PSTN participants are not supported in SVC conferences.
Dial-in Number (1)
ISDN/PSTN participants are not supported in SVC conferences.
Dial-in Number (2)
ISDN/PSTN participants are not supported in SVC conferences.
3 Click the Advanced tab.
The Conference Properties - Advanced dialog box opens.
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4 The following information is displayed in the Advanced tab:
Conference Properties - Advanced Parameters
Field/Option
Description
Encryption
Indicates the Encryption setting for the conference.
Packet Loss
Compensation (LPR
and DBA)
Packet Loss Compensation is not supported in SVC conferences.
Auto Terminate
When selected, indicates that the MCU will automatically terminate the conference
when Before First Joins, At the End-After Last Quits and At the End - When
Last Participant Remains parameters apply.
Auto Redialing
Dial-out is not supported in SVC conferences.
Exclusive Content
Mode
When selected, Content is limited to one participant.
Enable FECC
Far End Camera Control is not supported in SVC conferences.
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Conference Properties - Advanced Parameters
Field/Option
Description
FW NAT Keep Alive
When selected, sends a FW NAT Keep Alive message at specific Intervals for the
RTP, UDP and BFCP channels. The interval specifies how often a FW NAT Keep
Alive message is sent. For more information, see RealPresence Collaboration
Server (RMX) Network Port Usage.
TIP Compatibility
TIP Compatibility is not supported in SVC conferences.
MS AV MCU cascade
mode
This field is part of the Microsoft Lync utility and defines the resource usage in the
cascading relationship between Polycom RealPresence Collaboration Server and
Microsoft Lync server.
For more information see Video Resource Requirement Selection in Lync 2013
AVMCU Cascade.
5 Click the Video Quality tab.
The Conference Properties - Video Quality dialog box opens.
The following information is displayed:
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Conference Properties - Video Quality Parameters
Field/Option
Description
People Video Definition
Video Quality
Indicates the resolution and frame rate that determine the video quality set for the
conference. In SVC conferencing, only Sharpness is supported.
Maximum Resolution
In SVC conferencing, this is always Auto (default) - The Maximum Resolution
remains as selected in the Resolution Configuration dialog box.
Video Clarity
Video Clarity is not supported in SVC conferences.
Auto Brightness
Auto Brightness is not supported in SVC conferences.
Content Video Definition
AS-SIP
AS-SIP is not supported in SVC conferences.
Multiple Content
Resolutions
Indicates if Multiple Content Resolutions mode for content sharing is enabled. In
this mode, content is shared in multiple streams, one for each video protocol: H.263
and H.264. This allows endpoints with different protocols to connect and disconnect
without having to restart Content sharing in the middle of a conference. For more
information, see Multiple Content Resolutions.
Content Settings
Indicates the Content channel resolution set for the conference. Possible
resolutions are Graphics (default), Hi-res Graphics, Live Video, and Customized
Content Rate.
For more information, see Content Settings.
Content Protocol
Indicates the Content Protocol used for content sharing in Highest Common
Content Sharing Mode.
For more information, see MCU Usage Modes of Content Protocols.
Content Resolution
Indicates the Content Resolution and frame rate according to the selected Content
Sharing Mode (Highest common Content or Multiple Resolution Contents) and the
video protocol. For more information, see H.264 Cascade Optimized.
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6 Click the Video Settings tab to view the video parameters defined for the conference.
In SVC conferences, only Auto Layout is enabled and cannot be disabled. All other video settings are
disabled.
7 Click the Audio Settings tab to view the audio parameters defined for the conference.
In SVC conferences, all Audio Settings options are disabled.
8 Click the IVR tab to view the conference IVR settings.
9 Click the Information tab to view general information defined for the conference. Changes made to
this information once the conference is running are not saved to the CDR.
10 Click the Network Services tab to view the conference IVR settings.
11 Click the Layout Indications tab to view the conference IVR settings.
12 Click OK to close the Conference Properties dialog box.
Monitoring of Operator Conferences and Participants Requiring
Assistance
This feature is applicable only in CP and Mixed CP and SVC conferences.
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Operator conferences are monitored in the same way as standard ongoing conferences.
Each Operator conference includes at least one participant - the Operator.
You can view the properties of the Operator conference by double-clicking the conference entry in the
Conferences list or by right-clicking the conference entry and selecting Conference Properties.
Requesting Help
A participant can request help using the appropriate DTMF code from his/her touch tone telephone or the
endpoint’s DTMF input device. The participant can request Individual Assistance (default DTMF code *0) or
Conference Assistance (default DTMF code 00).
Participants in Entry Queues who failed to enter the correct destination conference ID or the conference
password will wait for operator assistance (provided that an Operator conference is active).
When requiring or requesting operator assistance, the Collaboration Server management application
displays the following:
● The participant’s connection Status changes, reflecting the help request. For details, see Participants
List Status Column Icons and Indications.
● The conference status changes and it is displayed with the exclamation point icon and the status
Awaiting Operator.
● The appropriate voice message is played to the relevant participants indicating that assistance will
be provided shortly.
The following icons and statuses are displayed in the Participant Status column:
Participants List Status Column Icons and Indications
Icon
Polycom®, Inc.
Status indication
Description
Awaiting Individual Assistance
The participant has requested the operator’s assistance for
himself/herself.
Awaiting Conference
Assistance
The participant has requested the operator’s assistance for the
conference. Usually this means that the operator is requested to
join the conference.
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When the Operator moves the participant to the Operator conference for individual assistance the
participant Status indications are cleared.
Request to Speak
Participants that were muted by the conference organizer/system operator can indicate that they want to be
unmuted by entering the appropriate DTMF code.
An icon is displayed in the Role column of the Participants list for 30 seconds.
Request to Speak is:
● Activated when the participant enters the appropriate DTMF code (default: 99).
The DTMF code can be modified in the conference IVR Service Properties - DTMF Codes dialog
box.
● Available for dial-in and dial-out participants.
● A participant can request to speak more than once during the conference.
● Supported in all conference types.
● Supported in H.323 and SIP environments.
● The duration of the icon display cannot be modified.
Participant Alerts List
The Participant Alerts list contains all the participants who are currently waiting for operator assistance.
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Participants are automatically added to the Participants Alerts list in the following circumstances:
● The participant fails to connect to the conference by entering the wrong conference ID or conference
password and waits for the operator’s assistance.
● The participant requests Operator’s Assistance during the ongoing conference.
This list is used as reference only. Participants can be assisted and moved to the Operator conference or
the destination conference only from the Participants list of the Entry Queues or ongoing conference where
they are awaiting assistance.
The participants are automatically removed from the Participant Alerts list when moved to any conference
(including the Operator conference).
Participant Level Monitoring
In addition to conference information, you can view detailed information regarding the status and
parameters of each listed participant, using the Participant Properties dialog box. Participant properties can
be displayed for all participants currently connected to a conference and for defined participants that have
been disconnected.
Properties differ for IP and ISDN/PSTN participants.
SIP SVC-based participant properties are similar to SIP AVC-based participant properties.
The table below lists the tabs in the Participant Properties dialog box, as viewed by each user type, for each
participant connection types.
Participant monitoring - Tab list per participant connection type and user
Admin
Chairperson
Operator
Tab Name
AVC H.323
AVC/SVC SIP
AVC H.323
AVC/SVC SIP
AVC H.323
AVC/SVC SIP
General
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Advanced
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Information




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Media Sources
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
H.245
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
SDP
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
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Connection
Status


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Channel Status
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Channel Status Advanced
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Participant monitoring - Tab list per participant connection type and user
Admin
Chairperson
Operator
Tab Name
AVC H.323
AVC/SVC SIP
AVC H.323
AVC/SVC SIP
AVC H.323
AVC/SVC SIP
Gatekeeper
Status

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


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Call Admission
Control




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
Participant Connection Monitoring (AVC and SVC-based Connections)
When a conference is selected in the Conference list, the Participants list header displays a summary of
participant connection status.
These numbers include:
● EP = the number of Endpoints currently connected to the conference (both defined and undefined
participants). This number includes participants whose status is connected with problem, connected
partially or connected as secondary.
Connected Cascading Links are not included and are detailed separately.
● L = the number of Cascading Links currently connected to the conference.
● T = the total number of all:
 connected Participants - both defined and undefined participants
 defined participants that are currently disconnected
 Cascading Links - both connected and disconnected
If more than one conference is selected, the EP:n L:n T:n numbers reflect the cumulative connection status
information of all the selected conferences.
If no conference is selected, the EP:n L:n T:n numbers are all zeroed.
If the T(otal) number of participants is higher than the number of connected participants, it indicates that
additional participants defined for the conference have not yet connected. For example, if the EP: is 12 and
the T: is 15, it means that 12 out of 15 of the expected participants have connected and that 3 defined
participants have not yet connected.
The Participant list displays the following participant indicators and properties:
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Participant Monitoring – Indicators and Properties
Column
Icon/Description
Name
Displays the name and type (icon) of the participant:
Audio Participant – Connected via IP phone or ISDN/PSTN.
Video Participant – Connected with audio and video channels.
Status
Displays the connection status (text and icon) of the participant. If there is no problem with
the participant’s connection no indication is displayed.
Connected – The participant is successfully connected to the conference.
Disconnected – The participant is disconnected from the conference. This
status applies only to defined participants.
Waiting for Dial-in – The system is waiting for the defined participant to dial into
the conference.
Partially Connected – The connection process is not yet complete; the video
channel has not been connected.
Faulty Connection – The participant is connected, but problems occurred in the
connection, such as synchronization loss.
Secondary Connection – The endpoint’s video channel cannot be connected to
the conference and the participant is connected only via audio.
Awaiting Individual Assistance (AVC-based connection) – The participant has
requested the user’s (operator’s) assistance.
Awaiting Conference Assistance (AVC-based connection) – The participant
has requested the operator’s assistance for the conference. This usually means
that the user (operator) has been requested to join the conference.
Connected, Noisy – Participant’s endpoint is requesting too many intras,
resulting in the MCU ceasing to send intras to the endpoint to preserve
conference quality for all other participants.
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Participant Monitoring – Indicators and Properties
Column
Icon/Description
Role
Displays the participant’s role or function in the conference:
Chairperson – The participant is defined as the conference chairperson. The
chairperson can manage the conference using touch-tone signals (DTMF
codes).
Lecturer (AVC-based connection) – The participant is defined as the conference
Lecturer.
Lecturer and Chairperson – The participant is defined as both the conference
Lecturer and Chairperson.
Cascade-enabled Dial-out Participant (AVC-based connection) – A special
participant functioning as a link in a cascaded conference.
Recording (AVC-based connection) – A special participant functioning as a
Recording Link.
Note: The Recording participant does not support H.264 High Profile. If
recording a conference set to H,264 High Profile, the Recording participant
connects as Audio Only and records the conference Audio while displaying the
recording icon for the conference.
Request to speak (AVC-based connection) - Participants that were muted by
the conference organizer/system operator can indicate that they want to be
unmuted by entering the appropriate DTMF code (default 99).
The icon is displayed for 30 seconds.
IP Address/Phone
The IP participant’s IP address or the ISDN/PSTN participant’s phone number.
Alias Name/
SIP Address
The participant’s Alias Name or SIP URI.
The alias of a Polycom® RealPresence® Media Suite if the participant is functioning
as a recording link.
Network
The participant’s network connection type – H.323, or SIP or ISDN/PSTN.
Dialing Direction
Dial-in – The participant dialed the conference.
Dial-out – The MCU dialed the participant.
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Participant Monitoring – Indicators and Properties
Column
Icon/Description
Audio
Displays the status of the participant’s audio channel.
If the participant’s audio is connected and the channel is neither muted nor blocked, no
indication is displayed.
Disconnected – Participant’s audio channel is disconnected. This is a defined
participant who is waiting to be connected to the conference.
Muted – Participant’s audio channel is muted. Indicates who initiated the Mute:
participant, Collaboration Server User or MCU. The participant can still hear the
conference.
Blocked – Transmission of audio from the conference to the participant is
blocked.
Muted and Blocked - Audio channel is muted and blocked.
Video
Displays the status of the participant’s video channel.
If there is no problem with the participant’s video connection and the channel is neither
suspended nor secondary, no indication is displayed.
Disconnected – Participant’s video channel is disconnected. This is a defined
participant who is waiting to be connected to the conference.
Suspended – Video transmission from the endpoint to the conference is
suspended.
Secondary – Participant is connected only through the audio channel due to
problems with the video channel.
Encryption
Service Name
(AVC-based connection) Indicates that the endpoint is connected to the
conference using encryption.
Displays the IP Network Service used to connect this participant to the conference.
FECC Token
Participant is the holder of the FECC token and has Far End Camera Control
capabilities.
The FECC token can be allocated to only one participant at a time and remains
un-allocated if no participant requests it.
Note: FECC is not supported with ISDN.
Content Token
Participant is the holder of the Content token and has content sharing
permission.
The Content token can be allocated to only one participant at a time and remains
un-allocated if no participant requests it.
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Viewing the Properties of Participants
To view the participant Properties:
» In the Participant List pane double-click the participant entry. Alternatively, right-click a participant
and then click Participant Properties.
The Participant Properties dialog box opens, displaying the last opened tab.
Media Sources properties are not available for SVC participants.
The Media Sources dialog box enables you to mute participant’s audio, suspend participant’s video
transmission and select a personal Video Layout for the participant.
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For ISDN/PSTN participants, only the following tabs are displayed in the Participant Properties dialog
box:
• General, Advanced, Information
• Media Sources
• Connection Status
• Channel Status
The General, Advanced and Information tabs include the same properties for new and defined
participants. For more information, see Adding a Participant to the Address Book.
Monitoring IP Participants
The following parameters are displayed for an IP participant.
Participant Properties - Media Sources Parameters
Field
Description
Name
Indicates the participant’s name.
Note: This field is displayed in all tabs.
Endpoint Website
(link)
Click the Endpoint Website hyperlink to connect to the internal website of the participant’s
endpoint. It enables you to perform administrative, configuration and troubleshooting
activities on the endpoint.
The connection is available only if the IP address of the endpoint’s internal site is filled in
the Website IP Address field in the Participant Properties - General dialog box.
Note: This field is displayed in all tabs (excluding ISDN/PSTN participants).
Endpoint Type
Indicates whether the participant is using an AVC-based or SVC-based endpoint.
Fields, tabs and options are enabled or disabled according to the endpoint type.
Note: This field is displayed in all tabs.
Layout Type
Indicates whether the video layout currently viewed by the participant is the Conference or
Personal Layout.
If Personal Layout is selected, you can select a Video Layout that will be viewed only by
this participant.
Video Layout
Indicates the video layout currently viewed by the participant. When Personal Layout is
selected in the Layout Type you can force participants to the video windows in a layout
that is specific to the participant. For more information, see Setting the Video Layout in
Cascading conferences.
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Participant Properties - Media Sources Parameters
Field
Description
Mute/Suspend
Indicates if the endpoint’s audio and/or video channels have been muted/suspended. The
entity that initiated audio mute or video suspend is also indicated.
• MCU – Audio or Video channel has been muted/suspended by the MCU.
• User – Channels have been muted/suspended by the Collaboration Server user.
• Participant – Channels have been muted/suspended by the participant from the
endpoint.
You can also cancel or perform mute and suspend operation using these check boxes.
Note: If the participant muted his/her audio channel, the system displays the mute icon
only for H.323.
Block
When checked, the audio transmission from the conference to the participant’s endpoint is
blocked, but the participant will still be heard by other participants.
1 Click the Connection Status tab to view the connection status, and if disconnected the cause of the
disconnection.
This dialog box is the same for AVC-based and SVC-based participants.
The following parameters are displayed:
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Participant Properties - Connection Status Parameters
Field
Description
Participant Status
Status
Indicates the connection status of the participant.
Connection Time
The date and time the participant connected to the conference.
Note: The time format is derived from the MCU’s operating system
time format.
Disconnection Time
The date and time the defined participant disconnected from the
conference.
Connection Retries
Left
Indicates the number of retries left for the system to connect defined
participant to the conference.
Call Disconnection
Cause
Displays the cause for the defined participant’s disconnection from
the conference. See Conference and Participant Monitoring.
Video Disconnection
Cause
Displays the cause the video channel could not be connected. For
more information, see Appendix A - Disconnection Causes.
Possible Solution
In some cases, a possible solution is indicated to the cause of the
video disconnection.
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2 Click the H.245 (H.323) or SDP (SIP) tab during or after the participant’s connection process to view
information that can help in resolving connection issues.
Participant Properties - H.245/SDP Parameters
Field
Description
Remote Capabilities
Lists the participant’s capabilities as declared by the endpoint.
Remote
Communication Mode
Displays the actual capabilities used by the endpoint when
establishing the connection with the MCU (Endpoint to MCU).
Local Communication
Mode
Displays the actual capabilities used by the MCU when establishing
the connection with the participant’s endpoint (MCU to Endpoint).
3 Click on the Channel Status tab to view the status of the various channels.
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The following parameters are displayed:
Participant Properties - Channel Status Parameters
Field
Description
Channels Used
When checked, indicates the channel type used by the participant to connect to the
conference: Incoming channels are endpoint to MCU, Outgoing channels are from
MCU to endpoint.
Channels:
• H.225/Signaling - The call-signaling channel.
• H.245/SDP - The Control channel.
• Audio in - Incoming audio channel
• Audio out - Outgoing audio channel
• Video in - Incoming video channel
• Video out - Outgoing video channel
• Content in - H.239/People+Content conferences
• Content out - H.239/People+Content conferences
• FECC in - The incoming FECC channel is open.
• FECC out - The outgoing FECC channel is open.
Columns:
• Faulty – A red exclamation point indicates a faulty channel condition. This is a
real-time indication; when resolved the indication disappears. An exclamation
point indicates that further investigation may be required using additional
parameters displayed in the Advanced Channel Status tab.
• Bit Rate – The actual transfer rate for the channel. When channel is inactive, bit
rate value is 0. For example, if the participant is connected without video, the bit
rate for the video channel is 0.
Note: The CTS Audio Auxiliary channel is used only for Content. In all other
cases, the bit rate shown in this column for this channel is 0.
• Packet Loss – The accumulated count of all packets that are missing according
to the RTCP report since the channel was opened. This field is relevant only
during the connection stage and does not display faulty indications.
• Fraction Loss (Peak) – The ratio between the number of lost packets and the
total number of transmitted packets since the last RTCP report. Peak (in
parentheses) indicates the highest ratio recorded since the channel was opened.
• Number of Packets – The number of received or transmitted packets since the
channel has opened. This field does not cause the display of the faulty indicator.
• Jitter (Peak) – Displays the network jitter (the deviation in time between the
packets) as reported in the last RTCP report (in milliseconds). Peak (in
parentheses) reflects the maximum network jitter since the channel was opened.
• Latency – Indicates the time it takes a packet to travel from one end to another
in milliseconds (derived from the RTCP report).
High latency value may indicate that there is a problem in the network, or that
the endpoint is sending an incorrect RTCP values.
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Participant Properties - Channel Status Parameters
Field
Description
Sync Status
•
•
•
•
•
•
Channel - The channel type: Video or Content.
Source - The name of the participant currently viewed by this participant.
Position - The video layout position indicating the place of each participant as
they appear in a conference.
Protocol Sync Loss - Indicates whether the system was able to synchronize the
bits order according to the selected video protocol.
Video Intra Sync - Indicates whether the synchronization on a video Intra frame
was successful.
Video Resolution - The video resolution of the participant.
Rx - Rate
The received line rate.
Tx - Rate
The transmitted line rate.
Tx - Video Sync Loss
When checked, indicates a video synchronization problem in the outgoing channel
from the MCU.
The counter indicates the sync-loss count.
Rx - Video Sync Loss
When checked, indicates a video synchronization problem in the incoming channel
from the endpoint.
The counter indicates the sync-loss count.
Tx - LPR Activation
When checked, indicates LPR activation in the outgoing channel.
Rx - LPR Activation
When checked, indicates LPR activation in the incoming channel.
FECC Token
When checked, indicates that the participant is the holder of the FECC Token.
Content Token
When checked, indicates that the participant is the holder of the Content Token.
4 Click the Channel Status Advanced tab to view additional information for selected audio and video
channels.
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In the Channel Status - Advanced tab, channels can be selected for viewing additional information:
Participant Properties - Channel Status Advanced Parameters
Field
Description
Channel Info
Select a channel to view its information:
• H.225
• H.245
• Audio in
• Audio out
• Video in
• Video out
• Content in
• Content Out
• SIP BFCP TCP
Collaboration Server
IP Address
The IP address and the transport protocol (TCP/UDP) of the MCU to
which the participant is connected and the port number allocated to
the participant incoming media stream on the MCU side.
Participant IP
Address
The IP address and the transport protocol (TCP/UDP) of the
participant and the port number allocated to the media stream on the
participant side.
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Participant Properties - Channel Status Advanced Parameters
Field
Description
ICE RealPresence
Collaboration Server
1800/2000/4000 IP
Address
The IP address, port number, and transport protocol of the MCU used
to pass through the media when ICE is functional. See Monitoring
Participant Connections.
ICE Participant IP
Address
The IP address, port number, and transport protocol of the endpoint
used to pass through the media when ICE is functional. See
Monitoring the Participant Connection in ICE Environment.
ICE Connection Type
Indicates the type of connection between the Collaboration Server
and the participant in the ICE environment:
• Local (or Host) - The endpoint (Remote) is on the same network
as the Collaboration Server and the media connection is direct,
using local addresses.
• Relay - Media between the Collaboration Server and the
participant passes through a media relay server.
• Firewall - Media connection between the Collaboration Server and
the participant is done using their external IP addresses (the IP
addresses as seen outside of the local network).
Media Info
This table provides information about the audio and video parameters,
such as video algorithm, resolution, etc. For more information, see
Appendix E - Participant Properties Advanced Channel Information.
RTP Statistics
This information may indicate problems with the network which can
affect the audio and video quality. For more information, see Appendix
E - Participant Properties Advanced Channel Information.
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5 For H.323 AVC-based participants only - Click the Gatekeeper Status tab to view its parameters.
Participant Properties - Gatekeeper Status Parameters
Field
Description
Requested
Bandwidth
The bandwidth requested by the MCU from the gatekeeper.
Allocated Bandwidth
The actual bandwidth allocated by the gatekeeper to the MCU.
Required Info Interval
Indicates the interval, in seconds, between registration messages
that the MCU sends to the gatekeeper to indicate that it is still
connected.
Gatekeeper State
Indicates the status of the participant’s registration with the
gatekeeper and the bandwidth allocated to the participant. The
following statuses may be displayed:
• ARQ – Admission Request - Indicates that the participant has
requested the gatekeeper to allocate the required bandwidth on
the LAN.
• Admitted – Indicates that the gatekeeper has allocated the
required bandwidth to the participant.
• DRQ – Disengage Request – The endpoint informs the gatekeeper
that the connection to the conference is terminated and requests
to disconnect the call and free the resources.
• None – Indicates that there is no connection to the gatekeeper.
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6 For SIP AVC-based and SVC-based participants - Click the Call Admission Control tab to view
its parameters.
Participant Properties - Gatekeeper Status Parameters
Field
Description
Requested
Bandwidth
The bandwidth requested by the MCU from the SIP server.
Allocated Bandwidth
The actual bandwidth allocated by the SIP server to the MCU.
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Monitoring SIP BFCP Content
In the SIP Participant Properties dialog box, BFCP status information appears in:
● All three panes of the SDP tab.
● The Channel Status tab.
● The Channel Status - Advanced tab.
For more information see Participant Level Monitoring.
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Detecting Endpoint Disconnection
Endpoint disconnect can be detected for
● SIP endpoints - Detecting SIP Endpoint Disconnection
● H.323 endpoints - H.323 Endpoint Disconnection Detection
Detecting SIP Endpoint Disconnection
When an abnormal disconnection of SIP endpoints occurs because of network problems or client
application failures, SIP endpoints remain connected to the conference causing connection disruptions. For
example, the video freezes in the layout or blocks content for SIP endpoints when a quick re-connection is
performed. It can take several minutes to detect the SIP endpoint disconnection using the SIP standard
behavior.
In a normal SIP video call, audio and video (RTP and RTCP) messages are sent from the endpoints to the
MCU to detect the signaling of connected endpoints. Conversely, SVC endpoints might not send video RTP
messages to the MCU when a participant is not displayed in the video layout of any of the participants in the
conference. For SVC endpoints, the MCU will only verify audio RTP and RTCP messages and video RTCP
messages. Video RTP messages will not be checked.
To detect the disconnection of SIP endpoints in a reasonable amount of time, a new system flag can be
defined to specify the amount of time that the MCU should wait for an RTCP or RTP message from the SIP
endpoint before the endpoint starts the disconnection process. The system default value is automatically
set to 20 seconds.
The system flag, DETECT_SIP_EP_DISCONNECT_TIMER, contains the amount of time in seconds to wait
for an RTCP or RTP message to be received from the endpoint. When the time that was set in the system
flag has elapsed and no RTCP or RTP audio or video message has been received on either the audio or
the video channel, the MCU disconnects the SIP endpoint from the conference. A CDR event record is
created with a Call Disconnection Cause of SIP remote stopped responding.
The Microsoft Lync add-in endpoint opens audio and content channels. Lync endpoints can send
RTCP/RTP messages and empty RTP audio messages. When the time that was set in the system flag has
elapsed and no RTCP or RTP message has been received on the audio channel, the MCU disconnects the
endpoint from the conference.
SIP audio only endpoints use the audio channel only. When the time that was set in the system flag has
elapsed and no RTCP or RTP message has been received on the audio channel, the MCU disconnects the
SIP audio endpoint from the conference.
Configuring the System Flag
The time out used for SIP endpoint disconnection detection is controlled by the
DETECT_SIP_EP_DISCONNECT_TIMER System Flag which must be added to the System Configuration
to view or modify its value.
Range: 0 - 300
When the value is set between 0 and 14, the feature is disabled and SIP endpoints are not detected for
disconnection. When the value is set between 15 and 300, the feature is enabled.
Default: 20
For more information see , and Modifying System Flags.
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H.323 Endpoint Disconnection Detection
In versions previous to version 8.4, when an H.323 endpoint disconnected, round trip messages were still
received, even when a gatekeeper was deployed. Because of this, the MCU failed to detect the endpoint
disconnection.
From version 8.4 onward, the MCU detects H.323 endpoint connection or disconnection by monitoring
RTCP/RTP messages reception using either the audio or video channels. When these messages are not
received within a predefined timeout interval, the endpoint is considered disconnected. Therefore as long
as either RTCP or RTP messages (interchangeably) are received on either of the video or audio channels,
the endpoint is considered connected.
If no messages are received through either channel within the predefined time out interval, the endpoint is
disconnected, and a disconnection message, H.323 remote stopped responding, is sent to the
endpoint.
No channel disconnection detection occurs while an endpoint’s video or audio are muted, or while the
endpoint is put on hold.
In audio calls, only the audio channel is polled for RTCP/RTP messages.
System Configuration Flag
The time out used for H.323 endpoint disconnection detection is controlled by the
DETECT_H323_EP_DISCONNECT_TIMER System Flag which must be added to the System
Configuration to view or modify its value.
Range: 16 - 300 (4-second units). Values indivisible by 4 will be rounded upward. Flag values between 0
and 15 disable the flag functionality.
Default: 20
For more information see , and Modifying System Flags.
Monitoring ISDN/PSTN Participants
Using the Participant Properties dialog box, you can monitor and verify the properties of an ISDN/PSTN
participant. The dialog box’s tabs contain information that is relevant to the participant’s status only while
the conference is running and is used to monitor the participant’s status when connection problems occur.
Maximum line rate at which ISDN endpoints can connect to a conference is 768 kbps.
The table below lists the audio algorithms that are supported for ISDN participants according to their
connection bit rate
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:
Supported Audio Algorithms vs Bit Rate
Bit Rate
Audio
Algorithm
96Kbps
(and Lower)
128Kbps – 192Kbps
256Kbps (and
Higher)
G722.1 16K
G722.1 C 32K
G722.1 C 48K
G722.1 C 24K
G722.1 C 24K
G722.1 C 32K
Siren14 24K
Siren14 32K
G722.1 C 24K
G722 48K
Siren14 24K
Siren14 48K
G722 56K
G722.1 32K
Siren14 32K
G722 64K
G722.1 24K
Siren14 24K
G711 56K
G722 48K
G722.1 32K
G711 64K
G722 56K
G722.1 24K
G722 64K
G722.1 16K
G711 56K
G722 48K
G711 64K
G722 56K
G722 64K
G711 56K
G711 64K
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To view the participant properties during a conference:
1 In the Participants list, right click the desired participant and select Participant Properties.
The Participant Properties - Media Sources dialog box is displayed.
ISDN/PSTN Participant Properties - Media Sources
Field
Description
Mute/Suspend
Indicates if the endpoint’s audio and/or video channels from the endpoint have been
muted/suspended.
The entity that initiated audio mute or video suspend is also indicated.
• MCU – Audio or Video channel has been muted/suspended by the MCU.
• User – Channels have been muted/suspended by the Collaboration Server user.
• Participant – Channels have been muted/suspended by the participant from the
endpoint.
You can also cancel or perform mute and suspend operation using these check
boxes.
Block (Audio)
When checked, the audio transmission from the conference to the participant’s
endpoint is blocked, but the participant will still be heard by other participants.
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2 Click the H.221 tab to view additional information that can help to resolve connection issues.
Participant Properties - H.221 Parameters
Field
Description
Remote Capabilities
Lists the participant’s capabilities as declared by the endpoint.
Remote
Communication Mode
Displays the actual capabilities used by the endpoint when
establishing the connection with the MCU (Endpoint to MCU).
Local Communication
Mode
Displays the actual capabilities used by the MCU when establishing
the connection with the participant’s endpoint (MCU to Endpoint).
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3 Click the Connection Status tab to view general information regarding the participant connection
and disconnection causes of the participant to the conference.
ISDN/PSTN Participant Properties - Connection Status
Field
Description
Status
Indicates the connection status of the participant to the conference. If
there is a problem, the appropriate status is displayed, for example,
Disconnected.
Connection Time
The date and time the participant connected to the conference.
Disconnection Time
The date and time the participant was disconnected from the
conference.
Connection Retries
Left
Indicates the number of retries left for the system to connect the
participant to the conference.
Call Disconnection
Cause
For a full list of Disconnection Causes, Appendix A - Disconnection
Causes.
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4 Click the Channel Status tab to view the status of a participant’s channels.
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ISDN/PSTN Participant Properties - Channel Status
Field
Description
Connected Media
Indicates if the participant is connected with Audio, Video and
Content media channels.
Channels Used
•
•
•
Channel – Indicates the channel used by the participants and
whether the channel is connected (indicated with a check mark) or
disconnected.
Participant Phone Number – In a dial-in connection, indicates the
participant’s CLI (Calling Line Identification) as identified by the
MCU.
In a dial-out connection, indicates the participant’s phone number
dialed by the MCU for each channel.
MCU Phone Number – In a dial-in connection, indicates the MCU
number dialed by the participant.
In a dial-out connection, indicates the MCU (CLI) number as seen
by the participant. This is the number entered in the MCU Number
field in the Network Service.
Tx - Video Sync Loss
When checked, indicates a video synchronization problem in the
outgoing channel from the MCU.
The counter indicates the sync-loss count.
Rx - Video Sync Loss
When checked, indicates a video synchronization problem in the
incoming channel from the endpoint.
The counter indicates the sync-loss count.
Content Token
A check mark indicates that the participant is the current holder of the
Content Token.
The Connected Media and Channels Used fields of an Audio Only participant are displayed as
follows:
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Monitoring Telepresence Participant Properties
A Telepresence status indicator is displayed in the Participant Properties - Advanced tab when
monitoring conference participants.
The Telepresence mode of the participant is indicated:
● RPX - The participant’s endpoint is transmitting 4:3 video format.
● TPX - The participant’s endpoint is transmitting 16:9 video format.
● None - The participant’s endpoint is neither RPX nor TPX.
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Recording and Streaming Conferences
Conferences running on the Collaboration Server can be recorded using the Polycom® RealPresence®
Media Suite in the following ways:
● The Polycom® RealPresence® Media Suite dials in MCU for conference recording via a SIP interface
from the Polycom® RealPresence® Media Suite. From the Polycom® RealPresence® Media Suite
Administration UI, you can start a conference recording by dialing out to a Collaboration Server. After
the Collaboration Server gets the request from the Polycom® RealPresence® Media Suite, the
recording will start if the recording is enabled on the Collaboration Server. During the conference
recording, you can pause or stop the recording from the Polycom® RealPresence® Media Suite
Administration UI. For more information on how to start a recording by dialing out to an inter-operable
endpoint see the Polycom® RealPresence® Media Suite User’s Guide.
● Collaboration Server dials out to Polycom® RealPresence® Media Suite for a conference recording.
Recording conferences is enabled via a dial-out Recording Link, which is a dial-out connection from
the conference to the recording system.
Note: SVC Conferencing Mode
Conference recording is not available in SVC Conferencing Mode.
The recording system can be installed at the same site as the conferencing MCU or at a remote site. Several
MCU’s can share the same recording system.
Recording conferences is enabled via a Dial Out Recording Link, which is a dial-out connection from the
conference to the recording system.
Recording can start automatically, when the first participant connects to a conference, or on request, when
the Collaboration Server user or conference chairperson initiates it.
Multiple Dial Out Recording Links may be defined.
Conference Dial Out Recording Links can be associated on the Collaboration Server with Virtual Recording
Rooms (VRR), created and saved on the Polycom® RealPresence® Media Suite.
Each Dial Out Recording Link defined on the Collaboration Server can be given a descriptive name and can
be associated with one VRR saved on the Polycom® RealPresence® Media Suite.
The following guidelines apply:
● A Dial Out Recording Link that is being used by an ongoing conference cannot be deleted.
● A Dial Out Recording Link that is assigned to a Profile cannot be deleted.
● The Dial Out Recording Link supports H.264 High Profile with H.323 connections.
● While a Profile is being used in an ongoing conference, it cannot have a different Dial Out Recording
Link assigned to it.
● Up to 100 Dial Out Recording Links can be listed for selection in the Conference Profile.
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● Multiple Dial Out Recording Links are supported in Continuous Presence and Video Switched
conferences.
● The number of Dial Out Recording Links available for selection is determined by the value of the
MAXIMUM_RECORDING_LINKS System Flag in system.cfg. Default value is 20 Dial Out Recording
Links.
● The Dial Out Recording Link can be encrypted when recording from an encrypted conference to the
Polycom® RealPresence® Media Suite that is set to encryption. For more details, see Dial Out
Recording Link Encryption.
Creating Multiple Virtual Recording Rooms on the
Polycom® RealPresence® Media Suite
If the environment includes a Polycom® RealPresence® Media Suite and you want to associate Dial Out
Recording Links on the Collaboration Server with Virtual Recording Rooms (VRR) created and saved on
the Polycom® RealPresence® Media Suite, perform the following operations on the Polycom®
RealPresence® Media Suite:
1 Modify the parameters of a recording Template to meet the recording requirements.
2 Assign the modified recording Template to a VRR. The recording and streaming server will assign a
number to the VRR.
3 Repeat step 1 and step 2 for each VRR to create additional VRRs.
For more information see the Polycom® RealPresence® Media Suite User Guide.
Configuring the Collaboration Server to Enable
Recording
To make recording possible the following components you must be configured on the Collaboration Server:
● Dial Out Recording Link – Defines the connection between the conference and the recording
system.
● Recording-enabled Conference IVR Service – Recording DTMF codes and messages must be set
in the Conference IVR Service to enable recording-related voice messages to be played and to allow
the conference chairperson to control the recording process using DTMF codes.
● Recording-enabled Profile – Recording must be enabled in the Conference Profile assigned to the
recorded conference.
If Multiple Dial Out Recording Links are being defined for Virtual Recording Rooms (VRRs), created and
saved on the Polycom® RealPresence® Media Suite, the MAXIMUM_RECORDING_LINKS System Flag
in system.cfg can be modified to determine the number of Dial Out Recording Links available for selection.
● Range: 20 - 100
● Default: 20
The flag value can be modified by selecting the System Configuration option from the Setup menu. For more
information, see Modifying System Flags.
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Defining the Dial Out Recording Link
The Dial Out Recording Link is defined once and can be updated when the H.323 alias or the IP address
(of the recording system) is changed. Only one Dial Out Recording Link can be defined in the Collaboration
Server. Its type must be H.323.
In Multiple Networks Configuration, Dial Out Recording Links use the default Network Service to
connect to conferences, therefore the recording system must be defined on the default IP Network
Service to enable the recording.
To define a Dial Out Recording Link:
1 In the RMX Management pane, click Recording Links (
).
2 In the Recording Links list, click the New Recording Link (
) button.
The New Recording Link dialog box is displayed.
3 Define the following parameters:
Recording Link Parameters
Parameter
Description
Name
Displays the default name that is assigned to the Recording Link.
If multiple Recording Links are defined, it is recommended to use a descriptive
name to be indicate the VRR to which it will be associated.
Default: Recording Link
Type
Select the network environment:
• H.323
• SIP
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Recording Link Parameters
Parameter
Description
IP Address
•
•
•
Alias Name
If no gatekeeper is configured, enter the IP Address of the Polycom®
RealPresence® Media Suite. Example: If the Polycom® RealPresence®
Media Suite IP address is 173.26.120.2, enter 173.26.120.2.
If a gatekeeper is configured, you can either enter the IP address or an alias (see
the alias description).
If SIP server is configured, enter the IP address of the SIP server instead of the
IP address of RealPresence Media Suite.
If using the endpoint’s alias instead of IP address, first select the alias type and
then enter the endpoint’s alias.
If you are associating this recording link to a VRR on the Polycom®
RealPresence® Media Suite, define the alias as follows:
• If you are using the Polycom® RealPresence® Media Suite IP address,
enter the VRR number in the Alias field. For example, if the VRR number is 5555,
enter 5555.
• Alternatively, if the Alias Type is set to H.323 ID, enter the Polycom®
RealPresence® Media Suite IP address and the VRR number in the format:
<RealPresence_Media_Suite_IP_Address>##<VRR number>
For example: If the Polycom® RealPresence® Media Suite IP is
173.26.120.2 and the VRR number is 5555, enter 173.26.120.2##5555
Alias Type
Depending on the format used to enter the information in the IP address and Alias
fields, select H.323 ID or E.164 (for multiple Recording links). E-mail ID and
Participant Number are also available.
4 Click OK.
The Dial Out Recording Link is added to the Collaboration Server unit.
Enabling the Recording Features in a Conference IVR Service
To record a conference, a Conference IVR Service in which the recording messages and DTMF codes are
activated must be assigned to the conference. The default Conference IVR Service shipped with the
Collaboration Server includes the recording-related voice messages and default DTMF codes that enable
the conference chairperson to control the recording process from the endpoint. You can modify these default
settings.
To modify the default recording settings for an existing Conference IVR Service:
1 In the RMX Management pane, click IVR Services (
).
The IVR Services are listed in the IVR Services list pane.
2 To modify the default recording settings, double-click the Conference IVR Service or right-click and
select Properties.
The Conference IVR Service Properties dialog box is displayed.
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3 To assign voice messages other than the default, click the General tab and scroll down the list of
messages to the recording messages.
4 Select the Recording In Progress message, and then select the appropriate message file (by
default, Recording_in_Progress.wav) from the file list to the right of the field.
5 Select the Recording Failed message, and then select the appropriate message file (by default,
Recording_Failed.wav) from the file list to the right of the field.
6 To modify the default DTMF codes, click the DTMF Codes tab.
7 To modify the DTMF code or permission for a recording function:
a Select the desired DTMF name (Start, Stop or Pause Recording), click the DTMF code entry
and type a new code.
Default DTMF Codes assigned to the recording process
Recording Operation
DTMF Code
Permission
Start or Resume Recording
*2
Chairperson
Stop Recording
*3
Chairperson
Pause Recording
*1
Chairperson
b In the Permission entry, select whether this function can be used by all conference participants
or only the chairperson.
8 Click OK.
Enabling the Recording in the Conference Profile
To be able to record a conference, the recording options must be enabled in the Conference Profile
assigned to it. You can add recording to existing Profiles by modifying them.
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To enable recording for a conference:
1 In the Collaboration Server Management pane, click Conference Profiles (
).
The Conference Profiles list is displayed.
2 Create a new profile by clicking New Profile ( ), or modify an existing profile by double-clicking or
right-clicking an existing profile and then selecting Profile Properties.
If creating a new profile, complete the conference definition. For more information on creating Profiles
see Defining AVC CP Conferencing Profiles.
3 In the New Profile dialog box, click the Recording tab.
4 Select the Enable Recording check box.
5 Define the following parameters
Conference Profile Recording Parameters
Parameter
Description
Enable Recording
Select to enable Recording Settings in the dialog box.
Recording Link
Select a recording link for the conference from the list.
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Conference Profile Recording Parameters
Parameter
Description
Start recording
Select one of the following:
• Immediately – conference recording is automatically started upon
connection of the first participant.
• Upon Request – the operator or chairperson must initiate the
recording (manual).
Audio only
Select this option to record only the audio channel of the conference.
Note: An Audio Only Recording Link cannot be used to record a
conference if there are no Voice resources allocated in the
Video/Voice Port Configuration.
Display Recording
Icon
Select this option to display Recording Indications to all conference
participants informing them that the conference is being recorded.
The recording icon is replaced by a Paused icon when conference
recording is paused.
For more information see New AVC CP Profile - Layout Indications
Parameters and Recording Indications
Play Recording
Message
Selected by default. A message is played to all participants
announcing that the conference is being recorded. Un check this box
to prevent the announcement from being played.
6 Click OK.
Recording is enabled in the Conference Profile.
Dial Out Recording Link Encryption
The Dial Out Recording Link can be encrypted when recording an encrypted conference. The encryption of
the Dial Out Recording Link is enabled when Encryption is selected in the Conference Profile on the
Collaboration Server and on the Polycom® RealPresence® Media Suite, and the system flag
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF is set to NO.
Dial Out Recording Link Encryption Guidelines:
● The Dial Out Recording Link connection type must be H.323.
● The Dial Out Recording Link uses the AES encryption format.
● The Polycom® RealPresence® Media Suite recorder must be set to support encryption. For more
information see the Polycom® RealPresence® Media Suite User Guide.
● Encryption must be selected in the Conference Profile.
Dial Out Recording Link Encryption Flag Setting
Dial Out Recording Links are treated as regular participants, however if the
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF System Flag is set to YES a
non-encrypted Dial Out Recording Link is to be allowed to connect to an encrypted conference.
The following table summarizes the connection possibilities for a Dial Out Recording Link that is to be
connected to a conference for each of the conference profile and Entry Queue encryption options.
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Connections by Recording Link and Conference Encryption Settings
Conference Profile
Setting
Dial Out Recording Link Connection Status according to flag:
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF
YES
NO
Encrypt All
Connected encrypted if possible,
otherwise connected non-encrypted.
Connected only if encrypted,
otherwise disconnected.
No Encryption
Connected non-encrypted.
Connected non-encrypted.
Encrypt when possible
Connected encrypted if possible,
otherwise connected non-encrypted.
Connected encrypted if possible,
otherwise connected non-encrypted.
Dial Out Recording Link Settings
The recording of encrypted conferences via an encrypted Dial Out Recording Link is enabled in the
Conference Profile by:
1 Selecting the Encryption option (Encrypt All or Encrypt when Possible) in the Advanced tab.
For more details, see Packet Loss Compensation - LPR and DBA.
2 Setting the Recording options in the Recording tab. For more details, see Enabling the Recording
in the Conference Profile.
Managing the Recording Process
When a conference is started and recording is enabled in its Profile, the system will automatically start the
recording if the Start Recording parameter is set to immediately. If it is set to Upon Request, the system waits
for the chairperson or Collaboration Server user’s request. Once the recording is initiated for a conference,
the MCU connects to the recording device (Polycom® RealPresence® Media Suite). The connection that is
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created between the conference and the recording device is represented as a special participant
(Recording) whose name is the Recording Link. Once the recording has started, the recording process can
be stopped and restarted from the Chairperson’s endpoint (using DTMF codes) or from the Collaboration
Server Web Client. After the recording process has finished, the recording can be identified in the Polycom®
RealPresence® Media Suite by its Collaboration Server conference name.
A conference participant and the Recording Link cannot have identical names, otherwise the
recording process will fail.
Recording Layout
When the video layout of the conference is set to Auto Layout, the recording of the conference will now
include all the conference participants and not n-1 participants as in previous versions.
In the new Auto Layout algorithm, the Recording Link is counted as a participant, and therefore it is excluded
from the layout display used for the recording. The layout used for the other participants will behave as in
the standard Auto Layout behavior.
The Recording Layout can be changed during an ongoing conference in the same manner as for any other
conference participant. For more information see the Participant Level Monitoring.
The default settings for Auto Layout for the conference and the Recording Link are summarized in the
following table:
Recording Link Default Layout Settings (Auto Layout Mode)
Number of Participants
Conference Default Auto Layout Settings
Recording Link Auto Layout Settings
0
N/A
N/A
1
2
3
4
5
6
7
8
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Recording Link Default Layout Settings (Auto Layout Mode)
Number of Participants
Conference Default Auto Layout Settings
Recording Link Auto Layout Settings
9
10 or more
When the Polycom® RealPresence® Media Suite dials in to the MCU for conference recording, you
can choose the video layout from the Polycom® RealPresence® Media Suite Admin UI. The following
layout is supported by Collaboration Server.
• Auto: automatic layout setting at the Collaboration Server side.
• Single View: record 1x1 layout.
• Dual View: record 1x2 layout.
The default settings for Auto Layout of the Recording Link cannot be changed, and the Auto Layout flags
do not apply to the Recording Link Auto Layout default settings.
Using the Collaboration Server Web Client to Manage the Recording
Process
To manage the recording process using the right-click menu:
» Right-click the Recording participant in the conference and select from one of the following options:
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Recording Participant Right-click Options
Name
Description
Suspend Video
The Suspend Video option prevents the incoming video of the recording link
participant to be part of the conference layout.
The Recording Link participant is set by default to Suspend Video. The Suspend
Video option toggles with the Resume Video option.
Resume Video
The Resume Video option enables the incoming video of the recording link
participant to be part of the conference layout.
This feature may be used to play back previously recorded video or audio feeds in
the conference layout. For more information, see the Polycom® RealPresence®
Media Suite User Guide.
Participant Properties
The Participant Properties option displays viewing only information for monitoring,
e.g. communication capabilities and channels used to connect to the conference.
Users will not be able to perform any functional requests from this window, i.e.
disconnect, change layout and mute.
To manage the recording process using the Conference toolbar:
» In the Conferences pane, click one of the following buttons in the Conference tool bar.
The recording buttons will only be displayed in the conference tool bar for a conference that is
recording-enabled.
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Conferences List - Recording Tool bar buttons
Button
Description
Start/Resume recording. This button toggles with the Pause button.
Stop recording.
Pause recording. This button toggles with the Start/Resume button.
Using DTMF Codes to Manage the Recording Process
By entering the appropriate DTMF code on the endpoint, the chairperson can Stop the recording (*3),
Pause it (*1), or Start/Resume the recording (*2). For more information on managing the recording process
via DTMF codes, see the Polycom® RealPresence® Media Suite User Guide.
Conference Recording with Codian IP VCR
Conference recording is available with Codian VCR 2210, VCR 2220 and VCR 2240.
Recording between the Collaboration Server and the Codian VCR is enabled by adding an IP participant to
the recorded conference that acts as a link between the conference and the recording device. This
participant is identified as a recording link to the Codian VCR according to the product ID sent from the VCR
during the connection phase, in the call setup parameters.
The video channel between the conference and the recording device is unidirectional where the video
stream is sent from the conference to the recorder.
If the Codian VCR opens a video channel to the conference - this channel is excluded from the conference
video mix.
To record a conference running on the Collaboration Server using Codian recorder:
» In the conference, define or add a dial-out participant using the Codian VCR IP address as the
address for dialing.
Once added to the conference, the MCU automatically connects the participant (the link to Codian VCR)
and the recording is automatically started on the Codian VCR.
A connection can also be defined on the Codian VCR, dialing into the recorded conference using the MCU
prefix and the Conference ID as for any other dial-in participant in the conference.
Monitoring the recording participant
This connection is monitored as any other participant in the conference. The connection can also be
monitored in the Codian VCR web client.
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Collaboration Server Users
Collaboration Server Web Client users are defined in the User’s table and can connect to the MCU to
perform various operations.
Note: Resource Capacities and Operational Specifications
For information about Resource Capacities and other operational specifications, see the
RealPresence® Collaboration Server (RMX) Hardware Guide specific to your system.
User Types
The authorization level determines a user’s capabilities within the system. The MCU supports the following
user Authorization Levels:
 Administrator
 Operator
 Machine Account (Application-user)
 Administrator Read-only
 Chairperson
 Auditor
Administrator
An administrator can define and delete other users, and perform all configuration and maintenance tasks.
A default user with administrator authorization is provided by all MCU types to be used with Polycom
products, having the following details:
● User name - SA_PLCM_Integration
● Password - Polycom_CS
This user is not considered a new user. Therefore, no New SA_PLCM_Integration user message
should be generated.
Since this user is provided in secure mode (JITC) as well, an active alarm is displayed upon login, indicating
the existence of an SA_PLCM_Integration default user, and recommending replacing it with an alternate
one for security reasons.
This user should be recognized by the DMA/XMA as well, thus enable their logging into the MCU without
any undue messages, with the exception of the active alarm stated above upon logging into a secure
machine.
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Administrator Read-only
A user with Administrator permission with the same viewing and monitoring permissions of a regular
Administrator. However, this user is limited to creating system backups and cannot perform any other
configuration or conference related operation.
Operator
An Operator can manage Meeting Rooms, Profiles, Entry Queues, and SIP Factories, and can also view
the Collaboration Server configurations, but cannot change them.
Administrator and Operator users can verify which users are defined in the system. Neither of them can view
the user passwords, but an Administrator can change a password.
Chairperson
A Chairperson can only manage ongoing conferences and participants, in both single and cascading MCUs
scenarios. The Chairperson does not have access to the Collaboration Server configurations and utilities.
Auditor
An Auditor can only view Auditor Files and audit the system.
Machine Account
User names can be associated with servers (machines) to ensure that all users are subject to the same
account and password policies. For more details, see Machine Account later on this chapter.
Listing Users
The Users pane lists the currently defined users in the system and their authorization levels. The pane also
enables the administrators to add and delete users.
The system is shipped with a default Administrator user called POLYCOM, whose password is POLYCOM.
However, once you have defined other authorized Administrator users, it is recommended to remove the
default user.
You can view the list of users that are currently defined in the system.
To view the users currently defined in the system:
1 In the RMX Management pane, click Users (
).
The Users pane is displayed.
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The list includes three columns: User Name, Authorization Level and Disabled:
 User Name - The login name used by the user to connect to the MCU.
 The Authorization - Indicates the Authorization Level assigned to the User: Administrator,
Administrator Read-only, Operator, Chairperson or Auditor.
 Disabled - Indicates whether the user is disabled and cannot access the system unless enabled
by the administrator. For more details, see Disabling a User.
Locked indicates whether the user has been locked out and cannot access the system unless
enabled by the administrator.
In Ultra Secure Mode (ULTRA_SECURE_MODE=YES), Users can be automatically disabled or
locked out by the system when they do not log into the Collaboration Server application for a
predefined period or if their login session does not meet Enhanced Security requirements. Users can
be manually disabled by the administrator. For more details, see Notes.
Adding a New User
Administrators can add new users to the system.
The User Name and Password must be in ASCII.
To add a new user to the system:
1 In the RMX Management pane, click Users (
).
2 The Users pane is displayed.
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3 Click New User (
), or right-click anywhere in the pane and then click New User.
The User Properties dialog box opens.
4 In the User Name text box, enter the name of the new user. This is the login name used by the user
when logging into the system.
5 In the Password text box, enter the new user’s password. This will be the user’s password when
logging into the system.
6 In the Authorization Level list, select the user type: Administrator, Administrator Read-Only,
Operator, Chairperson or Auditor.
7 To associate a user with a machine:
a In the User Properties dialog box, select the Associate with a machine check box.
b Enter the FQDN of the server that hosts the application who’s application-user name is being
added.
8 Click OK.
The User Properties dialog box closes and the new user is added to the system.
Deleting a User
To delete a user, you must have Administrator authorization. The last remaining Administrator in the
Users list cannot be deleted.
1 In the RMX Management pane, click Users (
2 Select the user and click Delete (
).
), or right-click the user and then click Delete User.
The system displays a confirmation message.
3 In the confirmation dialog box, select Yes to confirm or No to cancel the operation.
If you select Yes, the user name and icon are removed from the system.
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Changing a User’s Password
Users with Administrator authorization can change their own password and other users’ passwords. Users
with Operator authorization can change their own password.
To change a user’s password:
1 In the RMX Management pane, click Users (
).
2 Right-click the user and click Change User Password.
The Change Password dialog box opens.
3 Enter the Old Password (current), New Password and Confirm the New Password.
The Password must be in ASCII.
4 Click OK.
The user’s password is changed.
Disabling a User
An administrator can disable an enabled user. An indication is displayed in the Users List when the User is
disabled. An administrator can enable a disabled User.
To disable a user:
1 In the RMX Management pane, click Users (
).
The Users pane is displayed.
2 In the Users pane, right-click the user to be disabled and select Disable User in the menu.
A confirmation box is displayed.
3 Click YES.
The User status in the Users list - Disabled column changes to Yes.
Enabling a User
An administrator can enable a User who was disabled automatically by the system (in the Ultra Secure
Mode) or manually by the administrator.
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To enable a user:
1 In the RMX Management pane, click Users (
).
The Users pane is displayed.
2 Right-click the user to be enabled and select Enable User.
A confirmation box is displayed.
3 Click YES.
The User status in the Users list - Disabled column changes to NO.
Renaming a User
To rename a user:
1 In the RMX Management pane, click Users (
).
The Users pane is displayed.
2 Right-click the user to be renamed and select Rename User.
The Rename User dialog box is displayed.
3 Enter the user’s new name in the New User Name field and click OK.
The user is renamed and is forced to change his/her password.
Machine Account
User names can be associated with servers (machines) to ensure that all users are subject to the same
account and password policies.
For enhanced security reasons it is necessary for the Collaboration Server to process user connection
requests in the same manner, whether they be from regular users accessing the Collaboration Server via
the Collaboration Server Web Browser / RMX Manager.
Regular users can connect from any workstation having a valid certificate while application-users
representing applications can only connect from specific servers. This policy ensures that a regular user
cannot impersonate an application-user to gain access to the Collaboration Server in order to initiate an
attack that would result in a Denial of Service (DoS) to the impersonated application.
The connection process for an application-user connecting to the Collaboration Server is as follows:
1 The application-user sends a connection request, including its TLS certificate, to the Collaboration
Server.
2 The Collaboration Server searches its records to find the FQDN that is associated with the
application-user’s name.
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3 If the FQDN in the received certificate matches that associated with application-user, and the
password is correct, the connection proceeds.
Guidelines for defining a machine account
● Application-users are only supported when TLS security is enabled and Request peer certificate is
selected. TLS security cannot be disabled until all application-user accounts have been deleted from
the system.
● For Secure Communications, an administrator must set up on the Collaboration Server system a
machine account for the RealPresence DMA system with which it interacts. This machine account
must include a fully-qualified domain name (FQDN) for the RealPresence DMA system.
● Application-user names are the same as regular user names.
Example: the DMA application could have an application-user name of DMA1.
● The FQDN can be used to associate all user types: Administrator, Operator with the FQDN of a
server.
● Multiple application-users can be configured the same FQDN name if multiple applications are
hosted on the same server
● If the system is downgraded the application-user’s FQDN information is not deleted from the
Collaboration Server’s user records.
● A System Flag, PASS_EXP_DAYS_MACHINE, enables the administrator to change the password
expiration period of application-user’s independently of regular users. The default flag value is 365
days.
● The server hosting an application-user whose password is about to expire will receive a login
response stating the number of days until the application-user’s password expires. This is determined
by the value of the PASSWORD_EXPIRATION_WARNING_DAYS System Flag. The earliest
warning can be displayed 14 days before the password is due to expire and the latest warning can
be displayed 7 days before passwords are due to expire. An Active Alarm is created stating the
number of days before the password is due to expire.
● The MIN_PWD_CHANGE_FREQUENCY_IN_DAYS System Flag does not effect application-user
accounts. Applications typically manage their own password change frequency.
● If an application-user identifies itself with an incorrect FQDN, its account will not be locked, however
the event is written to the Auditor Event File.
● If an application-user identifies itself with a correct FQDN and an incorrect password, its account will
be locked and the event written to the Auditor Event File.
● An application-user cannot be the last administrator in the system. The last administrator must be
regular user.
● User names are not case sensitive.
Monitoring
● An application-user and its connection is represented by a specific icon.
Active Directory
● When working with Active Directory, the RealPresence DMA system cannot be registered within
Active Directory as regular users. The RealPresence DMA system application-users must be
manually.
● The only restriction is that TLS mode is enabled together with client certificate validation.
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● If the above configuration are set off it will not be possible to add machine accounts.
● When setting the TLS mode off the system should check the existence of a machine account and
block this operation until all machine accounts are removed.
Connections
The Collaboration Server enables you to list all connections that are currently logged into the MCU, e.g.
users, servers or API users. The MCU issues an ID number for each login. The ID numbers are reset
whenever the MCU is reset.
Viewing the Connections List
To list the users who are currently connected to the MCU:
1 In the Collaboration Server Management pane, click Connections (
).
A list of connected users is displayed in the Connections pane.
The information includes:
 The user’s login name.
 The user’s authorization level (Chairperson, Operator, Administrator or Auditor).
 The time the user logged in.
 The name/identification of the computer used for the user’s connection.
Notes
Notes are the electronic equivalent of paper sticky notes. You can use notes to write down questions,
important phone numbers, names of contact persons, ideas, reminders, and anything you would write on
note paper. Notes can be left open on the screen while you work.
Using Notes
Notes can be read by all system Users concurrently connected to the MCU. Notes that are added to the
Notes list are updated on all workstations by closing and re-opening the Notes window. Notes can be written
in any Unicode language.
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Users Connections and Notes
To create a note:
1 On the Collaboration Server menu, select Administration > Notes.
The Notes window opens.
2 In the Notes toolbar, click New Note (
select New Note.
), or right-click anywhere inside the Notes window and
3 In the Note dialog box, type the required text and click OK.
The new note is saved and closed. The Notes list is updated, listing the new note and its properties:
 Note – The beginning of the note’s text.
 Last Modified – The date of creation or last modification.
 Modified By – The Login Name of the user who last modified the note.
 Modified From – The Client Application and Workstation from which the note was created or
modified.
To open or edit a note:
» Double-click the entry to edit, or right-click the entry and select Note Properties.
The note opens for viewing or editing.
To delete a note:
1 In the Notes list, select the entry for the note to delete and click Delete Note (
entry and select Delete Note.
), or right-click the
A delete confirmation dialog box is displayed.
2 Click OK to delete the note, or click Cancel to keep the note.
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IP Network Services
IP Network Services enable the Collaboration Server to function within IP network environments. They
include the network parameters required for the MCU to connect with other IP devices on the same network
or outside the network through a firewall.
Collaboration Server IP Network Services Overview
Two types of IP Network Services are defined for the Collaboration Server:
● Management Network Service
● Default IP Service (Conferencing Service)
Connection between the Collaboration Server management applications (Web Client And RMX Manager)
and participant connections to conferences (Dial in, dial out) are supported within the following IP
addressing environments: IPv4, IPv6 and IPv6 & IPv4
When IPv4 is selected, IPv6 fields are hidden and conversely when IPv6 is selected, IPv4 fields are hidden.
When IPv6 & IPv4 is selected both IPv6 and IPv4 fields are displayed.
For the purposes of comprehensive documentation, all screen captures in this chapter pertaining to
RealPresence Collaboration Servers (RMX) 1800/2000/4000 show the dialog boxes as displayed with
IPv6 & IPv4 selected.
For more information on IPv6, see LAN Redundancy.
When the RMX is configured for IPv4 and IPv6 Addressing, the addition of the sdp-anat option tag in the
SIP Require and SIP Supported headers allows a mixture of IPv4 and IPv6 addressing to be specified by
the Session Description Protocol (SDP). For more information see Management Network (Primary).
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The IP Network Services are configured by selecting this option in the RMX Management pane of the
Collaboration Server Web Client/RMX Manager application.
Management Network (Primary)
The Management Network is used to connect between the Collaboration Server and the management
applications (Collaboration Server Web Client or RMX Manager application) and enable these applications
to control the MCU. It contains the network parameters, such as the IP address of the MCU’s control unit,
required for connection between the Collaboration Server and the management applications. You can use
this IP address to connect to the control unit should the MCU become corrupted or inaccessible.
During First Time Power-up, the Management Network parameters can be set either via a USB key or by
using a cable to create a private network. For more information, see First Entry Power-up and Configuration
and Appendix G - Configuring Direct Connections to the Collaboration Server in the RealPresence
Collaboration Server (RMX) Getting Started Guide.
Default IP Service (Conferencing Service)
The Default IP Service (Conferencing Service) is used to configure and manage communications between
the Collaboration Server and conferencing devices such as endpoints, gatekeepers, SIP servers, etc.
The Default IP Service contains parameters for:
● Signaling Host IP Address
● External conferencing devices
Calls from all external IP entities are made to the Signaling Host, which initiates call set-up and assigns the
call to the appropriate media card.
Conferencing related definitions such as environment (H.323 or SIP) are also defined in this service.
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Most of the Default IP Service is configured by the Fast Configuration Wizard, which runs automatically
should the following occur:
● First time power-up.
● Deletion of the Default IP Service, followed by a system reset.
For more information, see First Entry Power-up and Configuration in the Polycom® RealPresence
Collaboration Server (RMX) 1800/2000/4000 Getting Started Guide
Changes made to any of these parameters only take effect when the Collaboration Server is reset.
An Active Alarm is created when changes made to the system have not yet been implemented and
the MCU must be reset.
Using IPv6 Networking Addresses for Collaboration Server Internal
and External Entities
IPv6 addresses can be assigned to both Collaboration Server (Internal) and External Entity addresses.
Collaboration Server Internal Addresses (Default Management Network Service):
● Control Unit
● Signaling Host
● Shelf Management
● MPM1 (Media Card)
● MPM2 (Media Card)
External Entities:
● Gatekeepers (Primary & Secondary)
● SIP Proxies on EMA
● DNS Servers
● Default Router
● Defined participants
IPv6 Addressing Guidelines
● Internet Explorer 7™ is required for the Collaboration Server Web Client and RMX Manager to
connect to the Collaboration Server using IPv6.
● The default IP address version is IPv4.
● The IP address field in the Address Book entry for a defined participant can be either IPv4 or IPv6. A
participant with an IPv4 address cannot be added to an ongoing conference while the Collaboration
Server is in IPv6 mode nor can a participant with an IPv6 address be added while the Collaboration
Server is in IPv4 mode.
An error message, Bad IP address version, is displayed and the New Participant dialog box remains
open so that the participant’s address can be entered in the correct format.
● Participants that do not use the same IP address version as the Collaboration Server in ongoing
conferences launched from Meeting Rooms, Reservations and Conference Templates, and are
disconnected. An error message, Bad IP address version, is displayed.
● IP Security Protocols (IPSec) are not supported.
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Modifying the Management Network
The Management Network parameters need to be modified if you want to:
● Connect directly to the Collaboration Server from a workstation
● Modify routes
● Modify DNS information
To view or modify the Management Network Service:
1 In the RMX Management pane, click the IP Network Services (
) button.
2 In the IP Network Services list pane, double-click the Management Network (
) entry.
The Management Network Properties - IP dialog box opens
Management Network Properties - Collaboration Servers (RMX) 2000/4000
On the RealPresence Collaboration Server (RMX) 2000 an additional tab called LAN Ports appears.
For more information on the LAN Ports tab see Step 8.
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Management Network Properties - Collaboration Server (RMX) 1800
3 Modify the following fields:
Default Management Network Service – IP
Field
Description
Network Service Name
Displays the name of the Management Network. This name cannot be modified.
Note: This field is displayed in all Management Network Properties tabs.
IP Version
IPv4
Select this option for IPv4 addressing only.
IPv6
Select this option for IPv6 addressing only.
IPv4 & IPv6
Select this option for both IPv4 and IPv6 addressing.
Note: If the gatekeeper cannot operate in IPv6
addressing mode, the H323_RAS_IPV6 System Flag
should be set to NO.
For more information see The CS System Flags.
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Default Management Network Service – IP
Field
Description
IPv6 Configuration
Method
Auto (Stateless)
Manual Configuration
Method is
recommended with
IPv6.
Select this option to allow automatic generation of the
following addresses:
• Link-Local (For internal use only)
• Site-Local
• Global
Manual
Select this option to enable manual entry of the
following addresses:
• Site-Local
• Global
Manual configuration of the following address types is
not permitted:
• Link-Local
• Multicast
• Anycast
Control Unit IP Address
IPv4
The IPv4 address of the Collaboration Server. This IP
address is used by the Collaboration Server Web
Client to connect to the Collaboration Server.
IPv6
The IPv6 address of the MCU. This IP address is used
by the Collaboration Server Web Client to connect to
the Collaboration Server.
Note: Internet Explorer 7™ is required for the
Collaboration Server Web Client to connect to the
MCU using IPv6.
All
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Click the All button to display the IPv6
addresses as follows:
• Auto - If selected, Site-Local and Global
site addresses are displayed.
• Manual - If selected, only the Manual site
address is displayed.
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Default Management Network Service – IP
Field
Description
Shelf Management IP
Address
IPv4
The IPv4 address of the RMX Shelf Management
Server. This IP address is used by the Collaboration
Server Web Client for Hardware Monitoring purposes.
Collaboration Server
2000/4000 only)
IPv6
The IPv6 address of the RMX Shelf Management
Server. This IP address is used by the Collaboration
Server Web Client for Hardware Monitoring purposes.
Note: Internet Explorer 7™ is required for the
Collaboration Server Web Client to connect to the
MCU using IPv6.
All
Subnet Mask
Click the All button to display the IPv6
addresses as follows:
• Auto - If selected, Site-Local and Global
site addresses are displayed.
• Manual - If selected, only the Manual site
address is displayed.
Enter the subnet mask of the Control Unit.
Note: This field is specific to IPv4 and is not displayed in IPv6 only mode.
4 Click the Routers tab.
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5 Modify the following fields:
Default Management Network Service – Routers
Field
Description
Default Router IP
Address
IPv4
IPv6
Static Routes
(IPv4 Only Table)
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Enter the IP address of the default router. The default router
is used whenever the defined static routers are not able to
route packets to their destination. The default router is also
used when host access is restricted to one default router.
The system uses Static Routes to search other networks
for endpoint addresses that are not found on the local LAN.
Up to five routers can be defined in addition to the Default
Router. The order in which the routers appear in the list
determines the order in which the system looks for the
endpoints on the various networks. If the address is in the
local subnet, no router is used.
To define a static route (starting with the first), click the
appropriate column and enter the required value.
Router IP
Address
Enter the IP address of the router.
Remote IP
Address
Enter the IP address of the entity to be reached outside the
local network. The Remote Type determines whether this
entity is a specific component (Host) or a network.
• If Host is selected in the Remote Type field, enter the IP
address of the endpoint.
• If Network is selected in the Remote Type field, enter of
the segment of the other network.
Remote Subnet
Mask
Enter the subnet mask of the remote network.
Remote Type
Select the type of router connection:
• Network – defines a connection to a router segment in
another network.
• Host – defines a direct connection to an endpoint found
on another network.
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6 Click the DNS tab.
7 Modify the following fields:
Default Management Network Service – DNS
Field
Description
MCU Host Name
Enter the name of the MCU on the network.
Default name is RMX
DNS
Select:
• Off – If DNS servers are not used in the network.
• Specify –To enter the IP addresses of the DNS servers.
Note: The IP address fields are enabled only if Specify is selected.
Register Host Names
Automatically to DNS
Servers
Select this option to automatically register the MCU Signaling Host and Shelf
Management with the DNS server.
Local Domain Name
Enter the name of the domain where the MCU is installed.
DNS Servers Addresses
Primary Server
Secondary Server
The static IP addresses of the DNS servers.
A maximum of three servers can be defined.
Tertiary Server
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8 Click the Security tab.
9 Modify the following fields:
Management Network Properties – Security Parameters
Field
Description
Secured
Communication
Select to enable Secured Communication.
The Collaboration Server supports TLS 1.0 and SSL 3.0 (Secure Socket Layer).
A SSL/TLS Certificate must installed on the Collaboration Server for this feature to
be enabled. This box is checked by default when the MCU is in Ultra Secure Mode.
For more information see Appendix F - Secure Communication Mode.
Skip certificate
validation for user
logging session
Select this check box to prevent peer certificate requests being issued.
For more information see (PKI) Public Key Infrastructure.
This check box must be cleared when enabling Secured Mode. If it is not cleared
an Active Alarm is created and a message is displayed stating that Secured
Communications Mode must be enabled.
Revocation Method
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Management Network Properties – Security Parameters
Field
Description
Global Responder
URL
Use Responder
Specified in
Certificate
For a detailed description of these fields see Certificate Managementand Certificate
Revocation.
Allow Incomplete
Revocation Checks
Skip Certificate
Validation for OSCP
Responder
For a detailed description see Certificate Management and Certificate Revocation.
10 Click OK.
11 If you have modified the Management Network Properties, reset the MCU.
12 To define a white list, click the White List tab.
A White List contains the addresses of IP Networking Entities permitted to connect to the RMX’s
Management Network; Networking Entities such as Network Hosts, Control Workstations,
Gatekeepers SIP/ DNS Servers, etc.
For a detailed description of these fields see White List Access.
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Modifying the Default IP Network Service
The Default IP Network Service is defined initially during the First Time Power-up or if the Default IP Service
has been deleted, followed by an Collaboration Server restart. For details, see Polycom® RealPresence
Collaboration Server (RMX) 1800/2000/4000 Getting Started Guide, Procedure 4: Modifying the Default IP
Service and ISDN/PSTN Network Service Settings.
Once the Default IP Network Service is defined, you can modify its properties through the IP Network
Properties dialog boxes. The Default IP Service parameters need to be modified if you want to change the:
● Network type that the Collaboration Server connects to
● IP address of the Collaboration Server Signaling Host
● IP addresses of the Collaboration Server Media boards
● Subnet mask of the Collaboration Server’s IP cards
● Gatekeeper parameters or add gatekeepers to the Alternate Gatekeepers list
● SIP server parameters
To view or modify the Default IP Service:
1 In the RMX Management pane, click IP Network Services (
).
2 In the Network list pane, double-click the Default IP Service (
,
, or
) entry.
The Default IP Service - Networking IP dialog box is displayed.
Default IP Service - Collaboration Servers (RMX) 2000/4000
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Default IP Service - Collaboration Server (RMX) 1800
3 Modify the following fields:
Default IP Network Service – IP
Field
Description
Network Service Name
The name Default IP Service is assigned to the IP Network Service by the Fast
Configuration Wizard. This name can be changed.
Note: This field is displayed in all IP Signaling dialog boxes and can contain
character sets that use Unicode encoding.
IP Network Type
Displays the network type selected during the First Entry configuration. The
Default IP Network icon indicates the selected environment.
You can select:
• H.323 - For an H.323-only Network Service.
• SIP - For a SIP-only Network Service.
• H.323 & SIP - For an integrated IP Service. Both H.323 and SIP participants
can connect to the MCU using this service.
Note: This field is displayed in all Default IP Service tabs.
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Default IP Network Service – IP
Field
Description
Signaling Host IP
Address
On RealPresence Collaboration Server 1800 this field is disabled as only one IP
address is used for signaling and media transmission.
Enter the address to be used by IP endpoints when dialing in to the MCU.
Dial out calls from the Collaboration Server are initiated from this address.
This address is used to register the Collaboration Server with a Gatekeeper or a
SIP Proxy server.
Media Card 1 IP Address
Collaboration Server 1800: Enter the IP address of the media card to be used
by IP endpoints when dialing in to the MCU.
Collaboration Server 2000/4000: Enter the IP address of the first media card
as provided by the network administrator.first media card
Endpoints connect to conferences and transmit call media (video, voice and
content) via these addresses.
Media Card 2 IP Address
(Collaboration Server
2000/4000)
Collaboration Server 2000/4000: Enter the IP address of the second media
card if installed.
Endpoints connect to conferences and transmit call media (video, voice and
content) via these address.
Media Card 3 IP Address
(Collaboration Server
4000)
Collaboration Server 4000: Enter the IP address of the third media cards if
installed.
Endpoints connect to conferences and transmit call media (video, voice and
content) via these addresses.
Media Card 4 IP Address
(Collaboration Server
4000)
Collaboration Server 4000: Enter the IP address of the fourth media cards if
installed.
Endpoints connect to conferences and transmit call media (video, voice and
content) via these addresses.
Subnet Mask
Enter the subnet mask of the MCU.
Default value: 255.255.255.0.
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4 Click the Routers tab.
With the exception of IP Network Type, the field definitions of the Routers dialog box are the same
as for the Default Management Network. For more information see Default Management Network
Service – Routers.
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5 Optional. Click the DNS tab.
6 In the DNS field select Specify.
7 In the DNS Server Address field, enter the IP address of the DNS Server for the IP Network
Service.
 If the DNS field in the IP Network Service is set to Specify and the DNS is not configured or
disabled, the DNS configured for the Management Network will be used.
 When upgrading from a version that does not support a DNS per IP Network Service, the DNS
configured for the Management Network will be used.
 In both Standard Security and Ultra Secure Modes:
 A separate DNS can be configured for the Management Network Service and the IP Network
Service.
 If a Multiple Services Licence is installed, a separate DNS can be configured for each
additional IP Network Service that is defined. For more information see Multiple Network
Services.
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8 To view or modify the gatekeeper parameters, click the Gatekeeper tab.
9
Modify the following fields:
Default IP Service – Conferencing – Gatekeeper Parameters
Field
Description
Gatekeeper
Select Specify to enable configuration of the gatekeeper IP address.
When Off is selected, all gatekeeper options are disabled.
Primary Gatekeeper IP
Address or Name
Enter either the gatekeeper’s host name as registered in
the DNS or IP address.
Alternate Gatekeeper IP
Address or Name
Enter the DNS host name or IP address of the gatekeeper
used as a fallback gatekeeper used when the primary
gatekeeper is not functioning properly.
MCU Prefix in
Gatekeeper
Enter the number with which this Network Service registers in the gatekeeper.
This number is used by H.323 endpoints as the first part of their dial-in string
when dialing the MCU.
When PathNavigator or SE200 is used, this prefix automatically registers with
the gatekeeper. When another gatekeeper is used, this prefix must also be
defined in the gatekeeper.
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Note: When in
IPv4&IPv6 or in
IPv6 mode, it is
easier to use
Names instead of
IP Addresses.
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Default IP Service – Conferencing – Gatekeeper Parameters
Field
Description
Register as Gateway
Select this check box if the Collaboration Server is to be seen as a gateway, for
example, when using a Cisco gatekeeper.
Notes:
• Do not select this check box when using Polycom ReadiManager or a
Radvision gatekeeper.
• Gateway functionality is not supported by Collaboration Server (RMX) 1800
with no DSP cards.
Refresh Registration
every __ seconds
The frequency with which the system informs the gatekeeper that it is active by
re-sending the IP address and aliases of the IP cards to the gatekeeper. If the IP
card does not register within the defined time interval, the gatekeeper will not
refer calls to this IP card until it re-registers. If set to 0, re-registration is
disabled.
Note:
• It is recommended to use default settings.
• This is a re-registration and not a ‘keep alive’ operation – an alternate
gatekeeper address may be returned.
Aliases
Alias
The alias that identifies the Collaboration Server’s Signaling Host within the
network. Up to five aliases can be defined for each Collaboration Server.
Note: When a gatekeeper is specified, at least one alias must be entered in the
table.
Additional aliases or prefixes may also be entered.
Type
The type defines the format in which the card’s alias is sent to the gatekeeper.
Each alias can be of a different type:
• H.323 ID (alphanumeric ID)
• E.164 (digits 0-9)
• Email ID (email address format, e.g. abc@example.com)
• Participant Number (digits 0-9, * and #)
Note: Although all types are supported, the type of alias to be used depends on
the gatekeeper’s capabilities.
10 To view or modify the ports values, click the Ports tab.
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Settings in the Ports tab allow specific ports in the firewall to be allocated to multimedia conference
calls.
The port range recommended by IANA (Internet Assigned Numbers Authority) is 49152 to 65535.
The Collaboration Server uses this recommendation along with the number of licensed ports to
calculate the port range.
11 Modify the following fields:
Default IP Service – Conferencing – Ports Parameters
Field
Description
Fixed Ports
Leave this check box cleared if you are defining a Network Service for local calls
that do not require configuring the firewall to accept calls from external entities.
When cleared, the system uses the default port range and allocates 4 RTP and
4 RTCP ports for media channels (Audio, Video, Content and FECC).
Note: When ICE Environment is enabled, 8 additional ports are allocated to
each call.
Click this check box to manually define the port ranges or to limit the number of
ports to be left open.
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Default IP Service – Conferencing – Ports Parameters
Field
Description
TCP Port from - to
Displays the default settings for port numbers used for signaling and control.
To modify the number of TCP ports, enter the first and last port numbers in the
range.
The number of ports is calculated as follows:
Number of simultaneous calls x 2 ports (1 signaling + 1 control).
UDP Port from - to
Displays the default settings for port numbers used for audio and video.
To modify the number of UDP ports:
• Enter the first and last port numbers in the range, and the range must be 3000
ports per media card.
• When ICE environment is enabled, the range must be 6000 ports per media
card.
If you do not specify an adequate port range, the system will accept the settings but will issue a
warning. Calls will be rejected when the Collaboration Server’s ports are exceeded.
12 If required, click the QoS tab.
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Quality of Service (QoS) is important when transmitting high bandwidth audio and video information.
QoS can be measured and guaranteed in terms of:
• Average delay between packets
• Variation in delay (jitter)
• Transmission error rate
DiffServ and Precedence are the two QoS methods supported by the Collaboration Server. These
methods differ in the way the packet’s priority is encoded in the packet header.
The Collaboration Server’s implementation of QoS is defined per Network Service, not per endpoint.
The routers must support QoS in order for IP packets to get higher priority.
13 View or modify the following fields:
Default IP Service – Conferencing – QoS Parameters
Field
Description
Enable
Select to enable the configuration and use of the QoS settings.
When un-checked, the values of the DSCP (Differentiated Services Code Point)
bits in the IP packet headers are zero.
Type
DiffServ and Precedence are two methods for encoding packet priority. The
priority set here for audio video and IP Signaling packets should match the
priority set in the router.
• DiffServ: Select when the network router uses DiffServ for priority encoding.
The default priorities for both audio and video packets is 0x31. These values
are determined by the QOS_IP_VIDEO and QOS_IP_AUDIO flags in the
system.cfg file.
The default priority for Signaling IP traffic is 0x00 and is determined by the
QOS_IP_SIGNALING flag in the system.cfg file.
For more information Modifying System Flags.
• Precedence: Select when the network router uses Precedence for priority
encoding, or when you are not sure which method is used by the router.
Precedence should be combined with None in the TOS field.
The default priority is 5 for audio and 4 for video packets.
Note: Precedence is the default mode as it is capable of providing priority
services to all types of routers, as well as being currently the most common
mechanism.
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Default IP Service – Conferencing – QoS Parameters
Field
Description
Audio / Video
You can prioritize audio and video IP packets to ensure that all participants in
the conference hear and see each other clearly. Select the desired priority. The
scale is from 0 to 5, where 0 is the lowest priority and 5 is the highest. The
recommended priority is 4 for audio and 4 for video to ensure that the delay for
both packet types is the same and that audio and video packets are
synchronized and to ensure lip sync.
TOS
Select the type of Service (TOS) that defines optimization tagging for routing the
conferences audio and video packets.
• Delay: The recommended default for video conferencing; prioritized audio
and video packets tagged with this definition are delivered with minimal delay
(the throughput of IP packets minimizes the queue sequence and the delay
between packets).
• None: No optimization definition is applied. This is a compatibility mode in
which routing is based on Precedence priority settings only. Select None if
you do not know which standard your router supports.
14 Click the SIP Servers tab.
15 Modify the following fields:
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Default IP Network Service – SIP Servers
Field
Description
SIP Server
Select:
• Specify – To manually configure SIP servers.
• Off – If SIP servers are not present in the network.
Note: When set to Specify, the Security tab is displayed.
SIP Server Type
Select:
• Generic - For non Microsoft environments.
• Microsoft - For Microsoft SIP environments.
Refresh Registration
This defines the time in seconds, in which the Collaboration Server refreshes it’s
registration on the SIP server. For example, if “3600” is entered the
Collaboration Server will refresh it’s registration on the SIP server every 3600
seconds.
Transport Type
Select the protocol that is used for signaling between the Collaboration Server
and the SIP Server or the endpoints according to the protocol supported by the
SIP Server:
UDP – Select this option to use UDP for signaling.
TCP – Select this option to use TCP for signaling.
TLS – The Signaling Host listens on secured port 5061 only and all outgoing
connections are established on secured connections. Calls from SIP clients or
servers to non secured ports are rejected.
The following protocols are supported: TLS 1.0, SSL 2.0 and SSL 3.0.
Note: If TLS is selected, the Skip Certificate Validation and the other
certificate related fields are displayed.
Skip Certificate Validation
When checked, no Certificate Validation is performed.
Revocation Method
Global Responder URL
Use Responder Specified
in Certificate
Allow Incomplete
Revocation Checks
For a detailed description, see Certificate Management and Certificate
Revocation.
Skip Certificate Validation
for OSCP Responder
SIP Servers: Primary Server
Server IP Address
Enter the IP address of the preferred SIP server.
If a DNS is used, you can enter the SIP server name.
Note: When in IPv4&IPv6 or in IPv6 mode, it is easier to use Names instead of
IP Addresses.
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Default IP Network Service – SIP Servers
Field
Description
Server Domain Name
Enter the name of the domain that you are using for conferences.
The domain name is used for identifying the SIP server in the appropriate
domain according to the host part in the dialed string.
For example, when a call to EQ1@polycom.com reaches its outbound proxy,
this proxy looks for the SIP server in the polycom.com domain, to which it will
forward the call.
When this call arrives at the SIP server in polycom.com, the server looks for
the registered user (EQ1) and forwards the call to this Entry Queue or
conference.
Port
Enter the number of the TCP or UDP port used for listening. The port number
must match the port number configured in the SIP server.
Default port is 5060.
Outbound Proxy Servers: Primary Server
Server IP Address
By default, the Outbound Proxy Server is the same as the SIP Server. If they
differ, modify the IP address of the Outbound Proxy and the listening port
number (if required).
Note: When in IPv4&IPv6 or in IPv6 mode, it is easier to use Names instead of
IP Addresses.
Port
Enter the port number the outbound proxy is listening to.
The default port is 5060.
When updating the parameters of the SIP Server in the IP Network Service - SIP Servers dialog
box, the Collaboration Server must be reset to implement the change.
16 Click the Security tab.
(This tab is only displayed if the SIP Server field in the SIP Servers tab is set to Specify.)
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17 Modify the following fields:
Default IP Network Service – Security (SIP Digest)
Field
Description
SIP Authentication
Click this check box to enable SIP proxy
authentication.
Select this check box only if the authentication is
enabled on the SIP proxy, to enable the Collaboration
Server to register with the SIP proxy. If the
authentication is enabled on the SIP proxy and
disabled on the RMX, calls will fail to connect to the
conferences.
Leave this check box cleared if the authentication
option is disabled on the SIP proxy.
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User Name
Enter the user name the Collaboration Server will use
to authenticate itself with the SIP proxy. This name
must be defined in the SIP Proxy.
Password
Enter the password the Collaboration Server will use
to authenticate itself with the SIP proxy. This
password must be defined in the SIP proxy.
These fields can
contain up to 20 ASCII
characters.
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IP Network Services
Default IP Network Service – Security (SIP Digest)
Field
Description
H.323 Authentication
Click this check box to enable H.323 server
authentication.
Select this check box only if the authentication is
enabled on the gatekeeper, to enable the
Collaboration Server to register with the gatekeeper. If
the authentication is enabled on the gatekeeper and
disabled on the RMX, calls will fail to connect to the
conferences.
Leave this check box cleared if the authentication
option is disabled on the gatekeeper.
User Name
Enter the user name the Collaboration Server will use
to authenticate itself with the gatekeeper. This name
must be defined in the gatekeeper.
Password
Enter the password the Collaboration Server will use
to authenticate itself with the gatekeeper. This
password must be defined in the gatekeeper.
These fields can
contain up to 64 ASCII
characters.
If the Authentication User Name and Authentication Password fields are left empty, the SIP
Digest authentication request is rejected. For registration without authentication, the Collaboration
Server must be registered as a trusted entity on the SIP server.
18 To configure the ICE environment, click the SIP Advanced tab.
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IP Network Services
19 Modify the following fields:
Default IP Network Service – SIP Advanced
Field
Description
ICE Environment
Select MS (for Microsoft ICE implementation) to enable the ICE integration.
Server User Name
Enter the Collaboration Server User name as defined in the Active Directory.
For example, enter rmx1234.
This field is disabled if the ICE Environment field is set to None.
20 To view or modify the V35 gateway parameters, click the V35 Gateway tab.
The V35 Gateway dialog box is displayed.
21 Modify the following fields:
Network Service - V35 tab
Field
Description
V35 Gateway IP
Address
Enter the Management IP address of the management interface of the Serial
Gateway. For more information see the RealPresence Collaboration Server
(RMX) 1800/2000/4000 Deployment Guide for Maximum Security Environments,
Serial Gateway S4GW - Maximum Security Mode.
Username
Enter the User Name that the Collaboration Server uses to log in to the
management interface of the Serial Gateway.
Password
Enter the Password that the Collaboration Server uses to log in to management
interface of the Serial Gateway.
22 Click the OK button.
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IP Network Services
When updating the parameters of the SIP Server in the IP Network Service - SIP Servers dialog
box, the Collaboration Server must be reset to implement the change.
Ethernet Settings
The automatically identified speed and transmit/receive mode of each LAN port used by the system can be
manually modified if a specific switch requires it. These settings can be modified in the Ethernet Settings
dialog box.
Physical Mapping - Port Type to Label on RealPresence Collaboration Server 4000
Port Type
Annotation on MCU
Media
LAN 2
Modem
LAN 1
Management 1
LAN 2
Signaling 1
LAN 3
ShM
LAN 6
RTM LAN Card
RTM-IP 4000 Card
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IP Network Services
To modify the automatic LAN port configuration:
1 On the Collaboration Server menu, select Setup > Ethernet Settings.
The Ethernet Settings dialog box specific to the system you are using is displayed; some examples
are shown below.
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IP Network Services
RealPresence Collaboration Server (RMX) 4000:
Although the RTM LAN (media card) ports are shown as Port 1 in the Ethernet Settings and
Hardware Monitor, the physical LAN connection is Port 2.
2 Select the speed and transmit/receive mode for each ports as follows:
Ethernet Settings Parameters
Field
Description
Speed
The Collaboration Server 1800 has 2 LAN ports that can be configured.
The Collaboration Server 2000/4000 has 3 LAN ports on the RTM-IP (Management,
Signaling and Shelf Management), and additional LAN ports on each media card (RTM
LAN) and RTM ISDN cards. You can set the speed and transmit/receive mode
manually for these ports.
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Port
The LAN port number.
Note: Do not change the automatic setting of Port 1,4 and Port 5 of
the Management 2 and Signaling 2 Networks. Any change to the
speed of these ports will not be applied.
Speed
Select the speed and transmit/receive mode for each port.
Default: Auto – Negotiation of speed and transmit/receive mode starts
at 1000 Mbits/second Full Duplex, proceeding downward to
10 Mbits/second Half Duplex.
Notes:
To maximize conferencing performance, especially in high bit rate call
environments, a 1Gb connection is recommended.
• RealPresence Collaboration Server (RMX) 4000 - Do not select
1000 Full Duplex for any LAN ports in Slot 17.
Select only 100 Full Duplex.
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IP Network Services
Ethernet Settings Parameters
Field
Description
802.1x
Authentication
For more information about 8.2.1x Authentication see IEEE 802.1X Authentication.
User Name
Password
3 Click OK.
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IP Network Services
IP Network Monitoring
The Signaling Monitor is the Collaboration Server entity used for monitoring the status of external network
entities such as the gatekeeper, DNS, SIP proxy and Outbound proxy and their interaction with the MCU.
To monitor signaling status:
1 In the RMX Management pane, click Signaling Monitor (
).
2 In the Signaling Monitor pane, double-click Default IP Service.
The IP Network Services Properties – RMX CS IP tab opens:
The RMX CS IP tab displays the following fields:
IP Network Services Properties – RMX CS IP Parameters
Field
Description
Service Name
The name assigned to the IP Network Service by the Fast Configuration
Wizard.
Note: This field is displayed in all tabs.
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IP Network Services Properties – RMX CS IP Parameters
Field
Description
IPv4
IP Address
IPv6
Default Router
IP Address
The IP address of the default router. The default router is
used whenever the defined static routers are not able to
route packets to their destination. The default router is also
used when host access is restricted to one default router.
Subnet Mask
The subnet mask of the MCU.
Default value: 255.255.255.0.
Scope
IP Address
Default Router
IP Address
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Global
The Global Unicast IP address of the
Collaboration Server.
Site-Local
The IP address of the Collaboration Server
within the local site or organization.
The IP address of the default router. The default router is
used whenever the defined static routers are not able to
route packets to their destination. The default router is also
used when host access is restricted to one default router.
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IP Network Services
3 Click the H.323 tab.
The H.323 tab displays the following fields:
IP Network Services Properties – H.323 Parameters
Field
Description
Connection State
The state of the connection between the Signaling Host and the gatekeeper:
Discovery - The Signaling Host is attempting to locate the gatekeeper.
Registration - The Signaling Host is in the process of registering with the
gatekeeper.
Registered - The Signaling Host is registered with the gatekeeper.
Not Registered - The registration of the Signaling Host with the gatekeeper
failed.
Registration Interval
The interval in seconds between the Signaling Host’s registration messages to
the gatekeeper. This value is taken from either the IP Network Service or from
the gatekeeper during registration. The lesser value of the two is chosen.
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Role
Active - The active gatekeeper.
Backup - The backup gatekeeper that can be used if
the connection to the preferred gatekeeper fails.
ID
The gatekeeper ID retrieved from the gatekeeper during
the registration process.
Name
The gatekeeper’s host’s name.
IP Address
The gatekeeper’s IP address.
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IP Network Services
4 Click the SIP Servers tab.
The SIP Servers tab displays the following fields:
IP Network Services Properties – SIP Servers Parameters
Field
Description
Role
Active -The default SIP Server is used for SIP traffic.
Backup -The SIP Server is used for SIP traffic if the preferred proxy fails.
Name
The name of the SIP Server.
IP Address
The SIP Server's IP address.
Status
The connection state between the SIP Server and the Signaling Host.
Not Available - No SIP server is available.
Auto - Gets information from DHCP, if used.
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5 Click the ICE Servers tab.
The ICE Servers tab displays the following fields:
IP Network Services Properties – ICE Servers Parameters
Field
Description
Role
The ICE Server’s role is displayed:
• STUN password server
• STUN Server UDP
• STUN Server TCP
• Relay Server UDP
• Relay Server TCP
IP Address
The ICE Server’s IP Address.
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IP Network Services
IP Network Services Properties – ICE Servers Parameters
Field
Description
Status 1/2/3/4
A status is displayed for each media card installed in the Collaboration Server:
• Connection O.K.
• MS – register fail
• MS – subscribe fail
• MS – service fail
• Connection failed
• User/password failed
• Channel didn't receive any packets for 5 seconds
• Channel exceeded allotted bandwidth
• Unknown failure
In systems with multiple media cards, Status 1 refers to the uppermost media
card.
FW Detection
The Firewall Detection status is displayed:
• Unknown
• UDP enabled
• TCP enabled
• Proxy -TCP is possible only through proxy
• Block – both UDP & TCP blocked
• None
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IP Network Services
LAN Redundancy
LAN Redundancy enables the redundant LAN port connection to automatically replace the failed LAN port
by using another physical connection and NIC (Network Interface Card). When a LAN port fails, IP network
traffic failure is averted and network or endpoints disconnections do not occur. When LAN cables are
connected to both LAN 1 and LAN 2 ports, the RMX automatically selects which port is active and which is
redundant.
Media Redundancy - Collaboration Server (RMX) 2000/4000
On the RealPresence Collaboration Server (RMX) 2000 and RealPresence Collaboration Server (RMX)
4000, the LAN 1 and LAN 2 port on the RTM LAN card can be used as redundant media ports.
RealPresence Collaboration Server (RMX) 2000/4000 RTM LAN Card on Rear Panel
Media Redundancy on the RealPresence Collaboration Server (RMX) 2000/4000 is dependent on the
settings of the LAN_REDUNDANCY and MULTIPLE_SERVICES System Flags as summarized in the
following table:
RMX 2000 / 4000 - Media Redundancy - System Flags
System Flag / Value
RMX 2000
RMX 4000
LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = NO
No Redundancy
LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = YES
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = NO
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = YES
Media Redundancy Only
Full Redundancy
Full Media Redundancy
(If only one IP Network Service is defined per media card.)
Media and Signaling Redundancy - Collaboration Server (RMX) 1800
On the RealPresence Collaboration Server 1800 LAN 1 and LAN 2 are the redundant media and signaling
ports:
● LAN 1 port is used for standard communications
● LAN 2 port can be used to define a second Network Service or for LAN Redundancy
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IP Network Services
The following cables are connected to the LAN ports on the rear panel of the RealPresence Collaboration
Server 1800:
LAN Connections to the IP ports
IP Port
Description
LAN 1
For management network connections:
When LAN redundancy is enabled, LAN 1 is used for management, media, and
signaling network connections.
LAN 2
For media, and signaling network connections:
When LAN redundancy is enabled, LAN 2 is the backup for the LAN 1 port.
Media Redundancy on the RealPresence Collaboration Server (RMX) 1800 is dependent on the settings of
the LAN_REDUNDANCY and MULTIPLE_SERVICES System Flags as summarized in the following table:
RMX 1800 - Media Redundancy - System Flags
System Flag / Value
RMX 1800
LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = NO
No Redundancy.
If a second LAN cable is connected to Port 2, Network separation is
enabled (the Management Network Service is separated from the Default
IP Network Service)
LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = YES
No Redundancy.
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = NO
Full Redundancy
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = YES
These flags cannot be set to YES simultaneously.
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IP Network Services
Signaling and Management Redundancy - Collaboration Server
(RMX) 4000
On the RealPresence Collaboration Server (RMX) 4000, for Signaling and Management Redundancy when
LAN_REDUNDANCY = YES and MULTIPLE_SERVICES = NO, the LAN 3 port on the RTM-IP 4000 card
is redundant to the LAN 2 port. LAN ports 4 and 5 are never used.
RealPresence Collaboration Server (RMX) 4000 - RTM IP 4000 on Rear Panel2
On the RealPresence Collaboration Server (RMX) 1800/4000 Signaling and Management Redundancy is
implemented using the LAN ports on the RTM-IP card and is dependent on the settings of the
LAN_REDUNDANCY and MULTIPLE_SERVICES System Flags as summarized in the following table.
RMX 4000 - Signaling and Management Redundancy - System Flags
Port Usage
Flag / Value
LAN 2
LAN 3
LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = NO
Management
Signaling
LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = YES
Management
Not Used
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = NO
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = YES
Management & Signaling
(LAN3 is redundant to LAN 2)
Management
Management
Management Redundancy - Collaboration Server (RMX) 1800
On the RealPresence Collaboration Server 1800, for Management Redundancy, the LAN 2 port is
redundant to the LAN 1 port and must have a LAN cable connected.
LAN Redundancy is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
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IP Network Services
RealPresence Collaboration Server 1800 - LAN 2 connection on Rear Panel
On the RealPresence Collaboration Server 1800, Management Redundancy is implemented using the LAN
1 and LAN 2 ports and is dependent on the settings of the LAN_REDUNDANCY and
MULTIPLE_SERVICES System Flags as summarized in the following table.
RMX 1800 - Management Redundancy - System Flags
Port Usage
Flag / Value
LAN 1
LAN 2
LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = NO
Management
Media and Signaling
LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = YES
Management
Media and Signaling
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = NO
Management, Media and Signaling.
LAN 2 is redundant to LAN 1
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = YES
These flags cannot be set to YES simultaneously.
Configuration Requirements
LAN Redundancy is disabled by default and is enabled by changing the LAN_REDUNDANCY system flag
to YES and connecting the appropriate LAN cables to the LAN ports on the Collaboration Server as follows:
RealPresence Collaboration Server 1800
» Connect the additional LAN cable to LAN 2 port on the rear panel of the Collaboration Server 1800.
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IP Network Services
RealPresence Collaboration Server (RMX) 2000
On a RealPresence Collaboration Server (RMX) 2000, one RTM LAN card is required.
For more information see the RealPresence Collaboration Server (RMX) 2000 Hardware Guide,
Installing or Replacing the RTM LAN.
● Connect the additional LAN cable to LAN 1 port on the RTM LAN.
● In the Setup> System Configuration > System Flags dialog box, add the flag
RMX2000_RTM_LAN and set it to YES to activate the installed RTM LAN card.
● On the RealPresence Collaboration Server (RMX) 2000, LAN Redundancy can be enabled
simultaneously with Multiple Networks. To enable the Multiple Networks option, set the
MULTIPLE_SERVICES flag to YES.
● A system reset is required when adding the RMX2000_RTM_LAN flag.
RealPresence Collaboration Server (RMX) 4000
● Connect the additional LAN cable to LAN 1 port on the RTM LAN.
● On the RealPresence Collaboration Server (RMX) 4000, LAN Redundancy can be enabled
simultaneously with Multiple Networks. To enable the Multiple Networks option, set the
MULTIPLE_SERVICES flag to YES.
● If required, reset the Collaboration Server.
On all systems:
● LAN Redundancy can be disabled by setting the LAN_REDUNDANCY System Flag to NO.
● If the LAN_REDUNDANCY System Flag value set to NO, the LAN 2 port must be connected to the
IP network.
On the RMX 2000/4000, full media redundancy is supported if only one IP Network Service is defined
per media card.
Hardware Monitor Indications
When LAN Redundancy is enabled on the Collaboration Server, LAN 2 port is Active. With LAN
redundancy, when LAN LEDs are lit they indicate that a physical connection of the cables is present but
does not indicate their activity status. In the Hardware Monitor pane the Lan List displays the Collaboration
Server LAN ports together with their Status indication.
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IP Network Services
The Hardware Monitor Status indications are summarized in the following table:
RTM LAN Indications
Status
Description
Active
The LAN port cable is connected.
Inactive
The LAN port cable is not connected.
Standby
The LAN Redundancy option is enabled and this LAN port is the redundant and in
standby mode. In case of failure, this port becomes active.
Network Traffic Control
The Network Traffic Control mechanism controls the level of UDP packets generated by the system. It
regulates a set of queuing systems and mechanisms by which UDP packets are received and transmitted
to the network router.
During a conference blast-out UDP packets can cause overloads on the network. MCU bandwidth usage
can increase to above the designated conference participant line rate settings, causing network bandwidth
issues such as latency and packet loss.
Three Network Traffic Control Flags are used to control the Network Traffic mechanism:
● ENABLE_TC_PACKAGE When the flag is set to NO (default), Network Traffic Control is disabled on the Collaboration Server.
Set the flag to YES to enable Network Traffic Control.
● TC_BURST_SIZE This flag regulates the Traffic Control buffer or max burst size as a percentage of the participant line
rate. In general, higher traffic rates require a larger buffer. For example, if the flag is set to 10 and the
participants line rate is 2MB, then the burst size is 200Kbps.
Default = 10
Flag range: 1-30.
● TC_LATENCY_SIZE This flag limits the latency (in milliseconds) or the number of bytes that can be present in a queue.
Default = 500
Flag range: 1-1000 (in milliseconds).
SIP Proxy Failover With Polycom® Distributed Media
Application™ (DMA™) 7000
Collaboration Server systems that are part of a RealPresence DMA system environment can benefit from
the RealPresence DMA system’s SIP Proxy Failover functionality.
SIP Proxy Failover is supported in the RealPresence DMA system’s Local Clustering mode with redundancy
achieved by configuring two DMA servers to share a single virtual IP address.
The virtual IP address is used by the Collaboration Server as the IP address of its SIP Proxy.
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IP Network Services
No additional configuration is needed on the Collaboration Server.
Should a SIP Proxy failure occur in one of the RealPresence DMA system servers:
● The other RealPresence DMA system server takes over as SIP Proxy.
● Ongoing calls may be disconnected.
● Previously ongoing calls will have to be re-connected using the original IP address, registration and
connection parameters.
● New calls will connect using the original IP address, registration and connection parameters.
RealPresence Collaboration Server (RMX) Network
Port Usage
The following table summarizes the port numbers and their usage in the <Product Name>:
Server Network Port Usage Summary
Connection
Type
Port
Number
Protocol
Description
Configurable
HTTP
80
TCP
Management between the
Collaboration Server and
Collaboration Server Web Client.
No
HTTPS
443
TCP
Secured Management between the
Collaboration Server and
Collaboration Server Web Client.
No
DNS
53
UDP
Domain name server.
Can be disabled in the IP
Network Service.
DHCP
68
UDP
Dynamic Host Configuration
Protocol.
Can be disabled in the IP
Network Service.
SSH
22
TCP
Secured shell. It is the Collaboration
Server terminal.
SSH is not supported when the
Collaboration Server is in Ultra
Secure Mode. For more information
see Ultra Secure Mode.
No
NTP
123
UDP
Network Time Protocol. Enables
access to a time server on the
network.
No
H.323 GK RAS
1719
UDP
Gatekeeper RAS messages traffic.
No
H.323 Q.931
1720 incoming;
49152-599
99 outgoing
TCP
H.323 Q.931 call signaling.
Each outgoing call has a separate
port.
The port for each outgoing call is
allocated dynamically.
Yes - for outgoing calls
only.
It is configured in the
Fixed Ports section of the
IP service.
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IP Network Services
Server Network Port Usage Summary
Connection
Type
Port
Number
H.323 H.245
Protocol
Description
Configurable
49152 59999
TCP
H.245 control.
Each outgoing call has a separate
port.
The port for each outgoing call is
allocated dynamically. It can be
avoided by tunneling.
Yes - for outgoing calls
only.
It is configured in the
Fixed Ports section of the
IP service.
SIP server
5060
60000
UDP, TCP
Connection to the SIP Server.
Sometimes port 60000 is used
when the system cannot reuse the
TCP port. This port can be set in the
Central signaling (CS) configuration
file.
Yes - in the IP service.
SIP Outbound
proxy
5060
60000
UDP, TCP
Connection to the SIP outbound
proxy.
Sometimes port 60000 is used
when the system cannot reuse the
TCP port. This port can be set in the
Central signaling (CS) configuration
file.
Yes - in the IP service.
SIP-TLS
60002
TCP
Required for Binary Floor Control
Protocol (BFCP) functionality for
SIP People+Content content
sharing.
No - port is not opened if
SIP People+Content is
disabled.
RTP
49152 59999
UDP
RTP media packets.
The ports are dynamically allocated.
Yes - It is configured in the
Fixed Ports section of the
IP service.
RTCP
49152 59999
UDP
RTP control.
The ports are dynamically allocated.
Yes - It is configured in the
Fixed Ports section of the
IP service.
SIP -TLS
5061
TCP
SIP -TLS for SIP server, and
outbound proxy .
No
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Defining ISDN/PSTN Network Services
To enable the RealPresence Collaboration Server (RMX) to function within ISDN/PSTN network
environments, network parameters must be defined for the ISDN/PSTN Network Service.
The configuration dialog boxes for both these network services are accessed from the RMX Management
pane of the RMX Web Client or RMX Manager.
Note: DSP Cards
RealPresence Collaboration Servers (RMX) 1800 support ISDN/PSTN only with 3 DSP cards
installed, and with built-in ISDN hardware.
RealPresence Collaboration Server (RMX) 1800-0 does not utilize DSP cards and does not support
ISDN/PSTN connections.
ISDN/PSTN Network Services Overview
To enable ISDN and PSTN participants to connect to the MCU, an ISDN/PSTN Network Service must be
defined. A maximum of two ISDN/PSTN Network Services, of the same Span Type (E1 or T1) can be
defined for the Collaboration Server. Each Network Service can attach spans from either or both cards.
Most of the parameters of the first ISDN/PSTN Network Service are configured in the Fast Configuration
Wizard, which runs automatically if an RTM ISDN card is detected in the Collaboration Server during first
time power-up. For more information, see Procedure 1: First-time Power-up in the Polycom RealPresence
Collaboration Server (RMX) 1800/2000/4000 Getting Started Guide.
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Defining ISDN/PSTN Network Services
Supported Capabilities and Conferencing Features:
● ISDN video is supported only in Continuous Presence (CP) conferences.
● Only BONDING (using multiple channels as a single, large bandwidth channel) is supported.
● Simple audio negotiation.
● Supported video resolutions are the same as for IP.
● Supported video Protocols are the same as for IP: H.261, H.263, H.264.
● H.239 for content sharing.
● Lecture Mode.
● DTMF codes.
● Securing of conferences.
● Basic cascading between two MCUs using an ISDN link is available and forwarding of DTMF codes
can be suppressed.
Non Supported Capabilities and Conferencing Features:
● NFAS (Non-Facility Associated Signaling)
● Leased line usage
● Restricted Channel mode
● Aggregation of channels
● E1 and T1 spans cannot operate simultaneously
● V.35 serial standards
● Primary and secondary clock source configuration (they are automatically selected by the system)
● Auto detection of Audio Only setting at endpoint
● Auto re-negotiation of bit rate
● Additional network services (two currently supported)
● Change of video mode (capabilities) from remote side during call
● Audio algorithms G.729 and G.723.1
● FECC
● H.243 Chair Control
● T.120 data sharing protocol
● H.261 Annex D
● MIH Cascading using an ISDN connection as cascade link
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Defining ISDN/PSTN Network Services
Adding/Modifying ISDN/PSTN Network Services
The system administrator can use the RMX Management – ISDN/PSTN Network Services section of the
Collaboration Server Web Client to add a second ISDN/PSTN Network Service or modify the first
ISDN/PSTN Network Service.
A new ISDN/PSTN Network Service can be defined even if no RTM ISDN card is installed in the
system.
Obtaining ISDN/PSTN required information
Before configuring the ISDN/PSTN Network Service, obtain the following information from your ISDN/PSTN
Service Provider:
● Switch Type
● Line Coding and Framing
● Numbering Plan
● Numbering Type
● Dial-in number range
If the Collaboration Server is connected to the public ISDN Network, an external CSU or similar
equipment is needed.
To Add an ISDN/PSTN Network Service:
1 In the RMX Management pane, click ISDN/PSTN Network Services (
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Defining ISDN/PSTN Network Services
2 In the ISDN/PSTN Network Services list menu, click New ISDN/PSTN Service ( ), or right-click
anywhere in the ISDN/PSTN Network Services list and select New ISDN/PSTN Service.
The Fast Configuration Wizard sequence begins with the ISDN/PSTN dialog box:
3 Define the following parameters:
ISDN Service Settings
Field
Description
Network Service Name
Specify the service provider’s (carrier) name or any other name you choose,
using up to 20 characters. The Network Service Name identifies the ISDN/PSTN
Service to the system.
Default name: ISDN/PSTN Service
Note: This field is displayed in all ISDN/PSTN Network Properties tabs and can
contain character sets that use Unicode encoding.
Span Type
Select the type of spans (ISDN/PSTN) lines, supplied by the service provider,
that are connected to the Collaboration Server. Each span can be defined as a
separate Network Service, or all the spans from the same carrier can be defined
as part of the same Network Service.
Select either:
• T1 (U.S. – 23 B channels + 1 D channel)
• E1 (Europe – 30 B channels + 1 D channel) Default: T1
Note: E1 and T1 spans cannot operate simultaneously.
Service Type
PRI is the only supported service type. It is automatically selected.
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Defining ISDN/PSTN Network Services
4 Click Next.
The PRI Settings dialog box is displayed:
5 Define the following parameters:
ISDN Service Settings
Field
Description
Default Num Type
Select the Default Num Type from the list.
The Num Type defines how the system handles the dialing digits. For example,
if you type eight dialing digits, the Num Type defines whether this number is
national or international.
If the PRI lines are connected to the Collaboration Server via a network switch,
the selection of the Num Type is used to route the call to a specific PRI line. If
you want the network to interpret the dialing digits for routing the call, select
Unknown.
Default: Unknown
Note: For E1 spans, this parameter is set by the system.
Num Plan
Select the type of signaling (Number Plan) from the list according to information
given by the service provider.
Default: ISDN
Note: For E1 spans, this parameter is set by the system.
Net Specific
Select the appropriate service program if one is used by your service provider
(carrier).
Some service providers may have several service programs that can be used.
Default: None
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ISDN Service Settings
Field
Description
Dial-out Prefix
Enter the prefix that the PBX requires to dial out. Leave this field blank if a
dial-out prefix is not required.
The field can contain be empty (blank) or a numeric value between 0 and 9999.
Default: Blank
6 Click Next.
The Span Definition dialog box is displayed:
7 Define the following parameters:
Span Definition
Field
Description
Framing
Select the Framing format used by the carrier for the network interface from the
list.
• For T1 spans, default is SFSF.
• For E1 spans, default is FEBE.
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Span Definition
Field
Description
Side
Select one of the following options:
• User side (default)
• Network side
• Symmetric side
Note: If the PBX is configured on the network side, then the Collaboration
Server unit must be configured as the user side, and vice versa, or both must be
configured symmetrically.
Line Coding
Select the PRI line coding method from the list.
• For T1 spans, default is B8ZS.
• For E1 spans, default is HDB3.
Switch Type
Select the brand and revision level of switch equipment installed in the service
provider’s central office.
• For T1 spans, default is AT&T 4ESS.
• For E1 spans, default is EURO ISDN.
Note: For T1 configurations in Taiwan, Framing must be set to ESF and Line
Coding to B8ZS.
8 Click Next.
The Phones dialog is displayed.
9 To define dial-in number ranges click the Add button.
10 The Add Phone Number dialog box opens.
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11 Define the following parameters:
Phones Settings
Field
Description
First Number
The first number in the phone number range.
Last Number
The last number in the phone number range.
•
•
A range must include at least two dial-in numbers.
A range cannot exceed 1000 numbers.
12 Click OK.
The new range is added to the Dial-in Phone Numbers table.
13 To define additional dial-in ranges, repeat steps 8 to 10.
14 Enter the MCU CLI (Calling Line Identification).
In a dial-in connections, the MCU CLI indicates the MCU’s number dialed by the participant. In a
dial-out connection, indicates the MCU (CLI) number as seen by the participant
15 Click Save & Continue.
After clicking Save & Continue, you cannot use the Back button to return to previous configuration
dialog boxes.
The ISDN/PSTN Network Service is created and confirmed.
16 Click OK to continue the configuration.
The Spans dialog box opens displaying the following read-only fields:
 ID – The connector on the ISDN/PSTN card (PRI1 - PRI12).
 Slot – The media card that the ISDN/PSTN card is connected to (1 or 2)
 Service – The Network Service to which the span is assigned, or blank if the span is not assigned
to a Network Service
 Clock Source – Indicates whether the span acts as a clock source, and if it does, whether it acts
as a Primary or Backup clock source. The first span to synchronize becomes the primary clock
source.
 State – The type of alarm: No alarm, yellow alarm or red alarm.
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17 Attach spans to existing Network Services, by marking the appropriate check boxes in the Attached
field.
Each ISDN/PSTN card can support 7 E1 or 9 T1 PRI lines.
18 Click Save & Close.
Modifying an ISDN/PSTN Network Service
To Modify an ISDN/PSTN Network Service:
1 In the RMX Management pane, click ISDN/PSTN Network Services (
).
2 In the ISDN/PSTN Network Services list, double-click the ISDN or right-click the ISDN entry and
select Properties.
The ISDN Properties dialog boxes are displayed. They are similar to the Fast Configuration
Wizard’s dialog boxes. For more information see Adding/Modifying ISDN/PSTN Network Services.
The following ISDN Properties can be modified:
 PRI Settings
 Net Specific
 Dial-out Prefix
 Span Definition
 Framing
 Side
 Line Coding
 Switch Type
 Phones
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 Dial-in Phone Numbers
 MCU CLI
 Spans
 Attached
All other ISDN Properties can only be modified by deleting the ISDN/PSTN network service and
creating a new PSTN service using the Fast Configuration Wizard. For more information, see
Adding/Modifying ISDN/PSTN Network Services.
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System security can be enhanced by separating the Media, Signaling and Management Networks.
Collaboration Server (RMX) 4000
On the Collaboration Server (RMX) 4000, Media, Signaling and Management Networks are physically
separated to provide enhanced security. The Default IP Network Service and the Management Network
Service have been logically and physically separated from each other. In the IP Network Service each IP
address is assigned a physical port and media (RTP) inputs are routed directly to a media card. This
provides for a more secure network with greater bandwidth as each media card has its own dedicated port.
All signaling communications are processed on a single stack of the processor in the Collaboration Server.
Collaboration Server 1800
On the RealPresence Collaboration Server 1800 the Default IP Network Service and the Management
Network Service have been logically and physically separated from each other to provide enhanced
security. The Collaboration Server 1800 includes two LAN ports that can be used for separating the
management network from the signaling and media network or if Multiple Networks option is enabled, each
LAN port is used for signaling, media and management per Network Service.
Collaboration Server (RMX) 2000
On the RealPresence Collaboration Server (RMX) 2000 a RTM LAN or RTM ISDN card is required to enable
the separation between the networks. By defining Multiple Network Services, a separate network can be
defined for each media card installed in the system.
For more information see Procedure 5: Enable Network Separation (RMX 2000) in the RealPresence
Collaboration Server (RMX) 1800/2000/4000 Deployment Guide for Maximum Security Environments.
Multiple Network Services
Media, signaling and management networks can be physically separated on the Collaboration Server
system to provide enhanced security. This addresses the requirement in an organization that different
groups of participants be supported on different networks. For example, some participants may be internal
to the organization while others are external.
Up to eight media and signaling networks can be defined for the Collaboration Server (RMX) 4000, or four
for the RealPresence Collaboration Server (RMX) 2000 and two for the Collaboration Server (RMX) 1800.
Multiple IP Network Services can be defined; up to two for each media and signaling network connected to
the Collaboration Server. The networks can be connected to one or several Media cards in the Collaboration
Server unit.
The Management Network is logically and physically separated from the media and signaling networks.
There can be one Management Network defined per Collaboration Server system.
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Each conference on the Collaboration Server can host participants from the different IP Network networks
simultaneously.
The following figure shows the network topology with three different media and signaling networks and one
Management network connected to the Collaboration Server (RMX) 4000.
Collaboration Server (RMX) 4000 - Multiple Networks Topology Sample
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The following figure shows the network topology with two different media and signaling networks and one
Management network connected to the Collaboration Server (RMX) 1800.
Collaboration Server (RMX) 1800 - Multiple Networks Topology Sample
Guidelines
● Multiple Services system mode is a purchasable option and it is enabled in the MCU license.
● Multiple Network Services are supported in MCUs with at least 1024MB memory only. MCU units with
memory of 512MB support only one IP Network Service.
● Multiple Services system mode is enabled when the system configuration flag
MULTIPLE_SERVICES is added and set to YES.
● Only IPv4 is supported for the definition of Multiple Network Services.
The MULTIPLE_SERVICE System Flag cannot be set to YES when IPv6 Addressing is enabled.
● When configuring Multiple Networks on the RealPresence Collaboration Server (RMX) 2000, the
RMX2000_RTM_LAN flag must be set to YES in addition to the MULTIPLE_NETWORKS=YES flag.
● Multiple Network Services are not supported with Microsoft ICE Environments in versions prior to
Version 7.8.
● Up to two Network Services, one per LAN port, can be associated with each Media card.
● An IP Network Service can be associated with one or several media cards. If more than one card is
associated with the same Network Service, the system routes the calls to the appropriate card
according to resource availability.
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● On Collaboration Servers (RMX) 2000/4000, RTM ISDN (card with 9 PRI / 4 LAN ports) or RTM LAN
cards can be used for Multiple Services configurations. When using the RTM ISDN card, Multiple
Services can not be configured in conjunction with LAN Redundancy.
● On the Collaboration Server (RMX) 1800, LAN redundancy cannot be enabled in parallel to Multiple
Networks and the LAN_REDUNDANCY flag must be set to NO when the Multiple Networks option
is enabled.
● Participants on different networks can connect to the same conference with full audio, video and
content capabilities.
● Traffic on one network does not influence or affect the traffic on other networks connected to the same
MCU, unless they are connected to the same media card. If one network fails, it will not affect the
traffic in the other connected networks, unless they are connected to the same media card and the
card fails.
● Maximum number of services that can be defined per Collaboration Server platform:
Maximum Number of Network Services per Collaboration Server System
Total Media
Cards
Network Services
(Up to 2 per Media Card)
1800
(Excluding
(RMX) 1800
with no DSP
cards)
3 (DSP Cards)
Up to 2
1
1
2
Up to 4 when using 2 RTM LAN
cards.
Less when using a combination
of RTM ISDN and RTM LAN, or
2 RTM ISDN cards.
1
2
4
Up to 8 when using 4 RTM LAN
cards.
Less when using up to 2 RTM
ISDN cards and the remaining
RTM LAN cards.
1
4
2000
4000
Management
Services
Network Services
that Include ICE
(1 / Media Card)
MCU
● From Version 7.8 onwards, a DNS server can be specified for each IP Network Service and for the
Collaboration Server Management Network Service.
 In the Network Services that do not include the DNS, use the IP addresses of the various devices
to define them in the Network Services.
● Participants are associated with a Network Service and use it resources as follows:
 Dial-in participants - according to the network used to place the call and connect to the
Collaboration Server.
 Dial-out participant - according to the Network Service selected during the participant properties
definition or during conference definition, according to the Network Service selected as default.
● Recording Links use the default Network Service to connect to conferences, therefore the recording
system must be defined on the default network to enable the recording.
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Resource Allocation and Capacity
The Resolution Configuration affect the resource capacity of the MCU. They are reflected in the port gauges
displayed on the Collaboration Server management application’s main screen. In Multiple Networks mode,
the overall resources are divided between the Network Services. However, the port gauges do not reflect
the resource availability per Network Service. For more information see Resource Capacity.
First Time Installation and Configuration
First Time Installation and Configuration of the RealPresence Collaboration Server (RMX) 1800/2000/4000
consists of the following procedures:
1 Preparations:
 Gather Network Equipment and Address Information Get the information needed for integrating
the Collaboration Server into the local network for each of the networks that will be connected to
the Collaboration Server.
 Unpack the Collaboration Server.
 Modify the Management Network parameters on the USB Key.
2 Hardware Installation and Setup:
 Mount the Collaboration Server in a rack.
 Connect the necessary cables.
3 First Entry Power-up and Configuration:
 Power up the Collaboration Server.
 Register the Collaboration Server.
 Connect to the Collaboration Server.
 Configure the Default IP Network Service.
 Configure the ISDN/PSTN Network Service.
 Modify the required System Flag to enable Multiple Services and reset the MCU.
4 Add the required IP Network Services to accommodate the networks connected to the Collaboration
Server.
5 Select a Network Service to act as default for dial out and gateway calls for which the Network
Service was not selected.
6 Place several calls and run conferences to ensure that the system is configured correctly.
For details see First Time Installation and Configuration in the Collaboration Server (RMX) 1800/2000/4000
Getting Started Guide.
Upgrading to Multiple Services
1 Gather Network Equipment and Address Information for each of the networks that will be connected
to the Collaboration Server unit. For a list of required address, see the Collaboration Server (RMX)
1800/2000/4000 Getting Started Guide, Gather Network Equipment and Address Information.
2 Upgrade to the new version and install the activation key that contains the Multiple Services license
as described in the RealPresence Collaboration Server (RMX) 1800/2000/4000 Release Notes. .
3 Place several calls and run conferences to ensure that the system upgrade was completed
successfully.
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4 Modify the required System Flag to enable Multiple Services, DO NOT reset the MCU yet.
5 Connect the additional network cables to the Collaboration Server and change existing connections
to match the required configuration as described in the appropriate RealPresence Collaboration
Server (RMX) Hardware Guide.
At this point, the Management Network can be modified to match the required local network settings
.
If the RealPresence Collaboration Server (RMX) 2000 you are upgrading does not include RTM
ISDN or RTM LAN cards, you must install at least one RTM LAN card to enable the definition of
multiple Network Services. If no RTM ISDN or RTM LAN cards are installed, the RealPresence
Collaboration Server (RMX) 2000 works in a single Network Service mode and an alarm is issued by
the system. For more details about the installation of RTM LAN cards, see the RealPresence
Collaboration Server (RMX) 2000 Hardware Guide.
6 Reset the MCU.
7 Connect to the MCU and add the required IP Network Services to accommodate the networks
connected to the Collaboration Server unit.
8 Select a Network Service to act as default for dial out and gateway calls for which the Network
Service was not selected.
9 Place several calls and run conferences to ensure that the system is configured correctly.
Gather Network Equipment and Address Information - IP Network
Services Required Information
It is important that before connecting multiple networks and implementing Multiple Services in the
Collaboration Server, that you obtain the information needed to complete the IP Network Service
configuration for each connected network from your network administrator.
Network Equipment and Address Information per IP Network Service
Parameter
Note
Signaling Host IP address
Media Board IP address (MPM 1)
Media Board IP address (MPM 2)
RealPresence Collaboration Server (RMX) 2000/4000 only
If more than one media card is
associated with this Network Service
Media Board IP address (MPM 3)
Collaboration Server (RMX) 4000 only
If more than one media card is
associated with this Network Service
Media Board IP address (MPM 4)
Collaboration Server (RMX) 4000 only
If more than one media card is
associated with this Network Service
Gatekeeper IP address (optional)
DNS IP address (optional)
Only one DNS can be defined for the
entire Network topology
SIP Server IP address (optional)
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RealPresence Collaboration Server (RMX) Hardware Installation
When connecting the LAN cables of the various networks to the Collaboration Server it is
recommended to use a color system to differentiate between the networks, for example, using
colored cables.
Collaboration Server (RMX) 4000 Multiple Services Configuration
Collaboration Server (RMX) 4000 Rear Panel with LAN and PRI cables shows the cables connected to the
Collaboration Server (RMX) 4000 rear panel, when one RTM ISDN and three RTM LAN cards are installed
providing IP and ISDN connectivity. The RTM ISDN card can be used for both ISDN and IP calls and only
one IP network Service is associated with each RTM/LAN card.
Collaboration Server (RMX) 4000 Rear Panel with LAN and PRI cables
In this case, up to four different IP Network Services can be defined - one for each RTM LAN/RTM ISDN
cards installed in the system.
If two LAN ports per each installed RTM LAN card are used, up to three additional Network Services can be
defined, bringing it to a total of up to 7 IP Network Services.
Several cards can be assigned to the same IP Network Service. The definition of the network services
attached to the Collaboration Server unit and which cards are assigned to each network service is defined
in the IP Network Service.
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Connecting the cables to the RTM IP 4000
The following cables are connected to the RTM IP on the rear panel of the Collaboration Server (RMX) 4000.
LAN Connections to the RTM IP
RTM IP Port
Description
LAN 1
Modem
LAN 2
Management
LAN 3
–
LAN 4
–
LAN 5
–
LAN 6
Shelf Management
Connecting the cables to the RTM LAN
LAN Connections to the RTM LAN
RTM LAN Port
Description
LAN 1
Signaling and Media - additional (second) Network Service
LAN 2
Signaling and Media - existing (first) Network Service
RealPresence Collaboration Server (RMX) 2000 Multiple Services Configuration
If one LAN port per RTM ISDN/ RTM LAN card is used, up to two different IP Network Services can be
defined - one for each installed RTM LAN/RTM ISDN cards.
If two LAN ports per each installed RTM LAN card are used, up to four Network Services can be defined.
RealPresence Collaboration Server (RMX) 2000 Rear Panel with RTM LAN Cables shows the cables
connected to the RealPresence Collaboration Server (RMX) 2000 rear panel, when two RTM LAN cards are
installed providing IP connectivity. In this case, only one IP network Service can be associated with each
RTM LAN card.
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RealPresence Collaboration Server (RMX) 2000 Rear Panel with RTM LAN Cables
Connecting the cables to the RTM IP
The following cables are connected to the RTM IP on the rear panel of the RealPresence Collaboration
Server (RMX) 2000.
LAN Connections to the RTM IP
RTM IP Port
Description
LAN 1
–
LAN 2
Management
LAN 3
Modem
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Connecting the cables to the RTM LAN
If RTM LAN or RTM ISDN cards are not installed on the Collaboration Server, they must be installed
before connecting the additional network cables for media and signaling.
LAN Connections to the RTM LAN
RTM IP Port
Description
LAN 1
Signaling and Media - second Network Service (optional)
LAN 2
Signaling and Media - first Network Service (optional)
Collaboration Server (RMX) 1800 Multiple Services Configuration
Up to two media and signaling networks can be defined for the Collaboration Server (RMX) 1800. Each of
these networks can be assigned a different IP Network Services (Multiple IP Network Services).
Connecting the cables to the RealPresence Collaboration Server 1800
Two LAN cables are connected to the LAN ports on the rear panel of the Collaboration Server 1800:
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LAN Connections to the IP ports
IP Port
Description
LAN 1
Management, media and signaling for Network Service 1.
LAN 2
Media and signaling for Network Service 2.
Collaboration Server Configuration
Once the network cables are connected to the Collaboration Server, you can modify the default IP Network
Service and add additional Network Services.
System Flags and License Settings
The MULTIPLE_SERVICES System Flag determines whether the Multiple Services option will be activated
once the appropriate license is installed. Possible Values: YES / NO Default: NO
This flag must be manually added to the system configuration and set to YES to enable this option. For more
information see .
If the MULTIPLE_SERVICES System Flag is set to YES and no RTM ISDN or RTM LAN card is
installed in the RealPresence Collaboration Server (RMX) 2000, an Active Alarm is displayed.
If the values or either of the MULTIPLE_SERVICES or V35_ULTRA_SECURED_SUPPORT
System Flags are changed from YES to NO, the defined IP Network Services are not displayed in
the IP Network Services list pane: they are, however, saved in the system.
If either of the flag values are changed back to YES, the saved defined IP Network Services will be
displayed.
IP Network Service Definition
Use this procedure to define Network Services in addition to the Network Service already defined during
first entry installation and configuration. Each of the defined Network Service can be associated with one or
more media cards installed in the system (depending on the system type).
Once a media card is associated with a Network Service it cannot be associated with another network
service.
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To add new/additional Network Services:
1 In the Device Management pane, click IP Network Services (
2 In the Network Services list toolbar, click the
).
Add Network Service button.
The New IP Service - Networking IP dialog box opens.
3 Define the following fields:
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IP Network Service - IP Parameters
Field
Description
Network Service Name
Enter the IP Network Service name.
Note: This field is displayed in all IP Signaling dialog boxes and can contain
character sets that use Unicode encoding.
IP Network Type
Select the IP Network environment. You can select:
• H.323 - For an H.323-only Network Service.
• SIP - For a SIP-only Network Service.
• H.323 & SIP - For an integrated IP Service. Both H.323 and SIP participants
can connect to the Collaboration Server using this service.
Note: This field is displayed in all Default IP Service tabs.
Signaling Host IP
Address
Enter the address to be used by IP endpoints when dialing into the
Collaboration Server using this Network Service.
Dial out calls of participants to whom this network service will be assigned are
initiated from this address.
This address is used to register the Collaboration Server with a Gatekeeper or a
SIP Proxy server residing on this network.
Media Card 1 Port 1 IP
Address
If only one network is connected to this media card, it is enough to assign one
media card to this Network Service. In such a case, enter one IP address for the
media card according to the LAN Port used for the connection.
If each of the LAN ports on one media card is used with two different networks,
each port is assigned to its own Network Service. In such a case, enter the IP
address of the port to be assigned to this Network Service.
A LAN port that is already assigned to a different Network Service, displays the
IP Address of the assigned port and it cannot be assigned to this Network
Service (it is disabled).
Media Card 1 Port 2 IP
Address 2
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IP Network Service - IP Parameters
Field
Description
Media Card 2 Port 1 IP
Address
Media Card 2 Port 2 IP
Address
If only one network is connected to this media card, it is enough to assign one
media card to this Network Service. In such a case, enter one IP address for the
media card according to the LAN Port used for the connection, as provided by
the network administrator.
If each of the LAN ports on one media card is used with two different networks,
each port is assigned to its own Network Service. In such a case, enter the IP
address of the port to be assigned to this Network Service.
Notes:
• LAN Ports/Media cards that are already associated with another Network
Service cannot be associated with this Network Service.
• You can define a Network Service without assigning media cards to it.
• To change the assignment of a card from one service to another, the card
must first be removed from the service to which it is assigned prior to its
assignment to another service.
RealPresence Collaboration Server (RMX) 2000: If one card was already
assigned to another service, only one additional card can be assigned to this
service.
Collaboration Server (RMX) 4000: Depending on the number of media cards
installed in the system, you can assign up to 4 media cards to this network
service provided that they are not assigned to any other Network Service.
(RealPresence
Collaboration Server
(RMX) 2000/4000)
Media Card 3 Port 1 IP
Address/
Media Card 3 Port 2 IP
Address
(Collaboration Server
(RMX) 4000)
Media Card 4 Port 1 IP
Address
Media Card 4 Port 2 IP
Address
(Collaboration Server
(RMX) 4000)
Subnet Mask
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Enter the subnet mask of the Collaboration Server in that network service.
Default value: 255.255.255.0.
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4 Optional. Some system flags can be defined per Network Service, depending on the network
environment.
To modify these flags, click the Service Configuration button.
The Service Configuration dialog box opens.
All the flags must be manually added to this dialog box. For a detailed description of the flags and
how to add them, see .
Flags defined per Network Service override their general definition in the System Configuration.
The following flags can be defined per service:
 ALLOW_NON_ENCRYPT_PARTY_IN_ENCRYPT_CONF
 ENABLE_H239
 SIP_ENABLE_FECC
 ENABLE_CLOSED_CAPTION
 ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF
 NUMERIC_CONF_ID_LEN
 NUMERIC_CONF_ID_MIN_LEN
 NUMERIC_CONF_ID_MAX_LEN
 ENABLE_CASCADED_LINK_TO_JOIN_WITHOUT_PASSWORD
 MAX_CP_RESOLUTION
 QOS_IP_AUDIO
 QOS_IP_VIDEO
 QOS_IP_SIGNALING
 ENABLE_CISCO_GK
 SIP_FREE_VIDEO_RESOURCES
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 FORCE_CIF_PORT_ALLOCATION
 MS_ENVIRONMENT
 SIP_FAST_UPDATE_INTERVAL_ENV
 SIP_FAST_UPDATE_INTERVAL_EP
 H263_ANNEX_T
 H239_FORCE_CAPABILITIES
 MIX_LINK_ENVIRONMENT
 IP_LINK_ENVIRONMENT
 FORCE_STATIC_MB_ENCODING
 FORCE_RESOLUTION
 SEND_WIDE_RES_TO_IP
 DISABLE_WIDE_RES_TO_SIP_DIAL_OUT
 SEND_SIP_BUSY_UPONRESOURCE_THRESHOLD
5 Click the Routers tab.
6 Define the routers used in this network and that are other than the routers defined in the
Management Network. The field definitions of the Routers tab are the same as for the Default
Management Network. For more information see Default Management Network Service – Routers.
7 Click the DNS tab.
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8 Modify the following fields:
Default Management Network Service – DNS
Field
Description
Service Host Name
Enter the host name of this network Service. Each Network Service must have a
unique Host Name otherwise an error message is displayed.
DNS
Select:
• Off – If no DNS server is used in this network.
• Specify – To enter the IP address of the DNS server used by this network
service.
Notes:
• The IP address field is enabled only if Specify is selected.
• In both Standard Security and Ultra Secure Modes:
• A DNS can be configured for the Management Network Service that is
defined and the IP Network Service.
• If a Multiple Services Licence is installed, a DNS can be configured for each
additional IP Network Service that is defined.If the DNS field in the IP
Network Service is set to Specify and the DNS is not configured or disabled,
the DNS configured for the Management Network will be used.
• When upgrading from a version that does not support a DNS per IP Network
Service, the DNS configured for the Management Network will be used.
Register Host Names
Automatically to DNS
Servers
Select this option to automatically register this Network Service Signaling
Host with the DNS server.
Local Domain Name
Enter the name of the domain for this network service.
DNS Server Address
Enter the static IP address of the DNS server that is part of this network.
9 Click the Gatekeeper tab.
10 Define the Primary and Alternate Gatekeepers and at least one Alias for this network Service. The
field definitions of the Gatekeeper tab are the same as for the Default IP Network Service. For more
information see Default IP Service – Conferencing – Gatekeeper Parameters.
In Multiple Services mode, an Alias must be defined for the specified gatekeeper.
11 To view or modify the port settings, click the Ports tab.
Settings in the Ports tab allow specific ports in the firewall to be allocated to multimedia conference
calls. If required, defined the ports to be used multimedia conference calls handled by this Network
Service. The field definitions of the Ports tab are the same as for the Default IP Network Service.
For more information see Default IP Service – Conferencing – Ports Parameters.
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12 If required, click the QoS tab.
The Collaboration Server’s implementation of QoS is defined per Network Service, not per endpoint.
The routers must support QoS in order for IP packets to get higher priority.
The field definitions of the QoS tab are the same as for the Default IP Network Service. For more
information see Default IP Service – Conferencing – QoS Parameters.
13 Click the SIP Servers tab.
14 Define the Primary SIP Server for this network Service.
•
•
•
Starting with Version 7.1, Registration of conferencing entities with the SIP Servers was moved
to the conferencing entities and is defined in the Conference Profile.
If Microsoft Office Communications or Lync server are part of this network service, a certificate
must be created for this network service. If each network connected to the Collaboration Server
includes Microsoft Office Communications or Lync server, separate certificates must be created
and sent to the Collaboration Server for each of these networks.
If the Network Service does not include a DNS, you must use the IP address of the SIP Server
instead of its name.
The field definitions of the SIP Servers tab are the same as for the Default IP Network Service. For
more information see Default IP Network Service – SIP Servers.
15 Click the Security tab.
The field definitions of the Security tab are the same as for the Default IP Network Service. For more
information see Default IP Network Service – Security (SIP Digest).
16 To configure the ICE environment, click the SIP Advanced tab.
17 Modify the following fields:
Default IP Network Service – SIP Advanced
Field
Description
ICE Environment
Select MS (for Microsoft ICE implementation) to enable the ICE integration.
For more information, see LAN Redundancy.
18 Click the OK button.
The new Network Service is added to the IP Network Services list pane.
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Setting a Network Service as Default
The default Network Service is used when no Network Service is selected for the following:
● Dial out participants
● Reserving resources for participants when starting an ongoing conference
● Gateway calls
In addition, the Signaling Host IP address and the MCU Prefix in GK displayed on the Collaboration Server
Web Client main screen are taken from the default H.323 Network Service.
One IP Network Service can be defined as default for H.323 connections and another Network Service as
default for SIP connections. If the IP Network Service supports both H.323 and SIP connections, you can
set the same Network Service as default for both H.323 and SIP, or for H.323-only or for SIP-only.
To designate an IP Network Service as the default IP Network Service:
1 In the Device Management pane, click IP Network Services (
).
2 In the Network Services list pane right-click the IP Network Service to be set as the default, and
then click Set As H.323 Default, or Set As SIP Default.
The next time you access this menu, a check mark is added next to the network service type to
indicate its selection as default.
To set this IP Network Service for both H.323 and SIP connections, repeat step 2 and select the
option you need.
The following icons are used to indicate the default IP Network Service type:
Default IP Network Service Icons
Icon
Description
This Network Service supports both SIP and H.323 connections and is
designated as default for both SIP and H.323 connections.
This Network Service supports both SIP and H.323 connections and is
designated as default for H.323 connections.
This Network Service supports both SIP and H.323 connections and is
designated as default for SIP connections.
This Network Service supports only H.323 connections and is set as default for
H.323 connections.
This Network Service supports only SIP connections and is set as default for
SIP connections.
Ethernet Settings
The Collaboration Server is set to automatically identify the speed and transmit/receive mode of each LAN
ports located on the RTM LAN or RTM ISDN cards that are added to the system. These port settings can
be manually configured if the specific switch requires it, via the Ethernet Settings dialog box.
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Signaling Host IP Address and MCU Prefix in GK Indications
The Collaboration Server Web Client displays the Signaling Host IP Address and MCU Prefix in GK
parameters as defined in the Default H.323 Network Service.
Conference Profile
Registration of conferencing entities such as ongoing conferences, Meeting Rooms, Entry Queues, SIP
Factories and Gateway Sessions with SIP servers is done per conferencing entity. This allows better control
on the number of entities that register with each SIP server by selecting for each of the conferencing entities
whether it will register with the SIP server.
The registration is defined in the Conference Profile - Network Services tab.
In the IP Network Services table, the system lists all the defined Network Services (one or several
depending on the system configuration).
● To register the conferencing entity to which this profile is assigned to a Network Service, in the
Registration column click the check box of that Network Service.
● You can also prevent dial in participants from connecting to that conferencing entities when
connecting via a Network Service.
In the Accept Calls column, clear the check box of the Network Service from which calls cannot
connect to the conference.
Gateway Profiles
To enable the Collaboration Server to call the destination endpoint/MCU via IP connection, the Network
Service for the call must be selected in the Gateway Profile dialog box.
The Network Service set as default is used if no other Network Service is selected.
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If the same Network Service is used for H.323 and SIP calls, the Network Service Environment must
include both H.323 and SIP settings.
Note: Gateway Functionality
Gateway functionality is not supported by Collaboration Server (RMX) 1800 with no DSP cards.
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Hardware Monitor
The Hardware Monitor pane includes the status of the hardware elements contained within the
Collaboration Server, among which are the LAN ports on the RTM LAN cards.
It is possible to reset the MPMRx media cards by selecting the MPMRx media card, and then selecting
Reset from the right-click menu.
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Signaling Monitor
The Signaling Monitor pane includes the list of the IP Network Services defined in the system - up to two
in the Collaboration Server (RMX) 1800/2000 and up to four in the Collaboration Server (RMX) 4000.
Double-clicking a Network Service, displays it properties and status.
Conferencing
Each conference on the Collaboration Server can host participants from the different IP Network networks
simultaneously.
Defining Dial Out Participants
When defining dial out participants, you can select the Network Service to place the call according to the
network to which the endpoint pertains. If the endpoint is located on a network other than the selected
network, the participant will not be able to connect.
If no Network is selected, the system uses the IP Network Service selected for reserving the conference
resources, and if none is set for the conference it uses the Network Service set as default.
The IP Network Service is selected in the New Participant - Advanced dialog box.
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Reserving Video Resources for a Conference
This section is applicable only for Collaboration Servers 2000/4000.
When defining a new ongoing conference or a conference reservation, you can select the Network Service
that will be used to reserve the required resources. If no Network Service is selected, the default Network
Service is used. Therefore, make sure that not all conferences are reserving resources from the same
Network Service, otherwise you may run out of resources for that Network Service.
The IP Network Service is selected in the New Conference/New Meeting Room/New Reservation General dialog box.
Monitoring Conferences
The Conference Properties - Network Services dialog box shows for each Network Service with which
Network Service’s SIP proxy the conference should be registered and if the dial in call will be connected to
the conference.
In the Participant pane, a new column - Service Name was added, indicating the name of Network Service
used for the participant’s connection.
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Resource Report
The Resource Report displays a graph of the MCU’s total resource usage as well as a graph of the MCU’s
total resource usage. When Multiple Network Services are active, an additional table of resource usage per
network service is displayed.
Video resource allocations are reported in AVC HD720p30 units. The same amounts of system resources
are allocated to Voice (Audio) participants, as those allocated to CIF Video participants.
The user can select a view of either resource usage Totals (default) or resource usage per Network Service.
For Collaboration Servers 1800 and 2000/4000 with MPMRx media cards there is no differentiation
between Video and Voice (Audio) resource usage.
Resource Report - Collaboration Servers 1800 and 2000/4000 with MPMRx Cards
Port Usage Gauges
Collaboration Servers 1800 and 2000/4000 with MPMRx media cards do not differentiate between Video
and Voice (Audio) resource usage and display a single port usage gauge.
The Port Gauge(s) show the total resource usage for the Collaboration Server and not per Network Service.
So it may not be an accurate representation of the availability of resources for conferencing, as one Network
Service may run out of available resources while another Network Service may have all of it resources
available. In such a case, the port gauge(s) may show that half of the system resources are available for
conferencing, while calls via the Network Service with no available resources will fail to connect.
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Port Usage Gauge - Collaboration Servers 2000/4000 with MPMRx Cards
Port Usage Gauge - Collaboration Server 1800
Port Usage Gauge - Collaboration Server 1800
NAT (Network Address Translation) Traversal
NAT Traversal is a set of techniques enabling participants behind firewalls to connect to conferences,
hosted on the Collaboration Server, remotely using the internet.
All signaling and media for both SIP and H.323 will be routed through an SBC. The following SBC
environments are supported:
● SAM - a Polycom SBC
● Acme Packet - a 3rd party SBC
● VBP - Polycom Video Border Proxy
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Deployment Architectures
The following NAT Traversal topologies are given as examples. Actual deployments will depend on user
requirements and available infrastructure:
Remote Connection Using the Internet
The following Remote Connection call flow options are supported:
Remote Connections
Enterprise Client
Environment
Registered
SBC
SIP / H.323
Yes
SAM / Acme Packet
SIP / H.323
No
SAM / Acme Packet
SIP / H.323
No
SAM Only
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Business to Business Connections
The following Business to Business connection call flow options are supported:
Business to Business Connections
Enterprise A Client
Enterprise B Client
Environment
Registered
SBC
SBC
Registered
Environment
H.323
Yes
RealPresence
Access
Director

RealPresence
Access
Director
Yes
H.323
H.323
Yes

VBP
Yes
H.323
SIP
Yes
RealPresence
Access
Director

RealPresence
Access
Director
Yes
H.323
SIP
Yes
Acme Packet

Acme Packet
Yes
H.323
FW (Firewall) NAT Keep Alive
The Collaboration Server can be configured to send a FW NAT keep alive message at specific Intervals for
the RTP, UDP and BFCP channels.
This is necessary because port mappings in the firewall are kept open only if there is network traffic in both
directions. The firewall will only allow UDP packets into the network through ports that have been used to
send packets out.
By default the Collaboration Server sends a FW NAT Keep Alive message every 30 seconds. As there is no
traffic on the Content and FECC channels as a call begins, the firewall will not allow any incoming packets
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from the Content and FECC channels in until the Collaboration Server sends out the first of the FW NAT
Keep Alive messages 30 seconds after the call starts.
If Content or FECC are required within the first 30 seconds of a call the FW NAT Keep Alive Interval should
be modified to a lower value.
To enable and modify FW NAT Keep Alive:
FW NAT Keep Alive is enabled in the New Profile - Advanced dialog box.
» Select the FW NAT Keep Alive check box and if required, modify the Interval field within the range
of 5 - 86400 seconds.
System Configuration in SBC environments
In an environment that includes SAM (a Polycom SBC), to ensure that a RealPresence Mobile endpoint can
send content to a conference the value of the system flag
NUM_OF_INITIATE_HELLO_MESSAGE_IN_CALL_ESTABLISHMENT must be set to at least 3.
For more details on modifying the values of system flags, see Modifying System Flags.
SIP TCP Keep-Alive
SIP TCP Keep Alive behavior is defined for each IP Network Service and can be modified by adding the
following System Flags and modifying their values:
● SIP_TCP_KEEP_ALIVE_TYPE
● SIP_TCP_KEEP_ALIVE_BEHAVIOR
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IVR Services
Interactive Voice Response (IVR) is an application that allows participants to communicate with the
conferencing system via their endpoint’s input device (such as a remote control). The IVR Service includes
a set of voice prompts and a video slide used to automate the participants connection to a conference or
Entry Queue. It allows customization of menu driven scripts and voice prompts to meet different needs and
languages.
The IVR module includes two types of services:
● Conference IVR Service that is used with conferences
● Entry Queue IVR Service that is used with Entry Queues
The system is shipped with two default Conference IVR Services (one for the conferences and the other for
gateway calls) and one default Entry Queue IVR Service. The default services include voice messages and
video slides in English.
To customize the IVR messages and video slide perform the following operations:
● Record the required voice messages and create a new video slide.
For more information, see Creating a Welcome Video Slide.
● Optional. Add the language to the list of languages supported by the system.
● Upload the voice messages to the MCU (This can be done as part of the language definition or during
the IVR Service definition).
● Create the Conference IVR Service and upload the video slide, and if required any additional voice
messages.
● Optional. Create the Entry Queue IVR Service and upload the required video slide and voice
messages.
When upgrading the Collaboration Server software version new DTMF Codes and voice messages
are not automatically added to existing IVR Services in order to avoid conflicts with existing DTMF
codes. Therefore, to use new options, new Conference and Entry Queue IVR Services must be
created.
IVR Services List
You can view the currently defined Conference IVR and Entry Queue IVR Services in the IVR Services list
pane.
To view the IVR Services list:
1 In the Collaboration Server Management pane, expand the Rarely Used list.
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2 Click the IVR Services (
) entry.
The list pane displays the Conference IVR Services list and the total number of IVR services currently
defined in the system.
IVR Services Toolbar
The IVR Services toolbar provides quick access to the IVR Service definitions as follows:
IVR Toolbar buttons
Button
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Button Name
Descriptions
New Conference IVR Service
Creates a new Conference IVR Service.
New Entry Queue IVR Service
Creates a new Entry Queue IVR Service.
Delete Service
Deletes the selected IVR service(s).
Set Default Conference IVR
Service
Sets the selected Conference IVR Service as default. When
creating a new conference Profile the default IVR Service is
automatically selected for the Profile (but can be modified).
Set Default Entry Queue Service
Sets the selected Entry Queue IVR Service as default. When
creating a new Entry Queue the default Entry Queue IVR
Service is automatically selected.
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IVR Toolbar buttons
Button
Button Name
Descriptions
Add Supported
Languages
Adds languages to the IVR module, enabling you to download
voice prompts and messages for various languages.
Replace/Change Music File
To replace the currently loaded music file that is used to play
background music, the MCU is shipped with a default music file.
Adding Languages
You can define different sets of audio prompts in different languages, allowing the participants to hear the
messages in their preferred language.
The Collaboration Server is shipped with a default language (English) and all the prompts and messages
required for the default IVR Services, conference and Entry Queues shipped with the system.
You can add languages to the list of languages for which different messages are downloaded to the MCU
and IVR Services are created. This step is required before the creation of additional IVR messages using
languages that are different from English, or if you want to download additional voice files to existing files in
one operation and not during the IVR service definition.
To add a language:
1 In the Collaboration Server Management pane, expand the Rarely Used list.
2 Click the IVR Services (
) entry.
3 In the Conference IVR Services list, click the Add Supported Languages (
) button.
The Supported Languages dialog box opens.
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4 Click the Add Language button.
The New Language dialog box opens.
5 In the New Language dialog box, enter the name of the new language. The language name can be
typed in Unicode and cannot start with a digit. Maximum field length is 31 characters.
6 Click OK.
The new language is added to the list of Supported Languages.
Uploading a Message File to the Collaboration Server
You can upload audio files for the new language or additional files for an existing language now, or you can
do it during the definition of the IVR Service. In the latter case, you can skip the next steps.
•
•
Voice messages should not exceed 3 minutes.
It is not recommended to upload more than 1000 audio files to the MCU memory.
To upload messages to the MCU:
1 To upload the files to the MCU, in the Supported Languages dialog box, click the Add Message File
button.
The Add Message File dialog box opens.
Audio files are uploaded to the MCU one-by-one.
2 In the IVR Message Language list, select the language for which the audio file will be uploaded to
the MCU.
3 In the IVR Message Category list, select the category for which the audio file is uploaded.
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4 In the Message Type list, select the message type for which the uploaded message is to be played.
You can upload several audio files for each Message Type. Each file is downloaded separately.
The table below lists the Message Types for each category:
IVR Message Types by Message Category
Message
Category
Message Type
Message
Conference
Password
Request Conference
Password
Requests the participant to enter the conference password.
Request Conference
Password Retry
A participant who enters an incorrect password is requested
to enter it again.
Request Digit
Requests the participant to enter any digit in order to connect
to the conference. Used for dial-out participants to avoid
answering machines in the conference.
Welcome
Message
Welcome Message
The first message played when the participant connects to
the conference or Entry Queue.
Conference
Chairperson
Request Chairperson
Identifier
Requests the participants to enter the chairperson identifier
key.
Request Chairperson
Password
Requests the participant to enter the chairperson password.
Request Chairperson
Password Retry
When the participant enters an incorrect chairperson
password, requests the participant to enter it again.
General
Messages played for system related event notifications, for example, notification that the
conference is locked. Upload the files for the voice messages that are played when an
event occurs during the conference. For more information, see Conference IVR Service
Properties - General Voice Messages.
Billing Code
Requests the chairperson to enter the conference Billing Code.
Roll Call
Roll call related messages, such as the message played when a participant joins the
conference. Messages are listed in the Conference IVR Service - Roll Call dialog box.
Conference ID
Requests the participant to enter the required Conference ID to be routed to the
destination conference.
Conference / Chairperson Password Interaction for Dial-in Participants
Password
System Behavior
Conference
Chairperson
No
No
All endpoints connect without any IVR prompt.
Yes
No
The MCU issues an IVR prompt for the conference password. All endpoints
must respond with the conference password. The MCU will prompt three
times before non-responding endpoints are disconnected.
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Conference / Chairperson Password Interaction for Dial-in Participants
Password
System Behavior
Conference
Chairperson
Yes
Yes
The MCU issues an IVR prompt for the conference password. All endpoints
must respond with either the conference password or the chairperson
password. The MCU will prompt three times before non-responding
endpoints are disconnected.
No
Yes
The MCU issues an IVR prompt requesting the endpoint wanting to be
chairperson to respond with “#”.
When the endpoint wanting to be chairperson responds with “#”, the MCU
issues an IVR prompt requesting the chairperson password.
If no password is manually assigned to the conference, the MCU generates a random conference password.
5 Click Upload File to upload the appropriate audio file to the MCU.
The Install File dialog box opens.
6 Enter the file name or click the Browse button to select the audio file to upload.
The Select Source File dialog box opens.
7 Select the appropriate *.wav audio file, and then click the Open button.
The name of the selected file is displayed in the Install field in the Install File dialog box.
8 Optional. You can play a .wav file by selecting the Play button (
).
9 Click Yes to upload the file to the MCU.
The system returns to the Add Message File dialog box.
10 Repeat step 6 for each additional audio file to be uploaded to the MCU.
11 Once all the audio files are uploaded to the MCU, close the Add Message File dialog box and
return to the Add Language dialog box.
12 Click OK.
Defining a New Conference IVR Service
The Collaboration Server is shipped with two default Conference IVR Services and all its audio messages
and video slide. You can define new Conference IVR Services or modify the default Conference IVR
Service. For the definition of Conference IVR Service for gateway calls, see Defining the IVR Service for
Gateway Calls.
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Up to 80 IVR Services (Conference IVR Services and Entry Queue IVR Services) can be defined
per Collaboration Server.
Defining a New Conference IVR Service
To define a new Conference IVR Service:
1 On the IVR Services toolbar, click the New Conference IVR Service (
) button.
The New Conference IVR Service - Global dialog box opens.
2 Define the following parameters:
Conference IVR Service Properties - Global Parameters
Field/Option
Description
Conference IVR Service
Name
Enter the name of the Conference IVR Service. The maximum field length is 20
characters and may be typed in Unicode.
Language For IVR
Select the language of the audio messages and prompts from the list of
languages defined in the Supported languages. The default language is
English. For more information, see Adding Languages.
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Conference IVR Service Properties - Global Parameters
Field/Option
Description
External Server
Authentication
This option is not supported with Collaboration Server 1800.
You can configure the IVR Service to use an external database application to
verify a participant’s right to join the conference. For more information, see
Conference Access with External Database Authentication.
Select one of the following options:
• Never – The participant’s right to join the conference will not be verified with
an external database application (default).
• Always – Any participant request to join the conference is validated with the
external database application using a password.
• Upon Request – Only the participant request to join the conference as
chairperson is validated with the external database application using a
password. The validation process occurs only when the participant enters the
chairperson identifier key.
Number of User Input
Retries
Enter the number of times the participant will be able to respond to each menu
prompt before being disconnected from the conference. Range is between 1-4,
and the default is 3.
Timeout for User Input
(Sec)
Enter the duration in seconds that the system will wait for the participant’s input
before prompting for another input. Range is between 1-10, and the default
value is 5 seconds.
DTMF Delimiter
Enter the key that indicates the last input key. Possible values are the pound (#)
and star (*) keys. The default is #.
3 Click the Welcome tab.
The New Conference IVR Service - Welcome dialog box opens.
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4 Select the Enable Welcome Messages check box to define the system behavior when the
participant enters the Conference IVR queue. When participants access a conference through an
Entry Queue, they hear messages included in both the Entry Queue Service and Conference IVR
Service. To avoid playing the Welcome Message twice, disable the Welcome Message in the
Conference IVR Service.
5 Select the General Welcome Message, to be played when the participant enters the conference
IVR queue.
6 To upload an audio file for an IVR message, click Add Message File.
The Install File dialog box opens.
The Collaboration Server unit is bundled with default audio IVR message files. To upload a
customized audio file, see Creating Audio Prompts and Video Slides.
a Click the Browse button to select the audio file (*.wav) to upload.
The Select Source File dialog box opens.
b Select the appropriate *.wav audio file and then click the Open button.
c Optional. You can play a .wav file by selecting the Play button (
).
d In the Install File dialog box, click Yes to upload the file to the MCU memory.
The Done dialog box opens.
e Once the upload is complete, click OK and return to the IVR dialog box. The new audio file can
now be selected from the list of audio messages.
7 Click the Conference Chairperson tab.
The New Conference IVR Service - Conference Chairperson dialog box opens.
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8 Select the Enable Chairperson Messages check box to enable the chairperson functionality. If this
feature is disabled, participants are not able to connect as the chairperson.
When both Conference Password and Chairperson Password options are enabled and defined, the
system first plays the prompt Enter conference password. However, if the participant enters
the chairperson password, the participant becomes the chairperson.
To play the prompt requesting the Chairperson password, For conference chairperson
services..., do not select the Enable Password Messages option.
9 Select the various voice messages and options for the chairperson connection.
If the files were not uploaded prior to the definition of the IVR Service or if you want to add new audio
files, click Add Message File to upload the appropriate audio file to the Collaboration Server.
New Conference IVR Service Properties - Conference Chairperson Options and Messages
Field/Option
Description
Chairperson Identifier
Request
Select the audio file that requests the participants to enter the key that identifies
them as the conference chairperson.
Request Chairperson
Password
Select the audio file that prompts the participant for the chairperson password.
Retry Chairperson
Password
Select the audio file that prompts participants to re-enter the chairperson
password if they enter it incorrectly.
Chairperson Identifier
Key
Enter the key to be used for identifying the participant as a chairperson.
Possible keys are: pound key (#) or star (*).
Billing Code
The prompt requesting the chairperson billing code selected in the General tab.
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10 Click the Conference Password tab.
The New Conference IVR Service - Conference Password dialog box opens.
11 Select the Enable Password Messages check box to request the conference password before
moving the participant from the conference IVR queue to the conference.
When both Conference Password and Chairperson Password are enabled and defined, the system
first plays the prompt Enter conference password. However, if the participant enters the
chairperson password, the participant becomes the chairperson.
To play the prompt requesting the Chairperson password, For conference chairperson
services..., do not select the Enable Password Messages option.
12 Select the MCU behavior for password request for Dial-in and Dial-out participant connections.
Select the required system behavior as follows:
 Request password - The system requests the participant to enter the conference password.
 None - The participant is moved to the conference without any password request.
 Request Digit - The system requests the participant to enter any key. This option is used mainly
for dial-out participants and to prevent an answering machine from entering the conference.
13 Select the various audio messages that will be played in each case.
New Conference IVR Service Properties - Conference Password Parameters
Option
Description
Request Password
Select the audio file that prompts the participant for the conference password.
Retry Password
Select the audio file that requests the participant to enter the conference
password again when failing to enter the correct password.
Request Digit
Select the audio file that prompts the participant to press any key when the
Request Digit option is selected.
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14 Click the General tab.
The New Conference IVR Service - General dialog box opens.
The General dialog box lists messages that are played during the conference. These messages are
played when participants or the conference chairperson perform various operations or when a
change occurs.
15 To assign the appropriate audio file to the message type, click the appropriate table entry, in the
Message File column. A drop-down list is enabled.
16 From the list, select the audio file to be assigned to the event/indication.
17 Repeat steps 15 and 16 to select the audio files for the required messages.
The following types of messages and prompts can be enabled:
Conference IVR Service Properties - General Voice Messages
Message Type
Description
Blip on Cascade Link
Indicates that the link to the cascaded conference connected successfully.
Chairperson Exit
Informs all the conference participants that the chairperson has left the
conference, causing the conference to automatically terminate after a short
interval.
Note: This message is played only when the Requires Chairperson option is
selected in the Conference Profile - IVR dialog box.
Chairperson Help Menu
A voice menu is played upon a request from the chairperson, listing the
operations and their respective DTMF codes that can be performed by the
chairperson. The playback can be stopped any time.
Note: If you modify the default DTMF codes used to perform various operations,
the default voice files for the help menus must be replaced.
Change Chairperson
Password
Requests the participant to enter a new chairperson password when the
participant is attempting to modify the chairperson password.
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Conference IVR Service Properties - General Voice Messages
Message Type
Description
Change Conference
Password
Requests the participant to enter a new conference password when the
participant is attempting to modify the conference password.
Change Password
Failure
A message played when the participant enters an invalid password, for example
when a password is already in use.
Change Passwords
Menu
This voice menu is played when the participants requests to change the
conference password. This message details the steps required to complete the
procedure.
Conference is Locked
This message is played to participants attempting to join a Secured conference.
Conference is Secured
This message is played when the conference status changes to Secure as
initiated by the conference chairperson or participant (using DTMF code *71).
Conference is unsecured
This message is played when the conference status changes to Unsecured as
initiated by the conference chairperson or participant (using DTMF code #71).
Confirm Password
Change
Requests the participant to re-enter the new password.
Dial Tone
The tone that will be played to indicate a dialing tone, to let the calling participant
enter the destination number.
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.
Disconnect on Busy
The Busy Tone is played when the system retries to redial a busy destination
number and fails after exceeding the number of redials. This call is then
disconnected.
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.
Disconnect on No
Answer
The Reorder Tone is played when the system retries to redial a destination
number that does not answer and fails after exceeding the number of redials.
This call is then disconnected.
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.
Disconnect on Wrong
Number
A voice message is played when the call fails because of an incorrect
destination number. The message is followed the Reorder Tone and the call is
disconnected.
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.
End Time Alert
Indicates that the conference is about to end.
Enter Destination ID
Prompts the calling participant for the destination number. Default message
prompts the participant for the conference ID (same message as in the Entry
Queue IVR Service).
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.
First to Join
Informs the participant that he or she is the first person to join the conference.
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Conference IVR Service Properties - General Voice Messages
Message Type
Description
Incorrect Destination ID
If the participant entered an incorrect conference ID (in gateway calls it is the
destination number), requests the participant to enter the number again.
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.
Maximum Number of
Participants Exceeded
Indicates the participant cannot join the destination conference as the maximum
allowed number of participants will be exceeded.
Mute All Off
This message is played to the conference to inform all participants that they are
unmuted (when Mute All is cancelled).
Mute All On
Informs all participants that they are muted, with the exception of the conference
chairperson.
Note: This message is played only when the Mute All Except Me option is
activated.
No Video Resources
Audio Only.
Informs the participant of the lack of Video Resources in the Collaboration
Server and that he/she is being connected as Audio Only.
Participant Help Menu
A voice menu that is played upon request from a participant, listing the
operations and their DTMF codes that can be performed by any participant.
Password Changed
Successfully
A message is played when the password was successfully changed.
Recording Failed
This message is played when the conference recording initiated by the
chairperson or the participant (depending on the configuration) fails to start.
Recording in Progress
This message is played to participant joining a conference that is being recorded
indicating the recording status of the conference.
Redial on Wrong Number
A message is played requesting the participant to enter a new destination
number followed by up to five redial attempts. If all redial attempts fail, the
participant is alerted by an IVR message that the dialed number is unreachable,
followed by the Reorder Tone and disconnection.
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.
Request Billing Code
Requests the participant to enter a code for billing purposes.
Requires Chairperson
The message is played when the conference is on hold and the chairperson
joins the conference. For this message to be played the Conference Requires
Chairperson option must be selected in the Conference Profile - IVR dialog
box.
Ringing Tone
The tone that will be played to indicate that the system is calling the destination
number.
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.
Self Mute
A confirmation message that is played when participants request to mute their
line.
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Conference IVR Service Properties - General Voice Messages
Message Type
Description
Self Unmute
A confirmation message that is played when participants request to unmute their
line.
18 Click the Roll Call/Notifications tab.
The New Conference IVR Service - Roll Call dialog box opens.
The Roll Call and Tone Notification options are disabled in SVC and mixed CP and SVC
conferences.
The Roll Call feature of the Conference IVR Service is used to record the participants’ names for
playback when the participants join and leave a conference.
Roll Call announcements played upon a participant’s connection or disconnection from a conference
(Entry and Exit announcements) can be replaced by tones. These tones can be used as notification
when participants join or leave the conference but the identification of the participant is not required.
The system is shipped with two default tones: Entry Tone and Exit tone. When the Tone Notifications
option is enabled, no recording of the participant names will occur and the conference chairperson
will not be able to ask for a name review during the conference.
In Collaboration Server 2000/4000, the selection of tones in the IVR Service definition replaces the
functionality of the system flag IVR_ROLL_CALL_USE_TONES_INSTEAD_OF_VOICE.
19 Select one of the following options to determine the announcement mode:
a To enable the Roll Call feature, select the Enable Roll Call option.
b Select Enable Tones to enable the Tone Notifications option.
The dialog box changes to display the tone notification options.
c Select None to disable the Roll Call and Tone Notifications features.
20 If you enable the Roll Call option, assign the audio file to the message type in the Message File
column, and click the appropriate table entry.
An arrow appears in the Message File column.
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If the Roll Call option is enabled, you must assign the appropriate audio files to all message types.
21 Click the arrow to open the Message File list and select the appropriate audio file.
Conference IVR Service Properties - Roll Call Messages
Roll Call Message
Description
Roll Call Record
Requests participants to state their name for recording, when they connect to
the conference.
Note: The recording is automatically terminated after two seconds.
Roll Call Joined
A voice message stating that the participant has joined the conference.
Note: In Collaboration Server 2000/4000, when the system flag
IVR_ROLL_CALL_USE_TONES_INSTEAD_OF_VOICE is set to YES, the
system does not playback the Roll Call names when participants enter the
conference. However, the voice message will be played, unless it is replaced
with tone file. In such a case, the use of tones requires the uploading of the
appropriate tone files in *.wav format and replacing the Roll Call Joined
message file with the tone file.
Roll Call Left
A voice message stating that the participant has left the conference.
Note: In Collaboration Server 2000/4000, when the system flag
IVR_ROLL_CALL_USE_TONES_INSTEAD_OF_VOICE is set to YES, the
system does not playback the Roll Call names when participants exit the
conference. However, the voice message will be played, unless it is replaced
with tone file. In such a case, the use of tones requires the uploading of the
appropriate tone files in *.wav format and replacing the Roll Call Left message
file with the tone file.
Roll Call Review
Played when Roll Call is requested by the chairperson, introducing the names of
the conference participants in the order they joined the conference.
Once a conference is enabled for Roll Call, Roll Call can be disabled using the #42 or enabled using
the * 42 DTMF Codes. or enabled with all connected participants in the conference receiving the
<name> has joined/left the conference announcement.
For all connected participants in the conference while the Roll Call is disabled (#42)
 Joining participants are requested to say their name.
 The conference is not interrupted with join/left announcements.
 Roll call review is possible.
For all connected participants in the conference while the roll call is enabled (*42)
 Joining participant are requested to say their name.
 The conference is interrupted with join/left prompts.
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 Roll call review is possible.
If Roll Call is enabled, when a participant joins or leaves a conference, the system plays a voice
message to all participants. The audio message consists of the joining or leaving participant’s Roll
Call name followed by either “has joined the conference” or “has left the conference”.
These voice messages can be replaced with audio tones. The use of tones requires that the
appropriate tone files in *.wav format be uploaded and configured to replace the Roll Call
Joined.wav and Roll Call Left.wav message files.
22 If you enable the Tone Notifications option, select the Entry Tone or Exit tone.
a Click the appropriate table entry in the Message File column.
A drop-down list is enabled.
b From the list, select the audio file to be assigned to the event/indication.
If the Tones option is enabled, you must assign the appropriate audio files to all notification types.
The Collaboration Server system is shipped with two default tones: Entry_tone.wav and
Exit_tone.wav.
If required, you can upload customized audio files that will be played when participants join or leave
the conference.
If the option to play a tone when a cascading link connection is established, make sure that the tone
selected for Entry or Exit notification differ from the cascading link tone as the latter one cannot be
customized.
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23 Click the Video Services tab.
The New Conference IVR Service - Video Services dialog box opens.
.
The Click&View and Invite Participants features are disabled in SVC and mixed CP and SVC
conferences.
In addition to the low and high resolution slides included in the default slide set, customized low and
high resolution slides are supported.
The following guidelines apply:
 Two customized slides can be loaded per IVR Service:
 A low resolution slide, to be used with low resolution endpoints.
 A high resolution slide, to be used with high resolution endpoints.
The following table summarizes the recommended input slide formats and the resulting slides that
are generated:
IVR Slide - Input / Output Formats
Format
Slide Resolution
Input Slides
Generated Slides
HD1080p
HD720p
High
HD1080p (16:9)
or
HD720p (16:9)
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IVR Slide - Input / Output Formats
Format
Slide Resolution
Input Slides
Generated Slides
Low
4CIF (4:3)
or
CIF (4:3)
4SIF
SIF
CIF
 The source images for the high resolution slides must be in *.bmp or *.jpg format.
 If the uploaded slides are not of the exact SD or HD resolution, an error message is displayed and
the slides are automatically cropped or enlarged to the right size.
 If a slide that is selected in an IVR Service is deleted, a warning is displayed listing the IVR
Services in which it is selected. If deleted, it will be replaced with a default Collaboration Server
slide.
 The generated slides are not deleted if the system is downgraded to a lower software version.
 The first custom source file uploaded, whatever its format, is used to generate both high and low
resolution custom slides. High resolution source files uploaded after the first upload will be used
to generate and replace high resolution custom slides. Likewise, low resolution source files
uploaded after the first upload will be used to generate and replace low resolution custom slides.
 If there are two custom source files in the folder, one high resolution, one low resolution, and a
new high resolution custom source file is uploaded, new high resolution custom slides are
created. The existing low resolution custom slides are not deleted.
 If there are two custom source files in the folder, one high resolution, one low resolution, and a
new low resolution custom source file is uploaded, new low resolution custom slides are created.
The existing high resolution custom slides are not deleted.
24 Define the following parameters
:
New Conference IVR Service Properties - Video Services Parameters
Video
Services
Description
Click&View
Select this option to enable endpoints to run the Click&View application that enables
participants to select a video layout from their endpoint.
Note: This option is not available in SVC conferences and for SVC participants in mixed CP
and SVC conferences.
Video
Welcome
Slide
Select the Low Resolution and High Resolution video slides to be displayed when
participants connect to the conference.
) button.
To view any slide, click the Preview Slide (
Notes:
• When using one of the default Polycom slides, the slide will be displayed in the resolution
defined in the profile, i.e. CIF, SD, HD 720p or HD 1080p. When defining a gateway IVR
Service, the recommended default slide is: Default_GW_Welcome_Slide.
• Customized H.261 slides are not supported.
When Collaboration Server is configured to IPv6, the IVR slide is displayed without taking
into account the MTU Size.
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New Conference IVR Service Properties - Video Services Parameters
Video
Services
Description
Invite
Participant
See Inviting Participants using DTMF.
Note: The Invite Participant feature is not available in SVC conferences and for SVC
participants in mixed CP and SVC conferences.
Dial out
protocols
order
Select the order of the network protocols that will be used by the system to dial the
destination number. The system will start dialing using the first protocol, and if the call is not
answered it will continue with the second, third and fourth protocols (if they are enabled)
until the call is answered. By default, H.323 is set as the first protocol and SIP as the second
while the remaining protocols are disabled (set to Off).
For PSTN calls, select the PSTN protocol and not ISDN.
Set PSTN before ISDN if both PSTN and ISDN protocols are required.
DTMF
forward
duration
Use this field when connecting to another conferencing entity with an IVR, requiring the
input of a password, destination number or ID. Enter the number of seconds that the system
will wait for the input of an additional DTMF digits such as a password or conference
number.
Range: 10 - 600 seconds
Default: 60 seconds.
25 If the video slide file was not uploaded to the MCU prior to the IVR Service definition, click the:
 Add Slide - Low Resolution button to upload a Low Resolution Slide.
 Add Slide - High Resolution button to upload a High Resolution Slide.
The Install File dialog box opens. The uploading process is similar to the uploading of audio files.
For more information, see step 6 above.
•
•
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The video slide must be in a .jpg or .bmp file format. For more information, see Creating a
Welcome Video Slide.
Customized H.261 slides are not supported.
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26 Click the DTMF Codes tab.
The New Conference IVR Service - DTMF Codes dialog box opens.
This dialog box lists the default DTMF codes for the various functions that can be performed during
the conference by all participants or by the chairperson.
Participants and chairpersons can manage their connection to ongoing conferences from their
endpoints, using DTMF codes from their endpoints. Chairpersons can also control an ongoing
conference using DTMF codes.
To use the DTMF codes to control the conference, the DTMF input must be first enabled on the
endpoint remote control.
New Conference IVR Service Properties - DTMF Codes
Operation
DTMF String
Permission
Mute My Line
*6
Everyone
Unmute My Line
#6
Everyone
Increase Broadcast Volume
Note: This option is not available for SVC participants.
*9
Everyone
Decrease Broadcast Volume
Note: This option is not available for SVC participants.
#9
Everyone
Mute All Except Me
*5
Chairperson
Cancel Mute All Except Me
#5
Chairperson
Change Password
*77
Chairperson
Mute Incoming Participants
*86
Chairperson
Unmute Incoming Participants
#86
Chairperson
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New Conference IVR Service Properties - DTMF Codes
Operation
DTMF String
Permission
Play Help Menu
*83
Everyone
Enable Roll Call
Note: This option is not available in SVC conferences.
*42
Chairperson
Disable Roll Call
Note: This option is not available in SVC conferences.
#42
Chairperson
Roll Call Review Names
Note: This option is not available in SVC conferences.
*43
Chairperson
Roll Call Stop Review Names
Note: This option is not available in SVC conferences.
#43
Chairperson
Terminate Conference
*87
Chairperson
Start Click&View
Note: This option is not available for SVC participants.
**
Everyone
Start PCM
Note: This option is not available for SVC participants.
##
Chairperson
Invite Participant
Note: This option is not available for SVC participants.
*72
Everyone
Disconnect Last Invited Participant
Note: This option is not available for SVC participants.
#72
Chairperson
Change To Chairperson
*78
Everyone
Increase Listening Volume
Note: This option is not available for SVC participants.
*76
Everyone
Decrease Listening Volume
Note: This option is not available for SVC participants.
#76
Everyone
Override Mute All
Configurable
Everyone
Start Recording
*2
Chairperson
Stop Recording
*3
Chairperson
Pause Recording
*1
Chairperson
Secure Conference
*71
Chairperson
Unsecured Conference
#71
Chairperson
Show Number of Participants
Note: This option is not available in SVC conferences.
*88
Everyone
Request individual assistance
Note: This option is not available for SVC participants.
*0
Everyone
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New Conference IVR Service Properties - DTMF Codes
Operation
DTMF String
Permission
Request assistance for conference
Note: This option is not available for SVC participants.
00
Chairperson
Request to Speak
99
Everyone
Touch Control Prefix
Note: This option is not available for SVC participants.
*#
Everyone
Do not change the DTMF code of the Touch Control Prefix (*#).
If during the ongoing conference the Show Number of Participants DTMF option (default DTMF
*88) is used, when the displayed number of participants is removed, the message overlay text is also
removed.
27 To modify the DTMF code or permission:
a In the DTMF Code column, in the appropriate entry enter the new code.
b In the Permission column, select from the list who can use this feature (Everyone or just the
Chairperson).
By default, the Secure, Unsecure Conference and Show Number of Participants options are enabled
in the Conference IVR Service.When Secured Conference is enabled, it locks the conference
and prevents participants and Collaboration Server Users from joining the conference. A
Secured Conference cannot be monitored or controlled. In the Secured Conference mode,
the Administrator cannot view the participant or conference properties, but can manually
terminate them. In the Secured Conference mode, the chairperson and participants can
perform various operations, such as muting through DTMF codes. However, because
Secure conferences cannot be monitored, these operations or status changes are not
displayed in the Conferences or Participants panes.
The Secure, Unsecure Conference and Show Number of Participants options can be disabled by
removing their codes from the Conference IVR Service.
To disable the Text Indication option in the DTMF Code column, clear the DTMF code (*88) of Show
Number of Participants from the table.
To disable the Secure Conference options, in the DTMF Code column, clear the DTMF codes of
both Secured Conference (*71) and Unsecured Conference (#71) from the table.
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28 Click the Operator Assistance tab.
The Operator Assistance dialog box opens.
29 Select Enable Operator Assistance to enable operator assistance when the participant requires or
requests help during the connection process to the conference or during the conference.
The Operator Assistance option is disabled in SVC conferences.
30 In the Operator Assistance Indication Message field, select the audio message to be played
when the participant requests or is waiting for the operator’s assistance.
If the audio file was not uploaded prior to the definition of the IVR Service or if you want to add new
audio files, click Add Message File to upload the appropriate audio file to the Collaboration Server.
31 Click OK to complete the IVR Service definition.
The new Conference IVR Service is added to the IVR Services list.
Change to Chairperson
Regular participants can request to become the conference chairperson using the appropriate DTMF code
(default: *78), which enabled them to perform operations designated for chairpersons only.
The Change to Chairperson via the DTMF code (default: *78) is executed only if the following settings were
configured for the MCU and the conference:
● In the Conference IVR Service - Conference Chairperson dialog box, select the Enable
Chairperson Messages check box, and select the appropriate voice messages.
For more information, see New Conference IVR Service Properties - Conference Chairperson
Options and Messages.
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● When starting a new conference or defining a new Meeting Room, define the Chairperson
Password in the conference General dialog box.
For more information, see Creating a New Meeting Room.
Controlling the receipt of in-band and out-of-band DTMF Codes
The RFC2833_DTMF System Flag controls the receipt of in-band or out-of-band DTMF Codes.
When set to YES (default), the RMX will receive DTMF Codes sent in-band. When set to NO the RMX
receives DTMF Codes sent out-of-band. The RMX always sends DTMF Codes in-band (as part of the Audio
Media stream, but not as RTPevents). If you wish to modify the flag value, the flag must be added to the
System Configuration file. For more information see Modifying System Flags.
Entry Queue IVR Service
An Entry Queue (EQ) is a routing lobby for conferences. Participants are routed to the appropriate
conference according to the conference ID they enter.
An Entry Queue IVR Service must be assigned to the Entry Queue to enable the voice prompts and video
slide guiding the participants through the connection process.
An Entry Queue IVR Service is a subset of an IVR Service. You can create different Entry Queue Services
for different languages and personalized voice messages.
The Collaboration Server is shipped with a default Entry Queue IVR Service and all its audio messages and
video slide. You can define new Entry Queue IVR Services or modify the default Entry Queue IVR Service.
Defining a New Entry Queue IVR Service
To set up a new Entry Queue IVR Service:
1 In the RMX Management pane, click IVR Services (
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2 In the IVR Services list, click the New Entry Queue IVR Service (
) button.
The New Entry Queue IVR Service - Global dialog box opens.
3 Fill in the following parameters:
Entry Queue IVR Service Properties - Global Parameters
Option
Description
Entry Queue Service
Name
(Mandatory) Enter the name of the Entry Queue Service. The name can be
typed in Unicode. Maximum field length is 80 ASCII characters.
Language
Select the language in which the Audio Messages and prompts will be heard.
The languages are defined in the Supported Languages function.
External Server
Authentication
This option is used for Ad Hoc conferencing, to verify the participant’s
permission to initiate a new conference. For a detailed description see Appendix
D - Ad Hoc Conferencing and External Database Authentication.
Select one of the following options:
• None to start a new conference without verifying with an external database
the user right to start it.
• Conference ID to verify the user’s right to start a new conference with an
external database application using the conference ID.
Number of User Input
Retries
Enter the number of times the participant is able to respond to each menu
prompt before the participant is disconnected from the MCU.
Timeout for User Input
(Sec.)
Enter the duration in seconds that the system waits for input from the participant
before it is considered as an input error.
DTMF Delimiter
The interaction between the caller and the system is done via touch-tone
signals (DTMF codes). Enter the key that will be used to indicate a DTMF
command sent by the participant or the conference chairperson. Possible keys
are the pound key (#) or star (*).
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4 Click the Welcome tab.
The New Entry Queue IVR Service - Welcome dialog box opens.
If the files were not uploaded prior to the definition of the IVR Service or if you want to add new audio
files, click Add Message File to upload the appropriate audio file to the Collaboration Server.
5 Define the appropriate parameters. This dialog box contains options that are identical to those in the
Conference IVR Service - Welcome Message dialog box. For more information about these
parameters, see New Conference IVR Service Properties - Conference Chairperson Options and
Messages.
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6 Click the Conference ID tab.
The New Entry Queue IVR Service - Conference ID dialog box opens.
7 Select the voice messages:
Entry Queue IVR Service Properties - Conference ID
Field/Option
Description
Request Conference ID
Prompts the participant for the conference ID.
Retry Conference ID
When the participant entered an incorrect conference ID, requests the
participant to enter the ID again.
8 Assign an audio file to each message type, as follows:
 In the Message File column, click the table entry, and then select the appropriate audio message.
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9 Click the General tab.
The New Entry Queue IVR Service - General dialog box opens.
The administrator can enable an audio message that informs the participant of the lack of Video
Resources in the Collaboration Server and that he/she is being connected as Audio Only. The
message states: All video resources are currently in use. Connecting using audio only.
The following guidelines apply:
 The IVR message applies to video participants only. Audio Only participants will not receive the
message.
 Only H.323 and SIP participants receive the audio message.
 Downgrade to Audio Only is not supported for undefined ISDN dial in participants. These
participants are disconnected if there is a lack of Video Resources.
 The audio message is the first message after the call is connected, preceding all other IVR
messages.
 The message is called No Video Resources-Audio Only and the message file (.wav) is called
No video resources audio only.wav.
 The audio message must be added to the Conference and Entry Queue IVR Services
separately.
 The IVR message can be enabled/disabled by the administrator using the
ENABLE_NO_VIDEO_RESOURCES_ AUDIO_ONLY_MESSAGE System Flag in system.cfg.
Possible values: YES / NO, default: YES
If you wish to modify the flag value, the flag must be added to the System Configuration file. For
more information see the Modifying System Flags.
10 Enter the message Name and Message File name for the Audio Only message:
 Message Name: No Video Resources-Audio Only
 Message File name: No_Video_Resources_Audio_Only.wav
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11 Click the Video Services tab.
The New Entry Queue IVR Service - Video Services dialog box opens.
12 In the Video Welcome Slide list, select the video slide that will be displayed to participants
connecting to the Entry Queue. The slide list includes the video slides that were previously uploaded
to the MCU memory.
13 To view any slide, click Preview Slide (
).
14 If the video slide file was not uploaded to the MCU prior to the IVR Service definition, click the:
 Add Slide - Low Resolution button to upload a Low Resolution Slide.
 Add Slide - High Resolution button to upload a High Resolution Slide.
The Install File dialog box opens. The uploading process is similar to the uploading of audio files.
For more information, see step 6.
The video slide must be in a .jpg or .bmp file format. For more information, see Creating a Welcome
Video Slide.
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15 Click the Operator Assistance tab.
The Operator Assistance dialog box opens.
16 Select Enable Operator Assistance to enable operator assistance when the participant requires or
requests help during the connection process.
17 In the Operator Assistance Indication Message field, select the audio message to be played
when the participant requests or is waiting for operator’s assistance.
If the audio file was not uploaded prior to the definition of the IVR Service or if you want to add new
audio files, click Add Message File to upload the appropriate audio file to the Collaboration Server.
18 Click OK to complete the Entry Queue Service definition.
19 The new Entry Queue IVR Service is added to the IVR Services list. For more information, see IVR
Services List.
Setting a Conference IVR Service or Entry Queue IVR Service as the
Default Service
The first Conference IVR Service and Entry Queue IVR Service are automatically selected by default. The
IVR Services (Conference and Entry Queue) shipped with the system are also set as default. If additional
Conference IVR Services and Entry Queue IVR Services are defined, you can set another service as the
default for each service type.
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To select the default Conference IVR Service:
● In the IVR Services list, select the Conference IVR Service to be defined as the default, and then
click Set Default Conference IVR Service (
).
Alternatively, in the IVR Services list, right-click the Conference IVR Service and then select Set
Default Conference IVR Service.
The IVR Service is displayed in bold, indicating that it is the current default service.
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To select the Default Entry Queue IVR Service:
» In the IVR Services list, select the Entry Queue IVR Service to be defined as the default, and then
).
click Set Default Entry Queue IVR Service (
Alternatively, in the Conference IVR Services list, right-click the Entry Queue IVR Service and then
select Set Default Entry Queue IVR Service.
The default Entry Queue IVR Service is displayed in bold, indicating that it is the current default
service.
Modifying the Conference or Entry Queue IVR Service
Properties
You can modify the properties of an existing IVR Service, except the service name and language.
To modify the properties of an IVR Service:
1 In the RMX Management pane, click IVR Services.
2 In the