About global settings

The length of time (in seconds) for which a conference will continue with only Guest participants, after all Host participants have left.

Default: 60 seconds

The length of time (in seconds) for which a conference will continue with only one participant remaining. The type of participant (Host, Guest, automatically dialed, streaming etc) is irrelevant.

It also ends any gateway calls into a Microsoft Teams / Google Meet meeting when all of the Teams / Google Meet clients (or other gatewayed participants) have left.

The time can be configured to values between 60 seconds and 86400 (1 day), or to 0 (never eject).

Default: 0 (never eject)

The length of time (in seconds) for which a participant is allowed to remain at the PIN entry screen before being disconnected.

Default: 120 seconds

The length of time (in seconds) for which a Guest participant can remain at the waiting screen if a Host does not join, before being disconnected.

Default: 900 seconds

Controls the maximum call quality for participants connecting to Pexip Infinity services.

You can override this global setting for each individual service (VMR, Call Routing Rule etc). For example, you could use the default option of "HD" for most of your services by default, but enable Full HD on some specific services. The options are:

• SD: each participant is limited to SD quality.
• HD: each participant is limited to HD (720p) quality.
• Full HD (1080p): allows any endpoint capable of Full HD to send and receive its main video at 1080p.

Default: HD

See Setting and limiting call quality for more information.

When sending main video and presentation to a standards-based (SIP or H.323) or WebRTC endpoint, this defines the maximum percentage of the call bandwidth to allocate to the presentation content (with the remainder allocated to main video). It must be in the range 25% to 75%.

Default: 75%

Note that when sending two video streams to a dual-screen endpoint, the call bandwidth is always split 50-50 between video and presentation content (and cannot be changed).

Determines whether or not avatars for external participants are retrieved using a method appropriate for the external meeting type. Currently this only applies to Microsoft Teams conferences. For all other conference types, and for when this option is not selected, avatars may be retrieved via external policy or user records as per standard behavior. You can also configure this setting on individual Call Routing Rules for Microsoft Teams conferences.

Default: enabled.

Controls support for the SIP protocol over TCP and TLS across all Conferencing Nodes in your Pexip Infinity deployment.

Note that disabling SIP will disable support for Skype for Business / Lync (MS-SIP).

Default: enabled.

Allows or prevents incoming calls over SIP UDP.

Default: disabled.

Enable H.323 * If you change this setting, all existing calls will be disconnected and all Conferencing Nodes will be automatically restarted.

WebRTC

RTMP * If you change this setting, all existing calls will be disconnected and all Conferencing Nodes will be automatically restarted.

These boxes control support for the selected protocols across all Conferencing Nodes in your Pexip Infinity deployment.

Default: all of these settings are enabled by default.

Enables support for the Pexip Infinity client API. This is required for integration with the Connect apps (web, desktop and mobile), and any other third-party applications that use the client API, as well as for integration with Microsoft Teams and Poly OTD endpoints for One-Touch Join.

This setting must be enabled if you want to Enable WebRTC or Enable RTMP.

Default: enabled.

This setting enables media relay on TCP port 443 on all Conferencing Nodes. This is intended as a fallback mechanism for use by WebRTC clients that are behind strict firewalls that block RTP media to Pexip's standard ports. This setting should only be enabled when it is impossible to amend the firewall's rules to allow UDP media, as sending media over TCP can result in increased latency and jitter.

Enabling this setting may cause disruption to ongoing WebRTC sessions.

This setting is not compatible with the Connect desktop app for Citrix Workspace app.

Enables Connect apps and SIP/H.323 endpoints to send Far-End Camera Control (FECC) signals to supporting endpoints, in order to pan, tilt and zoom the device's camera.

Support for SIP/H.323 endpoints sending FECC signals is for gateway calls only.

Default: enabled.

Enables relay of chat messages between conference participants using Skype for Business / Lync and Infinity Connect clients.

You can override this setting on a per conference basis (for a Virtual Meeting Room or Virtual Auditorium).

Default: enabled.

Controls whether any calls can be made via the Infinity Gateway, and allows dial-out from a conference (via the Connect apps and the Administrator interface).

Default: enabled.

Placing calls via the Pexip Infinity Distributed Gateway

Manually dialing out to a participant from a conference

This setting controls the system behavior when dialing out via a Connect app or the client API to a participant from an ongoing conference.

When selected (enabled), calls placed via the:

• Connect apps always use automatic routing and thus must match an appropriate Call Routing Rule.
• Client API can either use automatic routing or they can specify a dial-out protocol without any need for a Call Routing Rule i.e. it allows end-users to perform arbitrary dial outs (and thus circumvent any administrator-set rules).

When not selected (disabled), calls that are placed via:

• any Connect app, or via the client API, always use automatic routing and thus must match an appropriate Call Routing Rule.

Note that dial out via the Administrator interface, management API or Automatically Dialed Participants (ADPs) is unaffected by this setting.

Default: disabled.

This is an optional field where you can specify the name of the SIP domain that is:

• Routed from Skype for Business / Lync to Pexip Infinity, either as a static route or via federation.
• Used as the default domain in the From address for outgoing SIP gateway calls and outbound SIP calls from conferences without a valid SIP URI as an alias.

You can also configure the Pexip Infinity domain on a per-location basis, which would override this global setting for Conferencing Nodes in that location.

Skype for Business with Pexip Infinity

When selected, this automatically escalates a Skype for Business / Lync audio call so that it receives video from a conference.

Default: disabled.

Controls support for Skype for Business Video-based Screen Sharing (VbSS).

Note that VbSS is always enabled for Microsoft Teams calls, regardless of this setting.

Default: disabled.

For information about enabling VbSS on your Skype for Business infrastructure see https://technet.microsoft.com/en-us/library/mt756736.aspx.

Allows an administrator to log in to the Management Node and all Conferencing Nodes over SSH.

This setting can be overridden on individual nodes.

Default: enabled.

When disabled, Connect apps display aliases from their own call history only.

When enabled, registered Connect apps additionally display the aliases of Virtual Meeting Rooms, Virtual Auditoriums, Virtual Receptions, and devices registered to the Pexip Infinity platform.

Default: enabled.

When enabled, if a location only contains Proxying Edge Nodes, then those nodes only require IPsec connectivity with other nodes in that location, the transcoding location, the primary and secondary overflow locations, and with the Management Node. When disabled, a full connectivity mesh is required between all nodes in the deployment.

Default: enabled.

Controls the media encryption requirements for participants connecting to Pexip Infinity services.

You can override this global setting for each individual service (VMR, Call Routing Rule etc). For example, you could use the default option of "best effort" for most of your services, but enforce encryption on some specific services.

• Best effort: each participant will use media encryption if their device supports it, otherwise the connection will be unencrypted.
• Required: all participants (including RTMP participants) must use media encryption.
• No encryption: all H.323, SIP and MS-SIP participants must use unencrypted media. (RTMP participants will use encryption if their device supports it, otherwise the connection will be unencrypted.)

Default: Best effort

The start and end values for the range of ports (UDP and TCP) that all Conferencing Nodes use to send signaling (for H.323, H.245 and SIP).

Default: 33000–39999.

The start and end values for the range of ports (UDP and TCP) that all Conferencing Nodes use to send media for H.323, SIP, Skype for Business / Lync, WebRTC and RTMP (note that RTMP uses TCP only). The media port range must contain at least 100 ports.

Default: 40000–49999.

Controls which codecs to offer in audio/video negotiation (SDPs).

Some third-party systems can experience issues if they are sent a large SDP from Pexip Infinity. You can reduce the size of the SDP by disabling specific, unwanted codecs.

Default: all codecs are selected except AAC-LD128, H.264 High (mode 0) and H.264 High (mode 1).

To enable the H.264 High Profile codec, move H.264 High (mode 1) into the list of Chosen Codecs. For optimal interoperability results, only enable H.264 High (mode 1) — leave H.264 High (mode 0) in the Available Codecs list.

Determines whether OCSP is used to check the status of TLS certificates.

Off: OCSP is not used.

On: OCSP is used, and the request is sent to the URL specified in the TLS certificate. If no URL is specified in the TLS certificate, the OCSP responder URL configured below is used.

Override: OCSP is used, and the request is sent to the OCSP responder URL specified in the OCSP responder URL field, regardless of any URL encoded in the TLS certificate.

Default: Off.

The URL to which OCSP requests are sent if either:

• the OCSP state is set to On but no URL is present in the TLS certificate, or
• the OCSP state is set to Override (in which case any URL present in the certificate is ignored).

Determines whether to verify the peer certificate for outbound connections over SIP TLS. The options are:

Off: the peer certificate is not verified; all connections are allowed.

On: the peer certificate is verified, and the peer's remote identities (according to RFC5922) are compared against the Application Unique String (AUS) identified by Pexip Infinity — the SIP URI — before the connection is allowed.

Default: Off.

The maximum number of days of logs and call history to retain on Pexip nodes. On busy systems, logs may still be rotated before this time due to limited disk space.

Enter 0 to have no set limit.

Default: 0.

Determines whether or not HTTP Content-Security-Policy (CSP) headers for Conferencing Nodes are enabled.

Default: enabled.

Defines the contents of the HTTP Content-Security-Policy headers for Conferencing Nodes when CSP is enabled.

The default header string contains multiple directives such as frame-src and script-src, delimited by the ; character.

For more information on CSP, see this MDN article.

Default: upgrade-insecure-requests; default-src 'self'; frame-src 'self' https://telemetryservice.firstpartyapps.oaspapps.com/telemetryservice/telemetryproxy.html https://*.microsoft.com https://*.office.com; style-src 'self' 'unsafe-inline' https://*.microsoft.com https://*.office.com; object-src 'self'; font-src 'self' https://*.microsoft.com https://*.office.com; img-src 'self' https://www.adobe.com data: blob:; script-src 'self' 'unsafe-inline' 'unsafe-eval' https://*.microsoft.com https://*.office.com https://ajax.aspnetcdn.com https://api.keen.io; media-src 'self' blob:; connect-src 'self' https://*.microsoft.com https://*.office.com https://example.com; frame-ancestors 'self';

These defaults are appropriate for typical usage of Pexip Infinity.

Note that:

• The Microsoft addresses are required for Outlook add-ins.
• The reserved domain example.com is included to support a third-party JavaScript library that is used by the Connect web app for PDF content sharing. You may need to add extra headers if you use custom plugins with the Connect web app.
• The 'unsafe-inline' keyword is only required if you use Connect Webapp2.

Select this option to instruct Pexip Infinity to temporarily block all access to a VMR that receives a significant number of incorrect PIN entry attempts.

You can override this setting on a per location basis.

Default: enabled.

The maximum number of PIN failures allowed in any 10-minute window before the VMR is blocked.

Default: 20.

Select this option to instruct Pexip Infinity to temporarily block service access attempts from any unknown source IP addresses that dial a significant number of incorrect aliases.

You can override this setting on a per location basis.

Default: enabled.

The maximum number of incorrect dial attempts in any 10-minute window before the source IP address is blocked.

Default 20.

Access for external systems is over HTTPS by default. If this box is selected, access over HTTP is also permitted.

Default: disabled.

Controls whether inactive users are automatically logged out of the Administrator interface after a period of time.

If enabled, users are logged out after a number of minutes of inactivity as specified in the Management web interface session timeout setting.

If disabled, users of the Administrator interface are never timed out. You may want to use this option if, for example, you have an administrator session that permanently monitors the system live view.

Default: enabled.

The number of minutes a browser session may remain idle before the user is logged out of the Management Node Administrator interface, if Enable management web interface session timeout is selected.

Default: 30 minutes.

Controls whether conferences and backplanes are shown in the Live View graph. If you have a very busy deployment, it may be useful to disable conferences and backplanes from the Live View for an improved viewing experience.

Note that when conferences and backplanes are removed, the conferences and participants counts in Live View are not shown.

Default: enabled.

Controls the first page you are directed to after logging into the Administrator interface.

Default: Live View

Site banner text

Banner background color

Banner text color

These options configure the site banner (across the top of every webpage) for the Pexip Infinity Administrator interface.

You can define the text to display, and use the color pickers to select the text and background colors.

Enable incident reporting

Incident reporting URL

Contact email address

If incident reporting is enabled, reports are sent to the specified URL.

The contact email address will be used by Pexip Support when following up on any incident reports received from your deployment.

These settings are configured during initial installation of the Management Node (when running the installation wizard).

Select this option to allow submission of deployment and usage statistics to Pexip. This will help us improve the product.

This setting is configured during initial installation of the Management Node (when running the installation wizard).

A range of advanced options to tune the event sink processes.

See Using event sinks to monitor conference and participant status for details.

Configuring dynamic bursting to the AWS cloud

Configuring dynamic bursting to the Microsoft Azure cloud

Configuring dynamic bursting to the Google Cloud Platform (GCP)

This option controls whether live captions are enabled on all VMRs and Call Routing Rules by default.

You can override this global setting for each individual VMR or Call Routing Rule.

Configuring Virtual Meeting Rooms (VMRs)

Configuring Call Routing Rules

Voice Focus is a platform-wide setting that improves the way in which voice activity is detected by better distinguishing between actual speech and background noise. This reduces the probability that people who are not speaking but have audible background noise will be switched into the main speaker position. Note that this does not remove any noise from the audio.

When enabled it applies to Virtual Meeting Rooms and Virtual Auditoriums, and it uses additional hardware resources (equivalent to an extra 6 audio connections per participant).

Default: disabled.

Select this option to enable live captions functionality and to display the Live captions configuration section on this Global Settings page (above).

This allows you to configure a VMR or Call Routing Rule so that it can convert the live meeting audio to a readable text (live transcription). This then enables participants using the web app to view those live captions.

Please contact your Pexip authorized support representative or your Pexip Solution Architect for guidance on enabling this feature specific to your environment.

When enabling this feature note that:

• You must configure your system locations (that are handling the conference participant connections) with the associated outgoing locations to use for connecting to the Live captions service API gateway.
• The captions are only shown to a web app user who has elected to view the captions. However, all of the other meeting participants are notified that the call is being transcribed.
• The Turn live captions on button is only displayed in the web app when the live captions are enabled via global settings and on the relevant VMR or Call Routing Rule.
• It is not available for Skype for Business / Lync and Microsoft Teams gateway calls.
• When using Connect Webapp2, the live transcriptions history can be viewed on the left side panel; this feature is not currently available for Connect Webapp3 users.

Live captions

About system locations

Softmute is advanced speech-aware audio gating which helps to minimize noise coming from a participant who has their microphone turned on in a conference, but is not speaking. If non-voice noise is detected, this feature softens the gain from that participant. It does not entirely suppress noise from an audio signal.

When it is enabled here, it can then be individually enabled or disabled at the VMR level (Services > Virtual Meeting Rooms > Advanced options > Softmute).

It uses additional hardware resources (equivalent to an extra 6 audio connections per participant). However, if Voice Focus is also enabled, then both features use a combined total of an extra 6 audio connections per participant i.e. there is no extra resource use for enabling Softmute when Voice Focus is also enabled.

Denoising is a server-side feature that removes the background noise from somebody who is speaking. In comparison, softmute softens the gain from a noisy, but non-speaking participant.

When it is enabled here, it can then be individually enabled or disabled at the VMR level (Services > Virtual Meeting Rooms > Advanced options > Denoise).

It uses additional hardware resources (equivalent to an extra 7 audio connections per participant). However, if Voice Focus is also enabled, then both features use a combined total of an extra 7 audio connections per participant i.e. there is no extra resource use for enabling Denoise when Voice Focus is also enabled. If Voice Focus, Softmute and Denoise are all enabled then it still uses a combined total of an extra 7 audio connections per participant.

Note that this feature is separate from Google Meet denoising, which is a client-side feature and applies to Google Meet integrations only.

This setting configures the use of breakout rooms on VMRs. When enabled, you can configure individual VMRs so that its participants can be sent into different breakout rooms.

Breakout rooms

Configuring Virtual Meeting Rooms (VMRs)