What's new in version 19?
The new features and enhancements and changes in functionality included in Pexip Infinity version 19 are described below.
For full information about this release, see the release notes.
For information about earlier versions of Pexip Infinity, see Features added in previous releases.
New features
Pexip Infinity platform
Feature | Description | More information |
---|---|---|
Video interoperability with Google Hangouts Meet |
The Pexip Distributed Gateway provides any-to-any video interoperability with Google Hangouts Meet. Third-party systems can connect to Hangouts Meet conferences via the Pexip Distributed Gateway either by dialing the conference directly or via a Virtual Reception (IVR). |
Integrating Google Hangouts Meet with Pexip Infinity |
Skype for Business Video-based Screen Sharing (VbSS) |
Pexip Infinity supports sending and receiving content via VbSS to and from Skype for Business meetings, and to and from Skype for Business clients that are either calling another endpoint via the Pexip Distributed Gateway, or calling into a Virtual Meeting Room. Note that VbSS is disabled by default on Pexip Infinity. It can be enabled via . (VbSS was previously available as technology preview in earlier versions.) |
Enable VbSS for Skype for Business |
Improved video quality in low bandwidth scenarios |
Participants now experience improved video quality in low bandwidth network scenarios, and when endpoints can only support low resolutions. Pexip Infinity prioritizes image sharpness over high frame rates, and now sends video at lower frame rates if 30 fps is not achievable. |
|
Enhancements to Live View |
The Live View summary contains the following enhancements when showing active conferences:
|
Viewing live and historical platform status |
Infinity Connect web app
Following are the changes to the Infinity Connect web app in Pexip Infinity version 19:
Feature | Description | More information |
---|---|---|
Ability to force a call protocol when adding a participant to a conference |
When adding a participant to a conference, you have the option to force a specific dial out protocol by prefixing the destination address with sip: or mssip: or h323: or rtmp: (which can be used to support dial out to streaming services). When a protocol has been explicitly added to the address, a Call Routing Rule is not required for the call to be placed. |
Streaming and recording a conference |
Diagnostics |
The About this app menu now has an option to copy logs to the clipboard. |
Logs |
Changes in functionality in this release
Feature | Description | More information |
---|---|---|
Consistent display of participant names in overlays | The way in which participant names are shown as text overlays within a conference has been standardized. Any protocol or trailing IP address and port information is stripped from the displayed name/alias. For example, alice@example.com is now shown instead of sip:alice@example.com. | Selecting the layout seen by participants |
Administrative modifications |
This release contains the following administrative modifications:
|