Role Studio

This user can login into the web UI and make calls to Journalists, Guest users, mobile or landline phone numbers and SIP Codecs

Select the role: “ROLE_STUDIO”


Configure permissions to access different options in the web UI

  • Administrator, allows access to administration section
  • Bitrate, user can override the bitrate for any call
  • Configure alias, user can create alias for calls
  • Can view stats, user can access the call statistics
  • Temporal admin, user can create guest users
    • Temporal group, if Temporal admin is enabled, the group the user can admin

Audio controls

In this section you can configure the different audio controls

  • Max calls, the maximum number of simultaneous calls. This is defined by your license type
  • On Air Groups, the number of On Air Groups that you will use in your Studio. Define this depending on your available resources in your PC and audio mixer
  • Show all on air group, if enabled you all On Air buttons, up-to 10. If not enabled it shows On Air 1 and the second On Air can be configured with the right button
  • Default bitrate, the bitrate that applies for every calls
  • Push to talk, if enabled you need to keep pressed the Talk button
  • Allow talk in on-air, it enables or disables talk button while the call is on-air
  • Deactivate PFL with on-air, if enabled PFL will be automatically deactivated when you click in any on-air button
  • Activate PFL with on-air, if enabled PFL will be automatically activated when you disable on-air button
  • PFL listen on-air, if enabled any user that is not in On Air will listen to On Air 1 input
Audio controls

Noise gate

Noise gate settings

  • Enable noise gate, automatically enable noise gate in every call
  • Configure noise gate, studio user can access the configuration of noise gate in live connections
  • Thresdhold in DBFS that apply by default
  • Attack time in seconds that apply by default
  • Release time in seconds that apply by default
Noise gate settings


SIP configuration. You can use the embedded SIP Server included with your license.

To use an external SIP Server it must comply with the SIP Over Websocket standard The websocket Protocol as a Transport for the Session Initiation Protocol SIP RFC-7118

If you need to register your IP Codec please open a ticket with the support team

  • SIP checkbox enables SIP Support
  • SIP username
  • SIP password
  • Auto answer extension, any call from this extension to the studio will be automatically answered
  • Auto reject, any call from any extension (except auto answer extension) will be automatically rejected
  • Transfer extension, allows to transfer any call from the studio to this extension
SIP configuration