Enabling and disabling SIP, H.323, WebRTC and RTMP

All call protocols, except for SIP over UDP, are enabled by default in your Pexip Infinity deployment.

If your deployment does not include any endpoints that support a particular protocol, for security reasons you may want to disable support for those protocols across your entire Pexip Infinity deployment.

SIP over UDP is disabled by default so as to reduce the impact of SIP spam which typically uses UDP.

The configurable protocols are:

• SIP over TCP and TLS, including MS-SIP (Skype for Business / Lync)
• SIP over UDP (for incoming calls)
• H.323
• WebRTC
• RTMP

WebRTC calls can originate from the Connect desktop app, the Connect web app via Google Chrome, Microsoft Edge, Firefox, and Safari (version 11 onwards) browsers, and the Connect mobile app.

RTMP and RTMPS (for encrypted RTMP) are used to send conference content to streaming and recording services. RTMP authentication is supported; in this case credentials are included in the URI using the syntax rtmps://username:password@host/....

The Live view page (Status > Live View) lets you review current and historic usage charts showing a breakdown of participants by location, protocol, license type and the different conference types being hosted.

To enable or disable a particular call protocol across your Pexip Infinity deployment:

1. Go to Platform > Global settings > Connectivity.

2. Select or clear the following checkboxes, as appropriate:

   • Enable SIP — this affects both SIP and MS-SIP (Skype for Business / Lync) over TCP and TLS
   • Enable SIP UDP — this affects incoming calls only over SIP UDP
   • Enable H.323
   • Enable WebRTC
   • Enable RTMP
3. If you have Enabled WebRTC or Enabled RTMP and wish to use the Infinity Connect clients or client API, you must also Enable support for Pexip Infinity Connect clients and Client API.