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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 (Lync / Skype for Business)
  • SIP over UDP (for incoming calls)
  • H.323
  • WebRTC
  • RTMP

WebRTC calls can originate from the Infinity Connect desktop client, and Infinity Connect via Google Chrome, Firefox and Opera browsers.

RTMP calls can originate from the Infinity Connect Web App via Internet Explorer, Edge, and Safari browsers. The Web App will attempt an encrypted RTMPS connection first. For a secure RTMP connection to be established, the SIP TLS FQDN must be configured on the Conferencing Node (via Platform configuration > Conferencing Nodes) and it must match the Common Name of its TLS server certificate. If RTMPS fails, it will use an unencrypted connection for media.

RTMP is also used when sending conference content to streaming and recording services.

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

  1. Go to Platform configuration > Global settings.
  2. Select or clear the following checkboxes, as appropriate:

    • Enable SIP — this affects both SIP and MS-SIP (Lync / Skype for Business) over TCP and TLS
    • Enable SIP UDP — this affects incoming calls only over SIP UDP
    • Enable H.323
    • Enable WebRTC
    • Enable RTMP
  3. After modifying any of these values, each Conferencing Node must be rebooted before the changes will take effect. (It will, however, have an immediate effect on outbound calls placed via the modified protocols.)