A couple of things we have found. First, WebRTC initially provisions over Https (port 443) to your ORG's region and a TLS session is pinned up. Once a call is established, it will transmit audio (RTP) over the media ports (
udp/16384-32768) to the Genesys Cloud Media IPs (
52.129.96.0/20). We have had very few customers that have 443 blocked and most of them have no problem hitting the STUN ports on Google. What I tell customers is to try the WebRTC phone and run the diagnostics first before going through the hassles of getting security to approve ports and IP's. ------------------------------
Robert Wakefield-Carl
Avtex Solutions, LLC
Contact Center Innovation Architect
robertwc@avtex.comhttps://www.Avtex.comhttps://RobertWC.Blogspot.com------------------------------