Ports and services for WebRTC phones under BYOC Premises
This reference article lists the ports required for access to specific services for WebRTC phones under BYOC Premises. For more information on other ports and services you may need to configure on your firewall, see About ports and services for your firewall.
|WebRTC signaling||tcp/443 (HTTPS)||Genesys Cloud, Amazon AWS||The secure connection for VoIP signaling (dialing, ringing, etc. for inbound and outbound calls).|
Genesys Cloud, Amazon AWS
|These ports must be opened for both the client and Edges. These are used for the srflx and relay candidates. If they are closed, calls will have a high rate of failure.|
|WebRTC Cloud||tcp/5061||Genesys Cloud, Amazon AWS||The connection for Edges to connect to the Genesys Cloud services for WebRTC phones|
|WebRTC Media||udp/16384-32768 (SRTP/TURN)||Genesys Cloud Media Tier, Genesys Cloud, Amazon AWS||The transmission of secured streaming media (audio).|
* Third-party service; not hosted by Genesys Cloud.
|February 16, 2022||Removed tcp/3478 (STUN) and tcp/19302 (STUN) from the WebRTC signaling row. Tcp is not used for STUN.|
|December 14, 2020||Broke out the main sections of the larger Ports and services for your firewall article into smaller articles. Created this article to cover the ports and services for WebRTC phones under BYOC Premises.|