Ports and services for WebRTC phones under Genesys Cloud Voice

This reference article lists the ports required for access to specific services for WebRTC phones under Genesys Cloud Voice. For more information on other ports and services you may need to configure on your firewall, see About ports and services for your firewall

Services Transport/Port (Application) Destination Description
WebRTC signaling tcp/443 (HTTPS) Genesys Cloud, Amazon AWS The secure connection for VoIP signaling (dialing, ringing, etc. for inbound and outbound calls).

tcp/3478 (STUN)

udp/3478 (STUN)

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.

tcp/19302 (STUN)

udp/19302 (STUN)

Google*

WebRTC Media udp/16384-32768 (SRTP/TURN) Genesys Cloud Edge devices, Genesys Cloud, Amazon AWS The transmission of secured streaming media (audio).

† Optional

* Third-party service; not hosted by Genesys Cloud.

Date Revision
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 Genesys Cloud Voice.