Ports and services for WebRTC phones under BYOC Premises

Coming December 7, 2024 - RTP UDP port expansion (Commercial) 16384–32768 to 16384–65535.

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.

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.
	udp/19302 (STUN)^†	Google^*
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).

† Optional

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

‡ Not currently in use, but should be open and reserved for future use.

Revision history

Modified date (YYYY-MM-DD)	Revision
2024-11-18	Updated the status of tcp/3478 (STUN) Transport/Port for the WebRTC signaling service.
2022-02-16	Removed tcp/3478 (STUN) and tcp/19302 (STUN) from the WebRTC signaling row. Tcp is not used for STUN.
2020-12-14	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.