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 Media Tier, 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 Media	udp/16384-65535 (SRTP/TURN)	Genesys Cloud Media Tier	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 historyClick to expand

Modified date (YYYY-MM-DD)	Revision
2024-11-18	Updated the status of tcp/3478 (STUN) Transport/Port for the WebRTC signaling service.
2024-10-07	Added Coming soon flag: October 28, 2024 – RTP UDP port expansion (Commercial) 16384–32768 to 16384–65535
2022-01-06	In the Destination column, renamed Genesys Cloud Edge Devices to Genesys Cloud Media Tier. For more information, see Edge terminology reset.
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 Genesys Cloud Voice.

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