Ports and services for WebRTC phones under BYOC Cloud
This reference article lists the ports required for access to specific services for WebRTC phones under BYOC Cloud. For more information on other ports and services you may need to configure on your firewall, see About ports and services for your firewall.
Note: Firewall settings for BYOC Cloud will be provided by your carrier.
| 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). |
|
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/16384-32768 (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.
| Date | Revision |
| February 15, 2023 | Added Genesys Cloud Media Tier to the Destination column for udp/3478 (STUN) |
| January 6, 2022 | In the Destination column, renamed Genesys Cloud Edge Devices to Genesys Cloud Media Tier. For more information, see Edge terminology reset. |
| 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 Cloud. |