Source note · WebRTC in modern contact centers
Under Genesys Cloud Voice, WebRTC phone audio (SRTP/TURN) uses UDP 16384-65535 to the Genesys Cloud Media Tier.
Checked 2026-09-24
- Vendor
- Genesys
- Product
- Genesys Cloud WebRTC phone
- Subsystem
- network
- Deployment
- multi-tenant
- Region
- not restricted
- Release range
- page updated 2024-11-18
- Checked
- 2026-09-24
Sources
- Ports and services for WebRTC phones under Genesys Cloud Voice — Genesys · tier 2 current vendor documentation · Ports table, WebRTC Media row
Cite this note
APA
WarmTransfer. (2026, September 24). WebRTC in modern contact centers: source note webrtc-deployment-genesys-media-range. WarmTransfer. https://warmtransfer.net/knowledge/claims/webrtc-deployment-genesys-media-range
BibTeX
@misc{warmtransfer-claim-webrtc-deployment-genesys-media-range,
title = {WebRTC in modern contact centers: source note webrtc-deployment-genesys-media-range},
author = {{WarmTransfer}},
year = {2026},
url = {https://warmtransfer.net/knowledge/claims/webrtc-deployment-genesys-media-range},
note = {Source note webrtc-deployment-genesys-media-range, checked 2026-09-24}
}