Source note · Troubleshooting Genesys Cloud WebRTC phone problems
Genesys recommends allowing 32-128 Kbps of bidirectional voice bandwidth per concurrent call for adaptive Opus phones, plus 5% of the high value for SIP control traffic.
Checked 2026-10-02
- Vendor
- Genesys
- Product
- Genesys Cloud
- Subsystem
- network readiness
- Deployment
- multi-tenant
- Region
- not restricted
- Release range
- current as of 2026-10-02
- Checked
- 2026-10-02
Sources
- Customer network readiness (Genesys Cloud network requirements) — Genesys · tier 2 current vendor documentation · Section: Network bandwidth
Cite this note
APA
WarmTransfer. (2026, October 2). Troubleshooting Genesys Cloud WebRTC phone problems: source note genesys-cloud-webrtc-troubleshooting-opus-bandwidth. WarmTransfer. https://warmtransfer.net/knowledge/claims/genesys-cloud-webrtc-troubleshooting-opus-bandwidth
BibTeX
@misc{warmtransfer-claim-genesys-cloud-webrtc-troubleshooting-opus-bandwidth,
title = {Troubleshooting Genesys Cloud WebRTC phone problems: source note genesys-cloud-webrtc-troubleshooting-opus-bandwidth},
author = {{WarmTransfer}},
year = {2026},
url = {https://warmtransfer.net/knowledge/claims/genesys-cloud-webrtc-troubleshooting-opus-bandwidth},
note = {Source note genesys-cloud-webrtc-troubleshooting-opus-bandwidth, checked 2026-10-02}
}