Source note · Troubleshooting audio delay and talk-over on voice calls
Genesys says the WebRTC Diagnostics jitter value should be under 30 ms and packet loss under 1 percent.
Checked 2026-10-02
- Vendor
- Genesys
- Product
- Genesys Cloud
- Subsystem
- WebRTC phone
- Deployment
- multi-tenant
- Region
- not restricted
- Release range
- current as of 2026-10-02
- Checked
- 2026-10-02
Sources
- Run the standalone Genesys Cloud WebRTC Diagnostics app — Genesys · tier 2 current vendor documentation · Network Test results
Cite this note
APA
WarmTransfer. (2026, October 2). Troubleshooting audio delay and talk-over on voice calls: source note call-latency-troubleshooting-genesys-webrtc-jitter-30ms. WarmTransfer. https://warmtransfer.net/knowledge/claims/call-latency-troubleshooting-genesys-webrtc-jitter-30ms
BibTeX
@misc{warmtransfer-claim-call-latency-troubleshooting-genesys-webrtc-jitter-30ms,
title = {Troubleshooting audio delay and talk-over on voice calls: source note call-latency-troubleshooting-genesys-webrtc-jitter-30ms},
author = {{WarmTransfer}},
year = {2026},
url = {https://warmtransfer.net/knowledge/claims/call-latency-troubleshooting-genesys-webrtc-jitter-30ms},
note = {Source note call-latency-troubleshooting-genesys-webrtc-jitter-30ms, checked 2026-10-02}
}