Ports and services for WebRTC phones under Genesys Cloud Voice

Publisher

Genesys

URL

https://help.genesys.cloud/articles/ports-and-services-for-webrtc-phones-under-genesys-cloud-voice/

Published

2024-11-18

Updated

unknown

Accessed

2026-09-24

HTTP status

200

License

Genesys copyright; all rights reserved; no-redistribution; short excerpts and locators only

Source notes citing this source

• Under Genesys Cloud Voice, WebRTC phone audio (SRTP/TURN) uses UDP 16384-65535 to the Genesys Cloud Media Tier. in context
• Under Genesys Cloud Voice, the WebRTC phone uses UDP 3478 STUN for server-reflexive and relay candidates, and Genesys warns calls will have a high failure rate if it is closed. in context

Cite this source record

APA

WarmTransfer. (2024, November 18). Ports and services for WebRTC phones under Genesys Cloud Voice. WarmTransfer. https://warmtransfer.net/knowledge/sources/genesys-help-ports-webrtc-cloud-voice

BibTeX

@misc{warmtransfer-genesys-help-ports-webrtc-cloud-voice,
  title  = {Ports and services for WebRTC phones under Genesys Cloud Voice},
  author = {{WarmTransfer}},
  year   = {2024},
  url    = {https://warmtransfer.net/knowledge/sources/genesys-help-ports-webrtc-cloud-voice},
  note   = {Genesys, accessed 2026-09-24}
}