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Category | BYOC Cloud/Genesys Cloud Voice | BYOC Premises | Opus | Registration & Failover | Provisioning & Mutual Authentication | TLS & SRTP |
|---|---|---|---|---|---|---|
| Managed | Y† | Y | Y† | Y | Y | Y |
| Unmanaged | Y | Y | Y† | Y‡ | N/A | Y§ |
| Remote | Y* | Y | N | N/A | N/A | N/A∝ |
| WebRTC | Y | Y¶ | Y | Y¶ | N/A | Yβ |
* Can incur additional toll charges.
† Depends on the phone model. For specific details on supported phone models, see Managed phones: models and features matrix.
‡ Failover setup can be configured with unmanaged phones if the phone supports it. Managed phones provide this by default without direct phone configuration.
§ While possible to configure TLS/SRTP, certificate exchange can be a challenge to manage.
¶ There is no registration for WebRTC phones. In an LDM environment, if the Internet connection is lost, you can not make, receive, or transfer calls.
∝ While under most carriers, a remote phone uses the UDP/RTP protocol, depending on the trunk settings, the protocol can be TLS/SRTP.
β While Genesys Cloud’s WebRTC implementation uses TLS and SRTP for signaling and media security, the security model is different from regular VoIP. More specifically, WebRTC uses SRTP to secure the media stream and Genesys Cloud uses TLS to secure the signaling. Furthermore, WebRTC does not transport the SRTP keying material in the SIP messages, but uses DTLS for key establishment. As such, WebRTC is more secure because even if the signaling traffic were captured or logged, the audio stream could not be decrypted.Get user feedback about articles.