Voice - WebRTC FAQ
Does RingCentral WebRTC Support HD Voice?
Does WebRTC include Call Control API support?
Yes. With WebRTC you can take control of the call with the following capabilities: initiate call, accept incoming call, adding callees via DTMF, hold / unhold, mute / unmute, park, flip, transfer, forward calls, start / stop recording and barge/whisper. These are documented in the
ringcentral-web-phone WebRTC SDK.
Can I set Caller ID with WebRTC?
Yes. You can set Caller ID (CLID) using the
ringcentral-web-phone WebRTC SDK. Just set the desired number as the
fromNumber in the
webPhone.userAgent.invite() method call.
Can WebRTC support area code matching for Caller ID?
Yes. You will need to supply your own area-code matching algorithm, but as long as you have matching numbers for use in your account, you can set the CLID as you wish. One approach is to store your area-code CLID numbers as Auto-Receptionist Company Numbers.
Are WebRTC call events captured via the event system?
Yes. Voice calls via WebRTC, RingOut and RingCentral endpoints are all captured via the event system. To check the presence status of an extension, you can call the extension presence API endpoint or subscribe to presence events via the subscription API or webhoks API.
Is WebRTC traffic encrypted?
Yes. All WebRTC communication using RingCentral is encrypted in transit.
Can a single browser-based app support both RingOut and WebRTC?
What is the "VoIP Calling" Permission used for?
VoIP Calling is an app permission which is needed by WebRTC. You can refer to details of RingCentral's WebRTC implementation in RingCentral GitHub repo for the WebRTC SDK: https://github.com/ringcentral/ringcentral-web-phone. When you are building an app using the RingCentral WebRTC SDK, please make sure your app has
VoIP Calling permission.