Phone
Phone is RocketAiFlow's browser-based WebRTC client.
Use it to register a SIP/WebRTC user, place and receive calls from the platform, and run controlled tests before changing inbound routes or outbound campaigns.
Use Phone as a WebRTC client
Phone can register against different Asterisk extensions or SIP users by using the credentials configured for that account.
After registration, you can receive incoming calls, dial numbers from the keypad, monitor transport/SIP/call state, adjust remote audio volume, and enable auto-answer when you want the browser client to answer incoming calls automatically.
- Full Name, User Name, and Password identify the SIP/WebRTC user to register.
- Connect starts the browser registration.
- The status badges show transport, SIP registration, and current call state.
- The keypad is used to place manual outbound calls.
- Auto-answer incoming calls can be enabled for controlled tests where the browser client should answer without manual action.
Test an AI voice agent from Phone
Voice agent test mode lets you pick an available AI voice agent and start a direct test call from the browser.
This is useful for checking inbound-style behavior before attaching the agent to a real inbound trunk route. It validates the agent conversation, prompt, voice, transfer, and hangup behavior without depending on external trunk routing.
- select the agent you want to test
- start the call from Phone
- verify that the agent answers and follows the expected workflow
- review the result in Inbound Call Records after the call completes
What to validate
Use Phone for small, repeatable tests. Change one setting, place one call, review the result, then continue.
- SIP/WebRTC registration succeeds with the selected user
- incoming and outgoing calls work from the browser
- microphone and remote audio are working
- auto-answer behaves as expected when enabled
- the selected AI voice agent responds correctly
- transfer, hangup, transcript, timing, and Inbound Call Record data match the expected result