OpenV WebRTC Phone
Modern in-browser SIP client for Asterisk (PJSIP + WSS)
Disconnected
Account
Test
Clear
WebSocket WSS URL (Asterisk/PJSIP)
SIP Username
Password
Domain / Realm
Display Name (optional)
Microphone
Speakers
Register
Unregister
Dialer
Ready
Call
Hangup
Answer
Hold
Mute
Video
Xfr
Rec
Duration
00:00
Ring / Progress
—
DTMF
—
Keypad
Close
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