WebPhone
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Debug settings (effective)
SIP URI
—
SIP Domain
—
WSS URL
—
STUN enabled
—
TURN enabled
—
ICE servers (saved)
—
Debug log
—
Connection Settings
Server + WebRTC options used for login and calls
SIP + WebSocket (WSS)
These must match your PBX/WebRTC gateway provisioning.
WSS URL (preview)
—
ICE + RTP Policies
Controls how candidates are gathered and negotiated.
Advanced (Asterisk / chan_sip)
These mirror settings used by Browser-Phone to improve reachability and call setup.
ICE Servers (JSON)
Paste `RTCConfiguration.iceServers` (array of objects).
Example: [{"urls":["stun:stun.l.google.com:19302"]}]
ICE Diagnostics
Runs local ICE gathering and summarizes candidate types.
WebPhone
Ready
Offline
Mic: Unknown
Media: Idle
TURN: —
Make a Call
J
No Active Call
Enter a number and press Call to start
0:00
Mic
Sound
Keypad
During a call, taps send DTMF tones.
Call queue
0
No calls waiting.
Call History