AbsoluteFlow - WebRTC Dialer

WebRTC Dialer

SIP Settings

Browser → FreeSWITCH → Kamailio

Server: sys_main

FreeSWITCH account

Disconnected

WebSocket URL

SIP Domain (Realm)

SIP ID / Extension

Password

STUN / TURN settings

ICE servers (JSON) Use your network’s TURN service if direct audio cannot connect.

* Microphone access must be granted by the browser. Open this page over HTTPS. Passwords are used only for SIP registration and are not saved by the dialer.

ABC

DEF

GHI

JKL

MNO

PQRS

TUV

WXYZ

Enter your FreeSWITCH account to register.

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