Update README.md

Author: cloudwebrtc

README.md

@@ -3,7 +3,7 @@ A dart-lang version of the SIP UA stack, ported from [JsSIP](https://github.com/
 
 ## Overview
 - Use pure [dart-lang](https://dart.dev)
-- SIP over WebSocket (use real SIP in your [flutter/web](https://flutter.dev) apps)
+- SIP over WebSocket (use real SIP in your flutter mobile, [desktop](https://flutter.dev/desktop), [web](https://flutter.dev/web) apps)
 - Audio/video calls ([flutter-webrtc](https://github.com/cloudwebrtc/flutter-webrtc)) and instant messaging
 - Support with standard SIP servers such as OpenSIPS, Kamailio, Asterisk and FreeSWITCH.