dart-sip-ua

A dart-lang version of the SIP UA stack, ported from JsSIP.

Overview

• Use pure dart-lang
• SIP over WebSocket (use real SIP in your flutter mobile, desktop, web apps)
• Audio/video calls (flutter-webrtc) and instant messaging
• Support with standard SIP servers such as OpenSIPS, Kamailio, Asterisk and FreeSWITCH.
• Support RFC2833 or INFO to send DTMF.

Currently supported platforms

• X iOS
• X Android
• X Web
• X macOS
• X Windows
• X Linux
• Fuchsia

Install

Android

• Proguard rules:

-keep class io.flutter.app.** { *; }
-keep class io.flutter.plugin.**  { *; }
-keep class io.flutter.util.**  { *; }
-keep class io.flutter.view.**  { *; }
-keep class io.flutter.**  { *; }
-keep class io.flutter.plugins.**  { *; }

-keep class com.cloudwebrtc.webrtc.** {*;}
-keep class org.webrtc.** {*;}

Quickstart

Run example:

• dart-sip-ua-example
• or add your example.

Register with SIP server:

• Asterisk
• FreeSWITCH
• OpenSIPS
• Kamailio
• or add your server example.

NOTE

Thanks to the original authors of JsSIP for providing the JS version, which makes it possible to port the dart-lang.

• José Luis Millán
• Iñaki Baz Castillo
• Saúl Ibarra Corretgé

Sponsors

The first version was sponsored by Suretec Systems Ltd. T/A SureVoIP.

Contributing

The project is inseparable from the contributors of the community.

• SureVoIP - Sponsor
• CloudWebRTC - Original Author
• Robert Sutton - Contributor
• Gavin Henry - Contributor
• Perondas - Contributor

License

dart-sip-ua is released under the MIT license.