sip_ua

Description:

sip ua

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 #

iOS
Android
Web
macOS
Windows
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.** {*;}
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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.