dart-sip-ua

dart-sip-ua is a Dart implementation of the SIP User Agent (UA) stack, ported from JsSIP. It allows for SIP over WebSocket and TCP, enabling real SIP functionality in Flutter mobile, desktop, and web applications.

Key features include:

• Audio/Video Calls: Integrated with flutter-webrtc.
• Instant Messaging: Support for SIP-based messaging.
• Standard Server Support: Works with OpenSIPS, Kamailio, Asterisk, 3CX, and FreeSWITCH.
• DTMF Support: Supports RFC2833 or INFO for sending DTMF tones.

In dart-sip-ua, the connection is split into two layers:

1. Signalling: Uses WebSocket (WS) or TCP to initiate and terminate sessions (SIP messages).
2. Media: Once the session is established, WebRTC transmits the actual audio/video data over UDP.

Note that the package does not support UDP for the signalling process itself.

To run the example project, you must ensure your Flutter SDK is using the dev channel. Follow these steps to set up and run the project:

1. Switch to the dev channel: flutter channel dev
2. Run the project creation script: ./scripts/project_tools.sh create
3. Run the application: flutter run

To change the launch image for the iOS version of your application, you can either replace the image files directly in the directory or use Xcode.

Method 1: Direct replacement Replace the existing image files in example/ios/Runner/Assets.xcassets/LaunchImage.imageset/ with your own assets.

Method 2: Using Xcode

1. Open the iOS project in Xcode by running open ios/Runner.xcworkspace from your terminal.
2. In the Xcode Project Navigator, navigate to Runner/Assets.xcassets.
3. Drag and drop your desired images into the asset catalog to replace the existing launch images.

If you prefer not to manage legacy Dart 1.x versions locally, you can use Docker to generate the grammar parser.

Docker Generation Steps

1. Install Docker on your system.
2. Clone the repository and run make commands:

   git clone https://github.com/flutter-webrtc/dart-sip-ua
   cd dart-sip-ua
   make peg && make grammar

When building for Android, you must add specific Proguard rules to prevent the obfuscation of Flutter and WebRTC classes, which is necessary for the plugin to function correctly.

-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.** {*;}

If you are targeting Desktop or Web platforms, use the following commands to run the example project:

• macOS: flutter run -d macos
• Web: flutter run -d web or flutter run -d chrome

flutter run -d macos
flutter run -d web|chrome

The grammar_parser.dart file is generated from grammar.peg using the peg tool. Because the peg tool is incompatible with Dart 2.x, you must use Dart 1.24.3 to perform the generation manually.

Manual Generation Steps

1. Install Dart 1.24.3:

   brew install dart@1
   export PATH="/usr/local/opt/dart@1/bin:$PATH"

2. Setup the peg tool:

   cd tools
   git clone https://github.com/cloudwebrtc/peg
   cd peg
   pub get

3. Run the generation script:

   ./tool/generate_grammar.sh

# Manual generation sequence
brew install dart@1
export PATH="/usr/local/opt/dart@1/bin:$PATH"
cd tools
git clone https://github.com/cloudwebrtc/peg
cd peg
pub get
./tool/generate_grammar.sh

The sockets parameter in Settings is mandatory and must be a non-empty list of SIPUASocketInterface. The configuration logic supports several ways to define sockets:

• Single Socket: sockets: socket (Note: The validation logic expects a List, so ensure it is wrapped in []).
• List of Sockets: sockets: [socket1, socket2]
• List of Objects with weights: sockets: [{socket: socket1, weight: 1}, {socket: socket2, weight: 0}]
• Mixed List: sockets: [{socket: socket1}, socket2]

If the sockets list is empty or not a list, a ConfigurationError will be thrown.

// Example of providing a list of sockets
settings.sockets = [
  myWebSocketSocket,
  myOtherSocket,
];

The error SIP/2.0 488 Not acceptable here typically indicates a codec mismatch between your PBX server and WebRTC.

WebRTC uses the following codecs:

• opus (payload type 111, 48kHz, 2 channels)
• red (payload type 63, 48kHz, 2 channels)
• G722 (payload type 9, 8kHz, 1 channel)
• ILBC (payload type 102, 8kHz, 1 channel)
• PCMU (payload type 0, 8kHz, 1 channel)
• PCMA (payload type 8, 8kHz, 1 channel)
• CN (payload type 13, 8kHz, 1 channel)
• telephone-event (payload type 110, 48kHz, 1 channel for wideband, 8000Hz, 1 channel for narrowband)

If you encounter the error WEBRTC_SET_REMOTE_DESCRIPTION_ERROR: Failed to set remote offer sdp: Called with SDP without DTLS fingerprint, it means your SIP server is not sending a DTLS fingerprint inside the SDP during the invite.

Because WebRTC requires encryption by default (using DTLS and SRTP), your PBX must be explicitly configured to use DTLS/SRTP when communicating with the sip_ua client.