https://github.com/flutter-webrtc/dart-sip-ua
A dart-lang version of the SIP UA stack.
https://github.com/flutter-webrtc/dart-sip-ua
client dartlang flutter jssip sip voip
Last synced: 24 days ago
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A dart-lang version of the SIP UA stack.
- Host: GitHub
- URL: https://github.com/flutter-webrtc/dart-sip-ua
- Owner: flutter-webrtc
- License: mit
- Created: 2019-08-17T23:11:49.000Z (almost 6 years ago)
- Default Branch: main
- Last Pushed: 2025-04-01T12:43:38.000Z (2 months ago)
- Last Synced: 2025-04-14T12:18:48.352Z (about 2 months ago)
- Topics: client, dartlang, flutter, jssip, sip, voip
- Language: Dart
- Homepage:
- Size: 1.5 MB
- Stars: 350
- Watchers: 28
- Forks: 286
- Open Issues: 152
-
Metadata Files:
- Readme: README.md
- Changelog: CHANGELOG.md
- Funding: .github/FUNDING.yml
- License: LICENSE
Awesome Lists containing this project
README
# dart-sip-ua
[](https://opencollective.com/flutter-webrtc) [](https://pub.dartlang.org/packages/sip_ua) [](https://join.slack.com/t/flutterwebrtc/shared_invite/zt-q83o7y1s-FExGLWEvtkPKM8ku_F8cEQ)
A dart-lang version of the SIP UA stack, ported from [JsSIP](https://github.com/versatica/JsSIP).
## Overview
- Use pure [dart-lang](https://dart.dev)
- SIP over WebSocket && TCP (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, 3CX 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](https://github.com/flutter-webrtc/dart-sip-ua/blob/master/example/README.md)
- or add your example.
Register with SIP server:
- [Asterisk](https://github.com/flutter-webrtc/dockers/tree/main/asterisk)
- FreeSWITCH
- OpenSIPS
- 3CX
- Kamailio
- or add your server example.
## FAQ's OR ISSUES
expand
## Server not configured for DTLS/SRTP
WEBRTC_SET_REMOTE_DESCRIPTION_ERROR: Failed to set remote offer sdp: Called with SDP without DTLS fingerprint.
Your server is not sending a DTLS fingerprint inside the SDP when inviting the sip_ua client to start a call.
WebRTC uses encryption by Default, all WebRTC communications (audio, video, and data) are encrypted using DTLS and SRTP, ensuring secure communication. Your PBX must be configured to use DTLS/SRTP when calling sip_ua.
## Why isn't there a UDP connection option?
This package uses a WS or TCP connection for the signalling processs to initiate or terminate a session (sip messages).
Once the session is connected WebRTC transmits the actual media (audio/video) over UDP.
If anyone actually still wants to use UDP for the signalling process, feel free to submit a PR with the large amount of work needed to set it up, packet order checking, error checking, reliability timeouts, flow control, security etc etc.
## SIP/2.0 488 Not acceptable here
The codecs on your PBX server don't match the codecs used by WebRTC
- **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)
## NOTE
Thanks to the original authors of [JsSIP](https://github.com/versatica/JsSIP) for providing the JS version, which makes it possible to port the [dart-lang](https://dart.dev).
- [José Luis Millán](https://github.com/jmillan)
- [Iñaki Baz Castillo](https://github.com/ibc)
- [Saúl Ibarra Corretgé](https://github.com/saghul)
## Sponsors
The first version was sponsored by Suretec Systems Ltd. T/A [SureVoIP](https://www.surevoip.co.uk).
## Contributing
The project is inseparable from the contributors of the community.
- [SureVoIP](https://github.com/SureVoIP) - Sponsor
- [CloudWebRTC](https://github.com/cloudwebrtc) - Original Author
- [Robert Sutton](https://github.com/rlsutton1) - Contributor
- [Gavin Henry](https://github.com/ghenry) - Contributor
- [Perondas](https://github.com/Perondas) - Contributor
- [Mikael Wills](https://github.com/mikaelwills) - Contributor
## License
dart-sip-ua is released under the [MIT license](https://github.com/cloudwebrtc/dart-sip-ua/blob/master/LICENSE).