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}\n-keep class io.flutter.plugin.**  { *; }\n-keep class io.flutter.util.**  { *; }\n-keep class io.flutter.view.**  { *; }\n-keep class io.flutter.**  { *; }\n-keep class io.flutter.plugins.**  { *; }\n\n-keep class com.cloudwebrtc.webrtc.** {*;}\n-keep class org.webrtc.** {*;}\n```\n\n## Quickstart\n\nRun example:\n\n- [dart-sip-ua-example](https://github.com/flutter-webrtc/dart-sip-ua/blob/master/example/README.md)\n- or add your example.\n\nRegister with SIP server:\n\n- [Asterisk](https://github.com/flutter-webrtc/dockers/tree/main/asterisk)\n- FreeSWITCH\n- OpenSIPS\n- 3CX\n- Kamailio\n- or add your server example.\n\n## FAQ's OR ISSUES\n\u003cdetails\u003e\n\n\u003csummary\u003eexpand\u003c/summary\u003e\n\n## Server not configured for DTLS/SRTP\n\nWEBRTC_SET_REMOTE_DESCRIPTION_ERROR: Failed to set remote offer sdp: Called with SDP without DTLS fingerprint.\n\nYour server is not sending a DTLS fingerprint inside the SDP when inviting the sip_ua client to start a call.\n\nWebRTC 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.\n\n\n## Why isn't there a UDP connection option?\n\nThis package uses a WS or TCP connection for the signalling processs to initiate or terminate a session (sip messages).\nOnce the session is connected WebRTC transmits the actual media (audio/video) over UDP.\n\nIf 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.\n\n## SIP/2.0 488 Not acceptable here\n\nThe codecs on your PBX server don't match the codecs used by WebRTC\n\n- **opus** (payload type 111, 48kHz, 2 channels)\n- **red** (payload type 63, 48kHz, 2 channels)\n- **G722** (payload type 9, 8kHz, 1 channel)\n- **ILBC** (payload type 102, 8kHz, 1 channel)\n- **PCMU** (payload type 0, 8kHz, 1 channel)\n- **PCMA** (payload type 8, 8kHz, 1 channel)\n- **CN** (payload type 13, 8kHz, 1 channel)\n- **telephone-event** (payload type 110, 48kHz, 1 channel for wideband, 8000Hz, 1 channel for narrowband)\n\n\u003c/details\u003e\n\n\n## NOTE\nThanks 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).\n- [José Luis Millán](https://github.com/jmillan)\n- [Iñaki Baz Castillo](https://github.com/ibc)\n- [Saúl Ibarra Corretgé](https://github.com/saghul)\n\n## Sponsors\nThe first version was sponsored by Suretec Systems Ltd. T/A [SureVoIP](https://www.surevoip.co.uk).\n\n## Contributing\nThe project is inseparable from the contributors of the community.\n- [SureVoIP](https://github.com/SureVoIP) - Sponsor\n- [CloudWebRTC](https://github.com/cloudwebrtc) - Original Author\n- [Robert Sutton](https://github.com/rlsutton1) - Contributor\n- [Gavin Henry](https://github.com/ghenry) - Contributor\n- [Perondas](https://github.com/Perondas) - Contributor\n- [Mikael Wills](https://github.com/mikaelwills) - Contributor\n\n## License\ndart-sip-ua is released under the [MIT license](https://github.com/cloudwebrtc/dart-sip-ua/blob/master/LICENSE).\n","project_url":"https://awesome.ecosyste.ms/api/v1/projects/github.com%2Fflutter-webrtc%2Fdart-sip-ua","html_url":"https://awesome.ecosyste.ms/projects/github.com%2Fflutter-webrtc%2Fdart-sip-ua","lists_url":"https://awesome.ecosyste.ms/api/v1/projects/github.com%2Fflutter-webrtc%2Fdart-sip-ua/lists"}