sip_ua 0.5.0 sip_ua: ^0.5.0 copied to clipboard
A SIP UA stack for Flutter/Dart, based on flutter-webrtc, support iOS/Android/Destkop/Web.
Changelog #
[0.5.0] - 2022.02.08
- Null safety.
- Bump version for flutter-webrtc.
[0.4.0] - 2021.10.13
- Add extended header support (#235)
- Add iceGatheringTimeout for UaSettings.
[0.3.9] - 2021.09.27
- Upgrade flutter-webrtc to 0.6.8
[0.3.8] - 2021.09.26
- Fix ice delay.
- Don't run ready if session has been terminated (#226)
- Support IceRestart when IceStateDisconnected (#218)
- Add options to the hangup (#224)
- Adaptive when answering audio or video calls.
[0.3.7] - 2021.08.24
- Fix the issue that unified-plan's onTrack does not call back AudioTrack.
- Export PeerConnection for call.
[0.3.6] - 2021.08.24
- Support custom MediaStream for call/answer.
- Fix the exception caused by speaker operation in web mode.
- bump dependencies (#216)
- Fix the parameters with double quotes in the Authentication header, and the unknown parameters are saved to auth_params.
- updated crypto and uuid versions (#188)
- Update dependency sdp_transform to ^0.3.0
- Fixed mute audio for unified-plan
- Add remote_has_audio/video method for Call.
- Configuring via_transport.
[0.3.5] - 2021.02.03
- Upgrade flutter-webrtc to 0.5.8.
- Set sdpSemantics (plan-b or unfied-plan) to unfied-plan by default.
- Add correct transport param to contact uri. close #161, close #160.
- Let the user override the call options by extending SIPUAHelper (#170).
[0.3.4] - 2021.01.08
- fix bug.
- Check Content-Length loosely.
- [example] 🐛 makes sure speaker is off to match UI state
[0.3.3] - 2020.11.27
- Fix uri parse.
- Upgrade flutter_webrtc to 0.5.7.
[0.3.2] - 2020.11.11
- Added dtmf options to Call (#154)
- Fix bug for digest authentication.
- Fix rport parse (#144).
- Support RFC2833.
- Upgrade flutter_webrtc to 0.4.1.
- Fix incorrect register assert (#139).
[0.3.1] - 2020.10.18
- fix rport in Via parser.
[0.3.0] - 2020.10.18
- Upgrade flutter_webrtc to 0.4.0
- Get more pub points (#138)
- Fix Uri.parse
- Force use case sensitivity in Websocket Upgrade to be compatible with old SIP servers
- Expose Register Expires setting and if Register at all (Thanks ghenry@SureVoIP)
- extraContactUriParams now working and tested against OpenSIPS 3.1 that has RFC8599 support (Thanks ghenry@SureVoIP)
[0.2.4] - 2020.08.25
- Add missing key field
Sec-WebSocket-Protocol
.
[0.2.3] - 2020.08.25
- Add display_name for Call.
- Add WebSocketSettings.
- Fix the invalid extraHeaders in Registrator.
- Exposed local_identity for Call.
- Fixed Sec-WebSocket-Key keys are not 24 bytes.
[0.2.2] - 2020.07.16
- Refactor call API, move answer, hangup, hold etc methos to Call class.
- Add SIP message listener to listen the new incoming SIP text message.
- Expose ha1 in UaSettings.
[0.2.1] - 2020.06.12
- Add UnHandledResponse for registrationFailed.
- Add allowBadCertificate for UaSettings.
- Upgrade recase and logger.
[0.2.0] - 2020.05.27
- Fixed bug for incoming call.
- Just wait for 3 seconds for ice gathering.
- Upgrade flutter-webrtc version to 0.2.8.
- Prevent sharing of config between different UA instances.
[0.1.0] - 2019.12.13
- Initial release.