sip_ua_livekit_webrtc 1.0.1 copy "sip_ua_livekit_webrtc: ^1.0.1" to clipboard
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A SIP UA stack for Flutter/Dart, based on flutter-webrtc, support iOS/Android/Destkop/Web.

Changelog #


[1.0.0] - 2024.08.24

  • allow to change UA uri in runtime (#425)
  • Overriding intervals for recovery connection (#472)
  • TcpSocket: Implement isConnecting & url (#464)
  • Uri configuration for call operation (#448)
  • Update add-line to python3 (#466)
  • Fixed work of calls on webkit browsers (#444)
  • Upgrade to video call implementation and dark mode (#462)
  • Add example apps. (#457)
  • Remove text media channels in SDP offers (#461)
  • set transport value using socket (#449)
  • Add sendInterval to dtmf (#443)
  • Feature/expose ice transport policy (#441)

[0.6.0] - 2024.05.08

[0.5.8] - 2023.05.11

  • Allow await on helper start call (#365)
  • Adding params support for sendMessage (#366)
  • Set intl version to the one used by flutter

[0.5.7] - 2023.05.11

  • Add sendMessage to Call
  • Bump version of intl

[0.5.6] - 2023.04.17

  • Reverted version constraint on intl
  • Bumped version of flutter_webrtc

[0.5.5] - 2023.03.08

  • Bump version for intl & lints
  • Update websocket_web_impl.dart (#345)
  • fix(hangup): set cancel reason nullable (#346)
  • Add sdp transformers (#350)
  • Hold fix (#351)

[0.5.4] - 2023.02.20

  • Bump version for flutter-webrtc
  • Fixed error handling in _receiveInviteResponse #344
  • Updated logger #342
  • Websocket message queue using streams and Delay between messages #335
  • Fixed bugs in message.dart & rtc_session.dart #332
  • Allow setting ice_gathering_timeout option #330
  • Add realm option to SIPUAHelper settings #331
  • Code quality #326

[0.5.3] - 2022.10.19

  • Bump version for flutter-webrtc
  • Fix nullability in subscriber
  • Fix flutter test
  • Fix subscription parsing grammar
  • Added ability to supply custom logger
  • Added ability to get call statistics

[0.5.2] - 2022.08.05

  • chore: Fix hold/unhold.

[0.5.1] - 2022.02.13

  • chore: Fix compilation error for web.

[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.
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A SIP UA stack for Flutter/Dart, based on flutter-webrtc, support iOS/Android/Destkop/Web.

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Documentation

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License

MIT (license)

Dependencies

collection, crypto, flutter_livekit_webrtc, intl, logger, path, random_string, recase, sdp_transform, text, uuid

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