sip_ua

What's Changed

• make sure the session terminates by @victortive in #485
• Hold and video upgrade fixed by @mikaelwills in #503
• sip_ua_helper: add sendOptions by @eschmidbauer in #476
• Update callscreen.dart by @HVaidehi in #427
• Rethrow errors by @victortive in #505

New Contributors

• @eschmidbauer made their first contribution in #476
• @HVaidehi made their first contribution in #427

Full Changelog: v1.0.0...v1.0.1

What's Changed

• Feature/expose ice transport policy by @komaxx in #441
• Add sendInterval to dtmf by @hiratake55 in #443
• Set transport value using socket via_transport by @Volsavr in #449
• Update sip_ua.dart by @NicolasGodfather in #456
• Remove text media channels in SDP offers by @victortive in #461
• Add example apps. by @cloudwebrtc in #457
• Upgrade to video call implementation and dark mode 2 by @mikaelwills in #462
• Fixed work of calls on webkit browsers by @nikzor in #444
• Uri configuration for call operation by @Volsavr in #448
• Update add-line to python3 by @victortive in #466
• TcpSocket: Implement isConnecting & url by @victortive in #464
• Overriding intervals for recovery connection by @Volsavr in #472
• allow to change UA uri in runtime by @furmur in #425

New Contributors

• @Volsavr made their first contribution in #449
• @NicolasGodfather made their first contribution in #456
• @victortive made their first contribution in #461
• @nikzor made their first contribution in #444
• @furmur made their first contribution in #425

Full Changelog: 0.6.0...v1.0.0

What's Changed

• Add status code to CallState on the progress event by @hiratake55 in #374
• expose rtc_session to/from-tag by @ftsef in #378
• #376: Added _settings.session_timers_refresh_method to UaSettings t… by @komaxx in #377
• expose instance_id by @VictorUvarov in #395
• Example app cleanup by @VictorUvarov in #394
• fix: parse expires from string header. by @cloudwebrtc in #396
• Add support for UAConfiguration registrar_server by @VictorUvarov in #397
• Feature/update dependency to intl v19 by @komaxx in #414
• Tcp socket implementation by @mikaelwills in #416
• feat: Adding extraHeaders in RegisterParams, exposing existing functionality via UaSettings() by @luan-evangelista in #432
• add contact_uri as parameter by @pjalbuquerque in #426
• Release/0.6.0 by @cloudwebrtc in #437

New Contributors

• @hiratake55 made their first contribution in #374
• @ftsef made their first contribution in #378
• @komaxx made their first contribution in #377
• @VictorUvarov made their first contribution in #395
• @mikaelwills made their first contribution in #416
• @luan-evangelista made their first contribution in #432
• @pjalbuquerque made their first contribution in #426

Full Changelog: 0.5.8...0.6.0

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

• Add sendMessage to Call
• Bump version of intl

• 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 #337
• 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

• chore: Fix hold/unhold.

• chore: Fix compilation error for web.

• Null safety.
• Bump version for flutter-webrtc.

• Add extended header support (#235)
• Add iceGatheringTimeout for UaSettings.

• Upgrade flutter-webrtc to 0.6.8

• 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.

• Fix the issue that unified-plan's onTrack does not call back AudioTrack.
• Export PeerConnection for call.

• 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.

• 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).

• fix bug.
• Check Content-Length loosely.
• [example] 🐛 makes sure speaker is off to match UI state

• Fix uri parse.
• Upgrade flutter_webrtc to 0.5.7.

• 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).

• fix rport in Via parser.

• 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)

• Add missing key field Sec-WebSocket-Protocol.

• 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.

• 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.

• Add UnHandledResponse for registrationFailed.
• Add allowBadCertificate for UaSettings.
• Upgrade recase and logger.

• 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.

• Initial release.