0.5.1

[0.5.1] - 2022.02.13

• chore: Fix compilation error for the web.

v0.5.0

[0.5.0] - 2022.02.08

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

v0.4.0

[0.4.0] - 2021.10.13

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

v0.3.9

[0.3.9] - 2021.09.27

• Upgrade flutter-webrtc to 0.6.8

v0.3.8

---

[0.3.8] - 2021.09.26

• Fix ice delay.
• Don't run ready if the session has been terminated (#226)
• Support IceRestart when IceStateDisconnected (#218)
• Add options to the hangup (#224)
• Adaptive when answering audio or video calls.

v0.3.7

[0.3.7] - 2021.08.24

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

v0.3.6

[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.

v0.3.5

[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).

v0.3.2

[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).

v0.3.1

[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)