vobiz_webrtc 0.1.0
vobiz_webrtc: ^0.1.0 copied to clipboard
Flutter SDK for making and receiving calls on the Vobiz platform using SIP-over-WebSocket and WebRTC. Includes automatic SIP registration, WebRTC audio setup, and call state management.
Changelog #
All notable changes to this project will be documented in this file.
The format is based on Keep a Changelog, and this project adheres to Semantic Versioning.
0.1.0 - 2026-04-29 #
Added #
-
Initial release of Vobiz WebRTC SDK for Flutter
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SIP Features:
- WebSocket-based SIP registration with digest authentication
- Automatic SIP keep-alive (OPTIONS every 60 seconds)
- Support for REGISTER, INVITE, ACK, BYE, CANCEL, OPTIONS methods
- CSeq tracking and dialog state management
- INVITE retransmit handling
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Call Features:
- Outbound calling to PSTN numbers
- Inbound call detection and routing
- Call state machine (idle → calling → ringing → inCall → ended)
- Answer/Reject call handling
- Mute/Unmute microphone control
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WebRTC Features:
- Automatic peer connection management
- Audio-only media streams
- ICE candidate collection and management
- SDP offer/answer negotiation
- Automatic ICE gathering
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Network Robustness:
- Auto-reconnect on WebSocket drop
- SIP registration refresh every 55 minutes
- ICE gathering with 8-second timeout
- Connection state tracking
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Platform Support:
- Android 5.0+ (API 21+)
- iOS 11.0+
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Documentation:
- Quick-start guide
- Platform setup instructions (Android & iOS)
- API reference
- Answer URL contract specification
- Troubleshooting guide
Known Limitations #
- Push notifications for background call wake-up not implemented yet
- No conference bridge (1:1 calls only)
- TURN server configuration optional (Google STUN only by default)
Upcoming #
0.2.0 Planned #
- Push notification support (APNs & FCM)
- Conference bridge
- Call recording
- Better reconnection logic