baresip_flutter 0.2.1

A Flutter plugin for SIP VoIP calls using the BareSip SDK. Supports registration, incoming/outgoing calls, audio routing, and real-time event streaming on Android.

0.2.1

Fix

• Fixed BareSipSdk-release.aar resolution when using pub.dev package — now looks for AAR in host app's android/app/libs/ first, then falls back to plugin's own libs

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Bug Fixes & Improvements

Audio Transmission Fixed:

• Fixed audio not transmitting to remote party (one-way audio issue)
• Fixed SDP advertising private IP — server RTP proxy now handles NAT traversal
• Fixed codec registration: normalized codec names to full baresip format (PCMU/8000/1, PCMA/8000/1, opus/48000/2, G722/16000/1)
• Fixed double audio focus acquisition causing AUDIOFOCUS_LOSS_TRANSIENT interruption

Stability Fixes:

• Fixed SIGSEGV crash when hangup() called after remote BYE (double ua_hangup on freed pointer)
• Fixed call_destroy causing pthread_mutex_lock on destroyed mutex (SIGABRT) — baresip owns call lifecycle on call closed event
• Fixed conf_configure failure (writing /config: Is a directory) when service restarts after app kill
• Fixed incoming call events silently dropped due to event_buf[256] overflow with long hostnames — increased to 2048 bytes
• Fixed SIP trace logging thread buffer truncating SDP lines — increased from 128 to 2048 bytes
• Added null pointer guards in ua_hangup and call_destroy JNI to prevent native crashes

Call Flow Improvements:

• Correct baresip API sequence: ua_call_alloc + call_connect for outgoing (not ua_connect)
• call_start_audio called at correct time (on ESTABLISHED event, not before)
• ended flag on SdkCallState prevents double hangup operations
• onCallClosed never calls call_destroy — baresip already owns the call memory
• Single audio focus acquisition per call (guarded against re-acquisition)

Configuration:

• Removed invalid config keys that caused conf_configure failures
• Removed non-existent .so module directives (all modules statically linked)
• Static module registration via mod_add() for codecs, audio, NAT modules
• Log level now correctly uses SdkConfig.logLevel

Breaking Changes:

• None

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0.1.0

Initial Release

Features:

• ✅ SIP registration and authentication
• ✅ Outgoing and incoming call support
• ✅ Real-time event streaming (registration, call state, audio route, network, errors)
• ✅ Call controls (answer, reject, hangup, mute, hold)
• ✅ Audio routing (earpiece, speaker, wired headset, Bluetooth)
• ✅ Permission checking helper
• ✅ Foreground service for background operation
• ✅ Automatic registration retry (5 attempts, 30s intervals)
• ✅ Service persistence after app kill
• ✅ Comprehensive error handling

Platform Support:

• Android: ✅ (minSdk 29 / Android 10+)
• iOS: ❌ (not supported)

Requirements:

• Flutter SDK: 3.19.0+
• Dart SDK: 3.3.0+
• Android compileSdk: 34+
• Kotlin: 2.2.x
• BareSip SDK AAR (included in setup)

Known Limitations:

• Android only
• Requires BareSip SDK AAR in host app's libs folder
• Service stops when app is killed (use credential persistence for auto-login)

Documentation:

• Comprehensive README with setup guide
• API reference with all methods and events
• Example app demonstrating all features
• Troubleshooting section

Breaking Changes:

• None (initial release)