Root cause of the intermittent dead-mic / no-audio / late-speaker: LiveKit's
automatic AVAudioSession config races CallKit's activation. The fix needs the
2.15+ audio API, which the build wasn't getting ('~> 2.0' resolved an older 2.x
from a stale spec cache). So:
- NativeCall.podspec: pin LiveKitClient '~> 2.15'.
- codemagic.yaml: 'pod install --repo-update' so the spec repo knows 2.15.x.
- Plugin: disable LiveKit auto audio-session config + keep the engine OFF; in
CXProvider didActivate set the session category and enable the engine; in
didDeactivate disable it. Request mic permission on connect so enabling the
engine in didActivate doesn't block on undetermined permission.
All AudioManager APIs verified against the raw 2.15.3 source (setEngineAvailability,
AudioEngineAvailability.default/.none, audioSession.isAutomaticConfigurationEnabled).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
The build failed: 'AudioManager' has no member 'setEngineAvailability' — that API
is only on unreleased/main docs, not in the resolved LiveKitClient 2.15.3. Use the
API that actually ships (per audio.md): disable LiveKit's automatic AVAudioSession
configuration (AudioManager.shared.audioSession.isAutomaticConfigurationEnabled =
false) and configure the session ourselves in CXProvider didActivate. Removed all
setEngineAvailability(.none/.default) calls.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Foundation for native LiveKit calling: the LiveKit iOS SDK carries the call media
natively and coordinates its audio engine with CallKit's AVAudioSession (AudioManager
.setEngineAvailability on didActivate/didDeactivate) — the thing WebKit's WebRTC can't
do. With this, the CallKit call stays active (stable ring + lock-screen answer +
background) AND the mic works. Native connection code + server LiveKit token in the VoIP
payload + WebView coordination come next. Don't build yet.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
The native call feature (iOS). Backward-compatible: without the plugin (current
builds) nativeCallOn() is false and every CallKit branch is skipped, so web/older
builds behave exactly as before.
Native (mobile/plugins/native-call, a local Capacitor plugin like audio-route):
- PushKit: registers for VoIP pushes, reports the VoIP token to JS (-> /api/v1/devices
'ios-voip'). On an incoming VoIP push, reports a CallKit incoming call (full-screen
ring, works when the app is force-killed).
- CallKit: answer/decline/end -> events to JS; configures the call AVAudioSession on
didActivate so the WebView's WebRTC audio rides a call-priority session (background).
- Outgoing calls register with CallKit too (reportOutgoingCall) so they get the same
active-call background-audio context.
- NativeCall.podspec (frameworks CallKit/PushKit/AVFoundation); added to mobile deps;
ios-patch.sh now sets UIBackgroundModes = [audio, voip] (voip required for PushKit).
Web bridge (home.html): setupNativeCall() registers the VoIP token, joins on CallKit
answer, leaves/declines on CallKit end; on CallKit devices the in-app call-invite popup
+ WebAudio ring are suppressed (the system rings instead); outgoing calls are reported
to CallKit; call-end events dismiss the CallKit call. calls.js threads a stable call
uuid through the dm-call/group-call WS events + start responses so both sides can match
the CallKit call.
Needs a Codemagic build to compile the plugin; first on-device iteration expected.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>