iOS 13+ terminates the app if a PushKit VoIP push doesn't call reportNewIncomingCall
before completion(). My earlier "no re-ring blip" change made the cancel handler SKIP
the report for already-known/ended calls — which is exactly what iOS kills the app
for, surfacing as a crash after several calls (a cancel push for a prior call's UUID).
Revert to ALWAYS report then immediately end: for a still-active/ringing UUID the
report errors harmlessly (no second ring) and reportCall ends it; only a late duplicate
cancel shows a brief, unavoidable blip. A crash is far worse than a blip.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
The .voiceChat AVAudioSession mode defaults to the earpiece, so call audio came
out of the earpiece and only settled after mute/unmute toggling. Switch to
.videoChat + .defaultToSpeaker + allow Bluetooth so audio goes to the loudspeaker
by default while wired/BT headsets still win. Add preferSpeaker() (override to
speaker when on the built-in receiver) in didActivate AND after mic toggles (the
route can flip back to earpiece on unmute). Report the chosen route in telemetry.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
The LiveKitClient CocoaPod on trunk caps at 2.0.18 (2.1+ is SPM-only), so the
2.15 CallKit audio API was unreachable. BUT the repo still ships a valid podspec
at tag 2.15.3, and its deps (LiveKitWebRTC 144.7559.11, LiveKitUniFFI 0.0.6,
SwiftProtobuf) ARE on trunk. So instead of a risky SPM migration:
- ios-patch.sh injects `pod 'LiveKitClient', :git => <repo>, :tag => '2.15.3'`
into the generated Podfile (after cap sync, before pod install). The NativeCall
podspec's '~> 2.0' is satisfied by 2.15.3. Idempotent; hard-fails if it can't
find the App target so we never silently fall back to 2.0.18.
- Plugin re-adds the CallKit<->LiveKit audio-session coordination, now compilable:
auto-config OFF + engine OFF at load; configure session + enable engine in
didActivate; disable in didDeactivate; request mic permission on connect.
Core Room APIs (connect/disconnect/setMicrophone) verified compatible between
2.0.18 and 2.15.3 against the real source.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
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>
Second failure showed 'AudioManager' has no member 'audioSession' either — even
though the 2.15.3 source has both audioSession and setEngineAvailability. So the
build's CocoaPods is resolving an OLDER 2.x (stale spec repo/cache) that predates
the engine-observer audio API. Rather than keep guessing, drop ALL LiveKit
AudioManager audio-session code and keep the last known-good audio behavior
(LiveKit defaults). The valuable fixes that use only CallKit/AVFoundation stay:
re-ring blip guard, CallKit<->UI mute sync, and the WS reportIncomingCall ring
path. Proper CallKit audio-session coordination is deferred until the pod is
pinned/upgraded to LiveKit 2.15+.
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>
Plugin (rides next build):
- Audio: disable LiveKit auto audio-session config + keep the engine OFF, then
configure the session and start the engine ONLY in CXProvider didActivate
(stop in didDeactivate). Fixes intermittent dead mic / no audio and the "speaker
turns on late" routing. Request mic permission on connect so enabling the engine
in didActivate can't block on undetermined permission (SDK #815).
- Re-ring blip: a cancel push for a call we already ended/known no longer reports
a NEW incoming call (that was the phantom "rings back for a second"); it ends
the known call cleanly, and only reports+ends for a truly unknown (cold) call.
- Mute display: answer/outgoing reflect muted-by-default on the CallKit screen;
setMuted now drives mute THROUGH CallKit so the system screen and the in-app
meeting UI stay in sync.
- reportIncomingCall: new method to ring CallKit from a WebSocket call event — a
2nd path alongside the VoIP push for when the app is open (push can be delayed);
deduped by UUID.
Web (deploys now; the WS ring path activates once the build has the new method):
- onDmCall/onGroupCall call nativeReportIncoming for native incoming calls.
- audioActivated telemetry.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Plugin connects the LiveKit room without enabling the mic (nothing captured/
published until the user taps Mic, which is also when iOS asks permission).
Web: meetMic starts false so the mic button shows muted; on callConnected the
web pushes the muted state to the plugin so builds whose plugin still connects
the mic live are muted too.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Root cause of "answered but the call disconnects": native calls carry media
over LiveKit and bypass the mesh, so the server only learned "answered" from a
WebView POST. On a cold/locked answer the app is still launching and that event
was lost, so the 40s unanswered timer fired and cancelled the live call. Fix:
- Plugin: notifyListeners("answerCall", retainUntilConsumed:true) so a killed/
locked pickup isn't lost before the WebView JS attaches.
- Server markDmAnswered: emit call-taken to the callee's OTHER devices (stop the
ring; no teardown) and call-answered to the caller (flip UI to connected).
- Server declineDmCall: ignore a decline once the call is answered, so dismissing
a stale ring on a second device can't kill the live call.
Also adds a UI-only in-app call screen for native calls (caller + callee):
avatar, name, live timer, mute (-> plugin), end (-> CallKit). Native media has
no meeting window of its own; this covers "no meeting window / can't unmute".
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Caller saw the native callee TWICE — the outgoing 'Ringing…' placeholder (never cleared,
since a native callee doesn't send the mesh 'answered') plus a separate id-labelled tile.
Now when a native LiveKit participant joins: remove the __waiting placeholder, stop
ringback, and label the tile from CONTACTS (name + DP) instead of the raw user id.
Also (plugin, needs build): connectRoom disconnects any previous LiveKit connection
before joining, so repeated calls never leave duplicate/stale participants in the room.
WebView fix deploys now (no rebuild); the connectRoom fix rides the next build.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Build error 'unable to resolve module dependency: LiveKit' — the LiveKitClient pod
sets no module_name, so CocoaPods names the module after the pod (LiveKitClient). The
SDK compiled/linked fine; only the import statement was wrong. Types (Room, etc.)
unchanged.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
The plugin now carries the call media NATIVELY: on answer it connects the LiveKit Room
from the VoIP payload's url+token and publishes the mic; outgoing calls connect via
reportOutgoingCall(url,token). CallKit stays ACTIVE for the whole call (foregrounds the
app, keeps it alive) — the mic works because LiveKit's audio runs natively and
coordinates with CallKit (unlike WebKit's WebRTC). setMuted -> setMicrophone; end ->
disconnect. Removed the handoff hack.
NOT testable yet: still need (a) WebView to stop joining the room for native calls (one
connection per identity) and drive outgoing via reportOutgoingCall, and (b) server
lifecycle endpoints for native calls (answered/ended), since native media bypasses our
mesh/WS signaling. LiveKit Swift API authored without a local compile — expect a build
iteration or two. Don't build/flip CALLKIT_ENABLED yet.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
CRASH: iOS terminates an app that receives a VoIP push without calling
reportNewIncomingCall. My re-ring 'fix' made the cancel path call completion() without
reporting -> crash. Always report then immediately end on cancel (a tiny ring blip is
unavoidable; a crash is worse).
MIC + earpiece: an ACTIVE CallKit call reserves the mic (WebView WebRTC gets a dead mic
+ earpiece routing). So on answer keep CallKit active only long enough to foreground the
app, then end it and fire 'callHandoff'; the WebView forces the loudspeaker and
re-acquires the mic (sfuSetMic off/on, retried while it finishes joining).
MISSED CALL: endDmCallByRoom now sends a plain missed-call banner to the callee when the
call ends unanswered (timeout / caller hung up before pickup); skipped on decline.
Server (missed banner) deploys now; plugin + web handoff need a Codemagic build.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Answering did not open the app: ending the CallKit call immediately (to free the mic)
made iOS cancel the app launch before it foregrounded. Keep the call ACTIVE on answer
— fulfilling an active-call answer is what foregrounds/unlocks the app — and fire the
answerCall event so the WebView joins. didActivate stays a no-op (don't fight WebKit's
mic). The CallKit call is ended later when the WebView call ends.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Pivot away from the native-LiveKit rewrite. Discovery: the 'voip' UIBackgroundMode
already keeps the WebView's call audio alive when backgrounded (confirmed on device),
so background audio is solved WITHOUT native media. The only issue was CallKit
reserving the mic. So use CallKit purely for the incoming RING:
- Plugin CXAnswerCallAction: fulfill, then immediately end the CallKit call
(reportCall endedAt) to RELEASE the mic, and fire answerCall to the WebView after a
~1s beat so iOS tears down the CallKit audio session first. didActivate no longer
reconfigures the session (was fighting WebKit).
- home.html: outgoing calls no longer register with CallKit (WebView-only → mic works);
incoming still rings via CallKit → hands off to the WebView on answer.
Net: native full-screen ring + working mic + background audio + all existing call
features. Needs a Codemagic build; then flip CALLKIT_ENABLED=1 to test.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
MIC BROKEN with CallKit: a CallKit call reserves the microphone, so the WebView's
WebRTC can't capture it — calls are unusable until native LiveKit media lands. Add a
server kill-switch (CALLKIT_ENABLED, default OFF) so CallKit can be flipped without an
app rebuild: /api/meetings/config now returns callkit; setupNativeCall bails when off
(-> WebView calls, mic works); push.js only sends VoIP/CallKit pushes when enabled.
Deploying with the flag unset immediately restores working WebView calls.
RE-RING: a late cancel push for an already-declined call hit the plugin's 'unknown
uuid' path and re-reported a fresh incoming call (~1s re-ring). Track endedCalls and
make a late cancel for an already-ended call a no-op.
Server part deploys now (no rebuild); plugin re-ring fix ships with the native build.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Bug: a killed/backgrounded callee is woken only for the CallKit ring and has no
WebSocket yet, so the existing dm-call active:false (WS-only) never reaches it and
it keeps ringing after the caller hangs up.
Fix: send a 'cancel' VoIP push on call teardown.
- push.js: sendCallCancel() sends a {type:'cancel',callUUID} VoIP push to the user's
ios-voip tokens; invites now carry type:'invite'.
- calls.js: endDmCallByRoom + endGroupCallByRoom fire sendCallCancel to the rung users.
- NativeCallPlugin: on a cancel push, end the reported call (reportCall endedAt); if the
invite was never seen, report-then-end to satisfy iOS's 'report a call per VoIP push'.
Server part deploys now; the plugin part needs the next Codemagic build.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- didReceiveIncomingPush: normalise payload.dictionaryPayload ([AnyHashable:Any])
to [String:Any] before storing/using (the reported build error at :121).
- CXProviderConfiguration(localizedName:) instead of the no-arg init (available on
all deployment targets, avoids an availability edge).
- Make two 'calls[uuid] ?? [:]' bindings explicitly [String:Any] to avoid empty-
literal inference ambiguity.
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>
Sravan