Calls — Testing

Strategy

Tests are split across three layers:

Layer	Tooling	Scope
Unit	Jest + React Native Testing Library	CallService state machine, CallMediaService audio routing, WebrtcSessionManager adapter wiring
Integration	Jest with mocked adapters	SignalingService mode selection, SDP/ICE relay path, WatermelonDB persistence
E2E	Maestro (two simulators)	Full call flow: start → accept → active → end

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Coverage Targets

Area	Target
CallService state transitions	100%
SignalingService routing logic	100%
WebrtcSessionManager SDP wiring	90%+
CallMediaService audio routes	100%
WatermelonDB record lifecycle	90%+
Overall call feature coverage	≥ 80%

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Mocking Rules

• WebrtcAdapter — always mocked in unit and integration tests; never connect to a real ICE agent.
• react-native-incall-manager — mock setSpeakerphoneOn and chooseAudioRoute; assert call args.
• WsSignalingAdapter / TcpClientAdapter — mock send and onMessage; return controlled SDP/ICE payloads.
• WatermelonDB — use the in-memory adapter (@nozbe/watermelondb/adapters/memory) for integration tests.

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Test Cases

Signalling — Unit / Integration

Scenario	Expected result
Initiator calls startCall(peerId, 'voice')	WebrtcAdapter.createOffer() called; offer SDP forwarded to SignalingService.send()
SignalingService in auto mode, LAN adapter available	Sends via TcpClientAdapter; does not use WsSignalingAdapter
SignalingService in auto mode, LAN adapter unavailable	Falls back to WsSignalingAdapter
SignalingService in server mode	Always sends via WsSignalingAdapter regardless of LAN state
Peer receives offer via relay	WebrtcSessionManager calls WebrtcAdapter.setRemoteDescription(offer)
Peer sends answer	Answer SDP relayed back to initiator; WebrtcAdapter.setRemoteDescription(answer) called on initiator
ICE candidate received	WebrtcAdapter.addIceCandidate(candidate) called on the correct adapter
ICE candidate send fails (transport error)	Error logged; candidate queued for retry; call not dropped immediately
Both sides complete ICE	onConnectionStateChange fires connected; CallService transitions to active

Call Lifecycle — Unit

Scenario	Expected result
startCall while another call is active	Returns error CALL_IN_PROGRESS; no new session created
Peer declines incoming call	CallService transitions to ended with status declined
Peer does not answer within 30 s	CallService transitions to ended with status missed; timeout cleared
endCall() called by initiator	WebrtcAdapter.close() called; CallMediaService.stopMedia() called; InCallManager.stop() called
endCall() called by peer	Same teardown sequence on peer side; call record updated with ended_at

WatermelonDB Persistence — Integration

Scenario	Expected result
Call initiated	call row created with status: 'ringing'; callparticipant row created for initiator
Call accepted by peer	call row updated to status: 'active'; callparticipant row created for peer with joined_at
Call ended	call row updated to status: 'ended'; callparticipant rows updated with left_at
Declined call	call row has status: 'declined'; no active transition recorded
Missed call	call row has status: 'missed'; ended_at set to timeout timestamp

Audio Routing — Unit

Scenario	Expected result
setAudioRoute('earpiece')	InCallManager.setSpeakerphoneOn(false) called once
setAudioRoute('speaker')	InCallManager.setSpeakerphoneOn(true) called once
setAudioRoute('bluetooth')	InCallManager.chooseAudioRoute('bluetooth') called
Audio route changed mid-call	Previous route deactivated; new route activated; no media interruption
CallMediaService.stopMedia() called	InCallManager.stop() called; all tracks stopped; streams set to null

Call Record Sync — Integration

Scenario	Expected result
Call ends with network available	SyncService.schedulePush() called; call and callparticipant rows included in push payload
Call ends with network unavailable	Rows remain in WatermelonDB; push deferred to next reconnect
Push succeeds	Server returns 200; local updated_at stays in sync
Push returns 409 (conflict)	SyncService re-pulls affected rows; local display updated

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E2E Scenarios (Maestro)

# calls-voice-flow.yaml
appId: com.ylpsapot.mobile
---
- launchApp
- tapOn: "Conversations"
- tapOn: "Peer Alpha"
- tapOn: "Voice call button"
- assertVisible: "Calling…"
- # On second simulator: accept call
- assertVisible: "Call active"
- tapOn: "End call"
- assertVisible: "Call ended"
- assertNotVisible: "Call active"

Run against two simulators: one as initiator, one as peer.

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Test File Locations

mobile-app/sapot-mobile-app/
  src/
    features/calls/
      __tests__/
        CallService.test.ts
        WebrtcSessionManager.test.ts
        CallMediaService.test.ts
        SignalingService.test.ts
        callPersistence.integration.test.ts
  e2e/
    calls-voice-flow.yaml
    calls-video-flow.yaml