Backend APIs return arrays directly, not { data, total } wrappers.
Changed 21 interface declarations to type aliases matching actual
API response format.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
All pages expected API responses in { data: [], total } format but
backend APIs return plain arrays. Changed data?.data ?? [] to data ?? []
across 22 page components.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
When API returns 401, clear stored tokens and redirect to /login
instead of showing an error message.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
The web-admin frontend was calling incorrect API paths that didn't match
the actual backend service routes through Kong gateway, causing all
requests to fail with 404 or route-mismatch errors.
URL corrections:
- servers: /api/v1/servers → /api/v1/inventory/servers
- runbooks: /api/v1/runbooks → /api/v1/ops/runbooks
- risk-rules: /api/v1/security/risk-rules → /api/v1/agent/risk-rules
- credentials: /api/v1/security/credentials → /api/v1/inventory/credentials
- roles: /api/v1/security/roles → /api/v1/auth/roles
- permissions: /api/v1/security/permissions → /api/v1/auth/permissions
- tenants: /api/v1/tenants → /api/v1/admin/tenants
- communication: /api/v1/communication → /api/v1/comm
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Problem:
- Text input area caused BOTTOM OVERFLOWED BY 135 PIXELS when keyboard opened
- Input bar overlapped with call control buttons
- Sent messages were not displayed on screen (only SnackBar feedback)
Solution — split into two distinct layouts:
1. Call Mode (default):
- Full-screen call UI: avatar, waveform, duration, large control buttons
- Keyboard button in controls toggles to chat mode
- No text input elements — clean voice-only interface
2. Chat Mode (tap keyboard button):
- Compact call header: green status dot + "iAgent" + duration + inline
mute/end/speaker/collapse controls
- Scrollable message list (Expanded widget — properly handles keyboard)
- User messages: right-aligned blue bubbles with attachment thumbnails
- Agent responses: left-aligned gray bubbles with robot avatar
- Input bar at bottom: attachment picker + text field + send button
Message display:
- User-sent text/attachments tracked in _messages list, shown as bubbles
- Agent responses sent back via LiveKit data channel (topic='text_reply')
from voice-agent → Flutter, displayed as assistant bubbles
- Auto-scroll to latest message
Voice-agent change (agent.py):
- After session.say(response), publish response text back to Flutter via
ctx.room.local_participant.publish_data() with topic='text_reply'
- Flutter listens for DataReceivedEvent to display agent responses
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Enable users to send text messages, images, and files to the Agent
while an active voice call is in progress. This addresses the case
where spoken instructions are unclear or screenshots/documents need
to be shared for analysis.
## Architecture
Data flows through LiveKit data channel (not direct HTTP):
Flutter → publishData(topic='text_inject') → voice-agent
→ llm.inject_text_message() → POST /api/v1/agent/tasks (same session)
→ collect streamed response → session.say() → TTS playback
This preserves the constraint that voice-agent owns the agent-service
sessionId — Flutter never contacts agent-service directly.
## Flutter UI (agent_call_page.dart)
- Add keyboard toggle button to active call controls (4-button row)
- Collapsible text input area with attachment picker (+) and send button
- Attachment support: gallery multi-select, camera, file picker
(images max 1024x1024 quality 80%, PDF supported, max 5 attachments)
- Horizontal scrolling attachment preview with delete buttons
- 200KB payload size check before LiveKit data channel send
- Layout adapts: Spacer flex 1/3 toggle, reduced bottom padding
## voice-agent (agent.py)
- Register data_received event listener after session.start()
- Filter for topic='text_inject', parse JSON payload
- Call llm.inject_text_message(text, attachments) and TTS via session.say()
- Use asyncio.ensure_future() wrapper for async handler (matches
existing disconnect handler pattern for sync EventEmitter)
## AgentServiceLLM (agent_llm.py)
- New inject_text_message(text, attachments) method on AgentServiceLLM
- Reuses same _agent_session_id for conversation context continuity
- WS+HTTP streaming: connect, pre-subscribe, POST /tasks with
attachments field, collect full text response, return string
- _injecting flag prevents concurrent _do_stream from clearing
session ID on abort errors while inject is in progress
- Same systemPrompt/voiceMode/engineType as voice pipeline
No agent-service changes required — attachments already supported
end-to-end (JSONB storage → multimodal content blocks → Claude).
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Previously, voice mode wrapped every user message with 【语音对话模式】
instructions, polluting conversation_messages history with repeated
instructions on every turn. Now:
- systemPrompt carries voice-mode instructions (set once, not per-message)
- prompt contains only the clean user text (identical to text chat pattern)
- Conversation history stays clean for multi-turn context
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
1. Remove on_enter greeting entirely (no more race condition)
2. voice-agent sends voiceMode: true when engine_type is claude_agent_sdk
3. AgentController.runTaskStream() filters thinking, tool_use, tool_result
events in voice mode — only text, completed, error reach the client
4. Detailed logging: each event logged with [FILTERED-voice] tag when skipped
Claude API mode is completely unaffected (voiceMode defaults to false).
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
1. Change on_enter greeting from generate_reply() to session.say() with
a static message — avoids spawning an Agent SDK task just for a greeting,
which caused a race condition when the user speaks before it completes.
2. Clear agent session ID when receiving abort/exit errors so the next
task starts a fresh session instead of trying to resume a dead process.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Full-stack implementation allowing users to choose between Claude Agent SDK
(default, with tool approval, skill injection, session resume) and Claude API
(direct, lower latency) in Flutter settings. Agent SDK mode wraps prompts with
voice-conversation instructions for concise spoken Chinese output.
Data flow: Flutter Settings → SharedPreferences → POST /livekit/token →
RoomAgentDispatch metadata → voice-agent → AgentServiceLLM(engine_type)
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Flutter (agent_call_page.dart):
- Add ConnectOptions with 15s timeouts for connection/peerConnection/iceRestart
- Add RoomReconnectingEvent/RoomAttemptReconnectEvent/RoomReconnectedEvent
listeners with "网络重连中" UI indicator during reconnection
- Add TimeoutException detection in _friendlyError()
voice-agent (agent.py):
- Wrap entrypoint() in try-except with full traceback logging
- Register room disconnect listener to close httpx clients (instead of
finally block, since session.start() returns while session runs in bg)
- Add asyncio import for ensure_future cleanup
voice-agent LLM proxy (agent_llm.py):
- Add retry with exponential backoff (max 2 retries, 1s/3s delays) for
network errors (ConnectError/ConnectTimeout/OSError) and WS InvalidStatusCode
- Extract _do_stream() method for single-attempt logic
- Add WebSocket connection params: open_timeout=10, ping_interval=20,
ping_timeout=10 for keepalive and faster dead-connection detection
- Use granular httpx.Timeout(connect=10, read=30, write=10, pool=10)
- Increase WS recv timeout from 5s to 30s to reduce unnecessary loops
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
LiveKit's use_external_ip auto-detected 154.84.135.121 (overseas) via
STUN, causing WebRTC ICE candidates to use an unreachable IP for
domestic mobile clients. Explicitly set node_ip to 14.215.128.96.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Flutter app now uses https://it0api.szaiai.com (nginx reverse proxy)
instead of direct IP:port. LiveKit URL uses China IP 14.215.128.96
for lower latency from domestic mobile clients.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
iconsulting-llm-gateway already occupies port 3008 on the host.
voice-service only has a single TCP port (no docker-proxy overhead),
so bridge networking with 13008:3008 mapping is sufficient.
Only livekit-server and voice-agent need host mode (UDP port ranges).
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Bridge mode created 600+ docker-proxy processes for LiveKit's UDP port-range
mappings (30000-30100, 50000-50200). Switch livekit-server, voice-agent, and
voice-service to network_mode: host for zero-overhead networking.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Default silence_duration_ms=350 is too aggressive for Chinese speech,
causing sentences to be fragmented into 1-3 character chunks. Increase
to 800ms and raise VAD threshold to 0.6 so the STT waits longer before
finalizing a turn, producing complete sentences for LLM processing.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
The OpenAI Realtime STT uses aiohttp WebSocket connections (not httpx),
so the existing httpx verify=False fix does not apply. LiveKit's
http_context creates aiohttp.TCPConnector without ssl=False, causing
SSL certificate verification errors when OPENAI_BASE_URL points to a
proxy with a self-signed certificate.
Monkey-patch http_context._new_session_ctx to inject ssl=False into the
aiohttp connector, fixing the "CERTIFICATE_VERIFY_FAILED" error for
Realtime STT WebSocket connections.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Add user-configurable TTS voice and tone style settings that flow from
the Flutter app through the backend to the voice-agent at call time.
## Flutter App (it0_app)
### Domain Layer
- app_settings.dart: Add `ttsVoice` (default: 'coral') and `ttsStyle`
(default: '') fields to AppSettings entity with copyWith support
### Data Layer
- settings_datasource.dart: Add SharedPreferences keys
`settings_tts_voice` and `settings_tts_style` for local persistence
in loadSettings(), saveSettings(), and clearSettings()
### Presentation Layer
- settings_providers.dart: Add `setTtsVoice()` and `setTtsStyle()`
methods to SettingsNotifier for Riverpod state management
- settings_page.dart: Add "语音" settings group between Notifications
and Security groups with:
- Voice picker: 13 OpenAI voices with gender/style labels
(e.g. "女 · 温暖", "男 · 沉稳", "中性") in a BottomSheet
- Style picker: 5 presets (专业干练/温柔耐心/轻松活泼/严肃正式/科幻AI)
as ChoiceChips + custom text input field + reset button
### Call Flow
- agent_call_page.dart: Send `tts_voice` and `tts_style` in the POST
body when requesting a LiveKit token at call initiation
## Backend
### voice-service (Python/FastAPI)
- livekit_token.py: Accept optional `tts_voice` and `tts_style` via
Pydantic TokenRequest body model; embed them in RoomAgentDispatch
metadata JSON alongside auth_header (backward compatible)
### voice-agent (Python/LiveKit Agents)
- agent.py: Extract `tts_voice` and `tts_style` from ctx.job.metadata;
use them when creating openai_plugin.TTS() — user-selected voice
overrides config default, user-selected style overrides default
instructions. Falls back to config defaults when not provided.
## Data Flow
Flutter Settings → SharedPreferences → POST /livekit/token body →
voice-service embeds in RoomAgentDispatch metadata →
voice-agent reads from ctx.job.metadata → TTS creation
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Switch from tts-1 to gpt-4o-mini-tts for lower latency and better quality
- Change voice from alloy to coral
- Add Chinese speech instructions for natural tone control
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Switch from batch STT (gpt-4o-transcribe via /audio/transcriptions)
to streaming Realtime API (WebSocket). This eliminates the ~2s batch
upload+process latency per utterance.
Also updated nginx proxy on 67.223.119.33 to support WebSocket upgrade
for /v1/realtime endpoint.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Pass httpx.AsyncClient(verify=False) to OpenAI STT/TTS to support
self-signed certificate on OPENAI_BASE_URL proxy
- Handle generate_reply calls with no user message by falling back to
system/developer instructions
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
In livekit-agents v1.x @server.rtc_session() pattern, ctx.room is not
yet connected when entrypoint is called. session.start() handles room
connection internally.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add room_input_options/room_output_options to session.start() so agent
binds audio I/O and stays in the room
- Add wait_for_participant() before starting session
- Filter AgentConfigUpdate items in agent_llm.py (no 'role' attribute)
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Replace deprecated WorkerOptions(entrypoint_fnc=...) with AgentServer() +
@server.rtc_session() decorator. Use server.setup_fnc for prewarm. Remove
manual ctx.connect() and ctx.wait_for_participant() calls that prevented
the pipeline from properly wiring up VAD→STT→LLM→TTS.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Subscriber transport was timing out on DTLS handshake for clients
behind complex NAT (VPN/symmetric NAT). Enable LiveKit's built-in
TURN server on UDP port 3478.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
RoomInputOptions is deprecated in livekit-agents 1.4.x. Switch to
RoomOptions with explicit audio_input/audio_output enabled.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Use Hardware.instance.setSpeakerphoneOn() to switch between speaker
and earpiece modes. Default to speaker on.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
LiveKit passes RoomAgentDispatch metadata through as job.metadata
(protobuf field), not via a separate agent_dispatch object. Also
use room_io.RoomInputOptions for participant targeting (livekit-agents 1.x).
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
livekit-agents 1.x removed the 'participant' parameter from
AgentSession.start(). Use room_input_options with participant_identity
instead.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
livekit_client 2.6.4 no longer has audioBitrate parameter.
Default AudioPublishOptions auto-selects optimal speech bitrate.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
The livekit_ws_url returned in token response needs to be the external
server address, not the internal Docker network name, so Flutter clients
can connect directly.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
The livekit package is the client SDK and doesn't include the server-side
API module. Switch to livekit-api which provides AccessToken, VideoGrants,
RoomAgentDispatch, and RoomConfiguration needed for token generation.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Upgrade websockets from ==12.0 to >=13.0 (openai[realtime] requires >=13)
- Install torch CPU-only build separately in Dockerfile to avoid ~2GB CUDA download
- Remove torch from requirements.txt (installed via --index-url cpu wheel)
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
FileType.custom with allowedExtensions causes Android system picker
to hide subdirectories on some devices. Changed to FileType.any with
post-selection extension validation instead.
- Unsupported file types are skipped with a SnackBar hint
- Allowed: jpg, jpeg, png, gif, webp, pdf
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Claude API supports up to 32MB PDFs; base64 encoding adds ~33% overhead.
50mb body limit covers the maximum single-document upload case.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
hailin