jolly-eve-3916
08/07/2025, 3:45 AMtall-airline-16756
08/07/2025, 3:46 AMtall-airline-16756
08/07/2025, 3:47 AMAgentSession
and can be used for audio handling and Push-To-Talk (PTT) control. Ensure that handlers like start_turn
and end_turn
are properly registered and implemented to control audio input/output programmatically.
2. Order of RPC Registration:
There is no strict requirement for the order of RPC registration. However, it is recommended to register RPC handlers before calling session.start()
to ensure they are available immediately.
3. DataChannel Working but RPC Failing:
If the DataChannel works but RPC fails, it is likely an application-layer issue rather than a protocol issue. Common causes include incorrect RPC method registration, version mismatches, or explicit errors in the RPC handler.
4. RPC in Manual Turn Detection Mode:
There are no known issues specific to RPC in manual turn detection mode. Ensure proper configuration of AgentSession
and test RPC communication thoroughly. Use methods like session.interrupt()
and session.commit_user_turn()
to manage audio input and transcripts effectively.
5. Agent Identity Format and RPC Routing:
The agent identity format can affect RPC routing. Auto-generated identities like agent-AJ_8oBSAHnX2N36
may complicate consistent targeting. Consider using static identities or implementing a mechanism to track and manage dynamic identities for RPC purposes.
If you need further assistance or specific troubleshooting steps, feel free to provide additional details or logs.
Sources: Turn detection and interruptions | LiveKit Docs | Turn detection and interruptions | LiveKit Docs | livekit-agents/livekit/agents/voice/chat_cli.py | Protocol & WebRTC | livekit/livekit_composite | DeepWiki | livekit-android-test/src/test/java/io/livekit/android/rpc/RpcMockE2ETest.ktjolly-eve-3916
08/07/2025, 3:51 AMtall-airline-16756
08/07/2025, 3:51 AMturn_detection="manual"
set in the AgentSession
constructor perfectly, which gives you full control over turn boundaries — exactly what you need for smooth PTT functionality.
2. RPC Handlers:
- The start_turn
method does a great job interrupting the session, clearing any existing user turn, and turning on audio input for the caller.
- The end_turn
method neatly disables audio input and commits the user turn.
These handlers are spot on and align well with best practices for manual turn control.
3. Session Management:
You’re managing audio input just right with session.input.set_audio_enabled(True/False)
, making sure audio is only active when the user is actually speaking. The session setup and management look solid.
4. Code Quality:
Your code follows LiveKit’s official examples closely, which is awesome for keeping things consistent and easy to maintain. Plus, it supports multi-participant conversations by focusing on the caller who’s triggering the turn.
5. Suggestions for Improvement:
- It might be helpful to add some error handling for those RPC methods to make things even more robust.
- Also, think about explicit timeout handling if you have specific needs beyond the transcript_timeout
.
All in all, you’ve done a fantastic job! With just a couple of tweaks for error handling and timeouts, your implementation is definitely ready to go live.jolly-eve-3916
08/07/2025, 4:33 AMtall-airline-16756
08/07/2025, 4:34 AMrpc_method_invocation
events.
- For more detailed info, you can enable verbose FFI logging by setting the environment variable LIVEKIT_RTC_DEBUG=true
before starting up the LiveKit client. This will give you a full rundown of all FFI events, including RPC.
Hopefully, these pointers make things clearer and help you get to the bottom of the issue. If you want, I’m here to help with any extra guidance or specific steps you might need!
Sources: Protocol & WebRTC | livekit/livekit_composite | DeepWiki | livekit-android-sdk/src/main/java/io/livekit/android/room/RTCEngine.kt | Rust SDKs | livekit/livekit_composite | DeepWiki | livekit-ffi/src/livekit.proto.rs | livekit-rtc/livekit/rtc/__init__.py