post webrtc intégration
This commit is contained in:
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---
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sessionId: session-260912-132834-fuse
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---
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# Requirements
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### Overview & Goals
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Replace the current placeholder raw-UDP voice relay (`src/udp/*`) with a real WebRTC media stack built on the `rustrtc` crate (already declared in `Cargo.toml`). The server becomes a centralized **SFU** (Selective Forwarding Unit): each connected client opens **one `PeerConnection`** with the server, the server decrypts/receives that client's audio (and later video) track and forwards it to every other member of the same voice `Channel`.
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### Scope
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**In Scope**
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- Remove `src/udp/server.rs`, `src/udp/router.rs` raw-socket logic (metrics module is kept/adapted).
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- Add a `voice` module hosting: `PeerConnection` lifecycle management via `rustrtc`, SDP offer/answer exchange, ICE candidate exchange, and SFU-style track forwarding scoped by `channel_id`.
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- Reuse the existing WebSocket gateway (`GatewayManager` / `RealtimeRouter` / `on_message` in `src/routes/gateway/mod.rs`) as the signaling transport — add a `Voice` namespace to the existing JSON event envelope (`GatewayEvent { namespace, action, content }`) instead of introducing a new endpoint.
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- Reuse `channel::ChannelType::Voice`, `channel_user`, and the `JOIN_VOICE` / `SPEAK` permission bits already defined in `src/permissions.rs` to authorize who may join/publish in a voice channel.
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- Update `config.toml` / `NetworkConfig` so the single `udp_port` continues to be the one UDP port used, bound through `rustrtc`'s `ice_udp_mux` (single-port ICE) instead of a raw `UdpSocket`.
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- Update `AppMetrics`/`core::mod::App::run` wiring so the new voice service starts/stops alongside the HTTP server, replacing `UdpServer`.
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**Out of Scope**
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- TURN/STUN server configuration (kept minimal for LAN/local network scenarios for now).
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- Video/screen-share specific handling (structure will allow it later, but only audio forwarding is implemented now).
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- Client-side (frontend) implementation details beyond the signaling contract needed by the server.
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### User Stories
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- As a user with `JOIN_VOICE` permission on a voice channel, I want to connect and have my microphone audio heard by other members of that same channel.
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- As a user already in a voice channel, I want to hear every other member's audio forwarded by the server without opening a direct connection to each peer.
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- As a server operator, I want the previous ad-hoc UDP relay removed and replaced by a maintained WebRTC stack (rustrtc) so voice traffic is encrypted (SRTP) and NAT-traversal (ICE) works out of the box.
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### Functional Requirements
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- A client sends an SDP offer through the gateway (`namespace: "Voice"`), scoped to a `channel_id`; the server validates channel membership/permissions, creates a `rustrtc::PeerConnection`, and returns an SDP answer through the same gateway connection.
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- ICE candidates generated by either side are exchanged as additional `Voice` gateway events.
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- Once connected, the server subscribes the client's inbound audio track to every other `PeerConnection` currently joined to the same voice channel (SFU fan-out), and adds new joiners' tracks to previously-connected peers as they arrive.
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- When a user disconnects (gateway socket closes) or leaves the voice channel, their `PeerConnection` is closed and their track is removed from all other peers' forwarding sets.
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- Existing `UdpMetrics`-style counters (packets/bytes in/out, errors) are preserved in spirit, adapted for voice metrics and reported by the periodic `reporter`.
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# Technical Design
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### Current Implementation
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- `src/udp/server.rs`: binds a raw `tokio::net::UdpSocket` on `NetworkConfig.udp_port`, loops on `recv_from`, and blindly forwards datagrams to registered peers under a hard-coded `"default"` channel without protocol isolation or encryption.
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- `src/udp/router.rs`: `RoutingTable` maps `ChannelId -> Vec<SocketAddr>`; dead placeholder API.
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- `src/udp/metrics.rs`: atomic counters (`packets_received`, `bytes_sent`, etc.) + `spawn_reporter` logging every interval, plugged into `AppMetrics.udp` (`src/metrics/mod.rs`).
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- `src/core/mod.rs`: `App::build`/`App::run` constructs `UdpServer::new(&config.network, udp_metrics)` and `tokio::spawn(udp_server.run())`, joined into shutdown-broadcast pattern.
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- `src/routes/gateway/mod.rs`: `GatewayManager` tracks client WebSocket connections; `GatewayClient::on_message` handles incoming messages and acts as the extension point for voice signaling.
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- `src/services/realtime_registry.rs`: maintains `channel_id -> HashSet<user_id>` membership computed from `computed_permission` (READ_CHANNEL).
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- `src/permissions.rs`: `JOIN_VOICE`, `SPEAK`, `STREAM`, `MOVE_OTHERS`, `DISCONNECT_OTHERS`, `MANAGE_VOICE_CHANNEL` bits already defined.
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- `src/models/channel.rs`: `ChannelType::Voice` variant already exists.
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### Key Decisions
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1. **Topology: centralized SFU** — one `rustrtc::PeerConnection` per connected client per voice channel; the server forwards each publisher's RTP track to every other subscriber in that channel. No direct peer-to-peer mesh.
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2. **Signaling transport: existing WebSocket gateway** — SDP offer/answer and ICE candidates are carried as `Voice` namespace events inside the current `GatewayEvent` envelope, handled in `GatewayClient::on_message` and dispatched to `VoiceService`.
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3. **Single UDP port reuse** — `rustrtc`'s `RtcConfiguration` is set up with `ice_udp_mux = true` and bound to `NetworkConfig.udp_port`, preserving existing firewall/network port configurations.
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4. **No TURN/STUN for now** — `RtcConfiguration.ice_servers` left default/empty for local and direct network setups.
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5. **Voice presence tracked separately from `RealtimeRegistry`** — an in-memory registry (`VoiceRoom` per `channel_id`) tracks active WebRTC connections independently from general WebSocket channel presence.
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### Proposed Changes
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- Replace `src/udp` module with `src/voice`:
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- `voice/mod.rs`: public API, exports `VoiceService`.
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- `voice/service.rs`: `VoiceService` owns `rustrtc::RtcConfiguration`, a map `channel_id -> VoiceRoom`, and handler methods for offer, ICE candidates, and channel leaves.
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- `voice/room.rs`: `VoiceRoom` holds `HashMap<user_id, Arc<PeerConnection>>` for a channel and orchestrates track fan-out across peers.
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- `voice/metrics.rs`: tracks voice counters (packets/bytes/errors), exposed through `AppMetrics.voice` and the existing reporter cadence.
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- Extend `GatewayEvent` handling: add `"Voice"` namespace with actions `offer`, `answer`, `ice-candidate`, `leave`.
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- Permission check on `offer`: verify `ChannelPermission::JOIN_VOICE` and `SPEAK` before initializing a `PeerConnection`.
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- Update `src/core/mod.rs`: replace `UdpServer` initialization with `VoiceService` held inside `AppState`.
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- Update `src/config.rs`: keep `NetworkConfig.udp_port` feeding `rustrtc`'s `ice_udp_mux_port`.
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### Data Models / Contracts
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```rust
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// Gateway JSON contract additions (namespace = "Voice")
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{ "namespace": "Voice", "action": "offer", "content": { "channel_id": Uuid, "sdp": String } }
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{ "namespace": "Voice", "action": "answer", "content": { "channel_id": Uuid, "sdp": String } }
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{ "namespace": "Voice", "action": "ice-candidate", "content": { "channel_id": Uuid, "candidate": String } }
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{ "namespace": "Voice", "action": "leave", "content": { "channel_id": Uuid } }
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```
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```rust
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pub struct VoiceService {
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config: rustrtc::RtcConfiguration,
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rooms: RwLock<HashMap<Uuid, VoiceRoom>>,
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}
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impl VoiceService {
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pub async fn handle_offer(&self, user_id: Uuid, channel_id: Uuid, sdp: String) -> anyhow::Result<String>;
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pub async fn handle_ice_candidate(&self, user_id: Uuid, channel_id: Uuid, candidate: String) -> anyhow::Result<()>;
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pub fn leave(&self, user_id: Uuid, channel_id: Uuid);
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}
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```
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### Architecture Diagram
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```mermaid
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graph TD
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ClientA[Client A] -- WebSocket: Voice offer/answer/ICE --> Gateway[GatewayManager / on_message]
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ClientB[Client B] -- WebSocket: Voice offer/answer/ICE --> Gateway
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Gateway -- dispatch --> VoiceService
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VoiceService -- creates/owns --> RoomA[VoiceRoom per channel_id]
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RoomA -- PeerConnection A --> PCA[rustrtc PeerConnection A]
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RoomA -- PeerConnection B --> PCB[rustrtc PeerConnection B]
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ClientA == RTP/SRTP media on UDP mux port ==> PCA
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ClientB == RTP/SRTP media on UDP mux port ==> PCB
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PCA -- forward audio track --> PCB
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PCB -- forward audio track --> PCA
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```
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### Risks
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- `rustrtc` API details: verify track forwarding primitives against `rustrtc 0.3.133` during implementation and encapsulate track re-subscription in `voice/room.rs`.
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- Permission enforcement: ensure permission checks take place before creating peer connections to prevent unauthorized audio relay.
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- Signaling message volume: ensure gateway handling remains non-blocking for high frequency ICE candidates.
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# Delivery Steps
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### Step 1: Scaffold the voice module and RtcConfiguration from NetworkConfig
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A new `src/voice` module exists with a `VoiceService` capable of building a `rustrtc::RtcConfiguration` from the app's network config, replacing the old `src/udp` module's role in `core/mod.rs`.
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- Create `src/voice/mod.rs`, `src/voice/service.rs`, `src/voice/room.rs`, and `src/voice/metrics.rs`.
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- Port `UdpMetrics`-style counters into `voice/metrics.rs` (packets/bytes in/out, errors), keeping the `Metrics`/`MetricsSnapshot` trait implementations used by `crate::metrics::reporter`.
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- Build `RtcConfiguration`/`RtcConfigurationBuilder` in `VoiceService::new(&NetworkConfig, metrics)` using `ice_udp_mux = true` and `ice_udp_mux_port = network.udp_port`, with no external ICE servers configured.
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- Remove `src/udp/server.rs` and `src/udp/router.rs`; delete unused `RoutingTable`.
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- Update `src/core/mod.rs` (`App::build`/`App::run`) to construct `VoiceService` instead of `UdpServer`, store it on `AppState`, and drop unneeded `udp_shutdown_tx`/`udp_handle` tasks.
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- Update `src/metrics/mod.rs` (`AppMetrics`) to reference the new voice metrics type.
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### Step 2: Implement per-channel PeerConnection lifecycle in VoiceRoom
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`VoiceService`/`VoiceRoom` can accept an SDP offer for a given channel, create a `rustrtc::PeerConnection`, and return an SDP answer, tracking connections per `channel_id`.
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- Implement `VoiceRoom` (`channel_id -> HashMap<user_id, Arc<PeerConnection>>`).
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- Implement `VoiceService::handle_offer(user_id, channel_id, sdp)`: looks up or creates the room, creates a `PeerConnection` via `rustrtc`, invokes `set_remote_description`/`create_answer`/`set_local_description`, and returns the answer SDP.
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- Implement `VoiceService::handle_ice_candidate` and `VoiceService::leave`, closing and removing the `PeerConnection` from its room.
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- Enforce `ChannelPermission::JOIN_VOICE` (and `SPEAK` for publishing) before creating a `PeerConnection`, reusing the `computed_permission` lookup pattern from `src/services/realtime_registry.rs`.
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### Step 3: Wire SFU track forwarding between peers in the same channel
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Audio published by one connected client in a voice channel is forwarded by the server to every other client connected to the same channel.
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- On `PeerConnection::on_track` for a given user's connection, register the inbound track on the owning `VoiceRoom`.
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- For every other `PeerConnection` already in that `VoiceRoom`, add and forward the new track (SFU fan-out) using `rustrtc`'s track/`MediaCapabilities` APIs.
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- When a new peer joins an existing room, subscribe it to all tracks already being forwarded by other room members.
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- On `leave` or disconnect, remove the peer's published track from all other peers' forwarding sets and close its `PeerConnection`.
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### Step 4: Expose Voice signaling through the existing WebSocket gateway
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Clients can perform the full offer/answer/ICE handshake over the existing gateway WebSocket connection using a new `Voice` namespace, with no new HTTP/WS endpoint introduced.
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- Extend `GatewayClient::on_message` in `src/routes/gateway/mod.rs` to parse `GatewayEvent{namespace: "Voice", action, content}` messages (`offer`, `answer`, `ice-candidate`, `leave`).
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- Dispatch parsed messages to `VoiceService` (accessed via `AppState`) and send the resulting answer/ICE-candidate events back through the client's existing `mpsc::UnboundedSender<Message>`.
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- On gateway disconnect (`GatewayClient::on_disconnect`), call `VoiceService::leave` for any channel the user was actively connected to in voice.
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- Update `config.toml`'s `DEFAULT_CONFIG_TOML` comment for `udp_port` to reflect its new role as the rustrtc ICE/media mux port.
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@@ -0,0 +1,134 @@
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---
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sessionId: session-260912-132945-1get
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---
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# Requirements
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### Overview & Goals
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Separate WebRTC voice signaling from the main real-time chat gateway by introducing a **dedicated Voice WebSocket endpoint** (`/ws/voice`). The voice signaling protocol (SDP offer/answer exchange, ICE candidates, room joins/leaves) is isolated from chat events, mirroring the architecture of production communication platforms (e.g. Discord's separate Gateway and Voice Gateway).
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### Scope
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**In Scope**
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- Implement a dedicated WebSocket route `GET /ws/voice` (nested alongside `/ws/gateway` under `ws_routes`).
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- Authenticate incoming voice WebSocket connections using the existing `CurrentUser` extractor (supporting Authorization headers, cookies, and `?token=` query parameters).
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- Define and implement a lightweight, dedicated JSON signaling protocol for voice sessions (`offer`, `answer`, `ice-candidate`, `leave`, `error`).
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- Bind the voice connection lifecycle directly to the WebSocket lifecycle: connecting/joining initiates the session; closing the socket cleanly tears down the `PeerConnection` and removes the participant from the `VoiceRoom`.
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- Remove voice signaling handlers (`namespace: "Voice"`) from the general chat gateway (`src/routes/gateway/mod.rs`), restoring clean separation of concerns.
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- Maintain full compatibility with `VoiceService` and `rustrtc` SFU track fan-out.
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**Out of Scope**
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- Changing the underlying `rustrtc` SFU forwarding engine or UDP multiplexing port (`udp_port`).
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- Implementing TURN/STUN relay configurations.
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- Video/screen-sharing UI controls (handled in future iterations).
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### User Stories
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- As a client joining a voice channel, I want to open a dedicated WebSocket connection for voice so that media signaling traffic (large SDP payloads, ICE candidates) does not delay or congest my chat message stream.
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- As a client leaving a voice channel, I want closing my voice WebSocket connection to immediately and cleanly release all server WebRTC resources without affecting my general gateway connection.
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- As a backend developer, I want voice signaling isolated in its own module/handler so the main gateway router remains focused strictly on application domain events.
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### Functional Requirements
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- **Dedicated Route**: `GET /ws/voice` upgrades to a WebSocket connection authenticated via `CurrentUser`.
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- **Signaling Exchange**:
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- Client sends `offer` with `{ "channel_id": Uuid, "sdp": String }`. The server validates permissions (`JOIN_VOICE`), creates/updates the `rustrtc::PeerConnection`, and returns an `answer` event with the server SDP.
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- Client and server exchange `ice-candidate` events with `{ "channel_id": Uuid, "candidate": String }`.
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- Client sends `leave` or closes the socket; server removes peer from `VoiceRoom` and unsubscribes tracks.
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- **Error Handling**: Server returns `{ "action": "error", "message": String }` on permission denial, invalid payload, or WebRTC negotiation failure.
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- **Gateway Decoupling**: `/ws/gateway` no longer processes `"Voice"` namespace events.
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### Non-Functional Requirements
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- **Performance**: High throughput and low latency signaling with asynchronous message handling via Tokio tasks.
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- **Reliability**: Deterministic cleanup on socket disconnection preventing orphaned WebRTC sessions or memory leaks.
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# Technical Design
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### Current Implementation
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- `src/routes/mod.rs`: Mounts `ws_routes` containing `/ws/gateway`.
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- `src/routes/gateway/mod.rs`: `GatewayClient::on_message` inspects JSON events; if `namespace == "Voice"`, it dispatches `offer`, `ice-candidate`, and `leave` to `AppState.voice`.
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- `src/voice/service.rs`: Manages `RtcConfiguration`, `VoiceRoom` instances, SDP answers, ICE candidates, and permissions.
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- `src/http/middleware.rs` & `src/http/context.rs`: `auth_middleware` extracts JWTs from Bearer headers, cookies, or `?token=` query parameters and populates `CurrentUser`.
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### Key Decisions
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1. **Dedicated WebSocket Route (`/ws/voice`)**: Voice signaling is completely extracted into its own endpoint and handler module (`src/routes/voice/` or `src/voice/ws.rs`), eliminating protocol coupling in `src/routes/gateway/`.
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2. **Connection-bound Voice Lifecycle**: The dedicated WebSocket lifetime directly reflects voice room occupancy. When the WebSocket connection drops, `VoiceService::leave` is triggered automatically.
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3. **Simplified Signaling Protocol**: Elimination of the generic `GatewayEvent` envelope wrapper (`{ namespace, action, content }`) in favor of direct, typed voice signaling messages (`{ action, ... }`).
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4. **Standardized Authentication**: Reuse of `CurrentUser` extractor ensures uniform JWT validation across REST, Gateway, and Voice WebSocket endpoints.
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### Proposed Changes
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- **New Route & Handler**:
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- Create `src/routes/voice/` with `mod.rs`, `handlers.rs`, and `routes.rs` (or integrate under `src/voice/` and expose via `src/routes/mod.rs`).
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- Add `GET /voice` route to `ws_routes` in `src/routes/mod.rs` (resulting in `/ws/voice`).
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- Implement `voice_ws_handler` accepting `WebSocketUpgrade`, `State(AppState)`, and `CurrentUser(user)`.
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- **Voice WebSocket Protocol**:
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- Define incoming message types: `Offer { channel_id, sdp }`, `IceCandidate { channel_id, candidate }`, `Leave { channel_id }`.
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- Define outgoing message types: `Answer { channel_id, sdp }`, `IceCandidate { channel_id, candidate }`, `Error { message }`.
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- **Gateway Cleanup**:
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- Remove `namespace == "Voice"` handling from `GatewayClient::on_message`.
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- Remove voice-specific imports from `src/routes/gateway/mod.rs`.
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- **VoiceService Integration**:
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- Update `VoiceService::handle_offer` to take the dedicated voice WebSocket sender channel for emitting server ICE candidates.
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### Data Models / Contracts
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```rust
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// Voice WebSocket Client -> Server messages
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#[derive(Debug, Deserialize)]
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#[serde(tag = "action", rename_all = "kebab-case")]
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pub enum VoiceClientMessage {
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Offer {
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channel_id: Uuid,
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sdp: String,
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},
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IceCandidate {
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channel_id: Uuid,
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candidate: String,
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},
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Leave {
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channel_id: Uuid,
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},
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}
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// Voice WebSocket Server -> Client messages
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#[derive(Debug, Serialize)]
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#[serde(tag = "action", rename_all = "kebab-case")]
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pub enum VoiceServerMessage {
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Answer {
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channel_id: Uuid,
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sdp: String,
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},
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IceCandidate {
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channel_id: Uuid,
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candidate: String,
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},
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Error {
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message: String,
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},
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}
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```
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### Components
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- `VoiceWsHandler` (`src/routes/voice/handlers.rs`): Handles WebSocket upgrades, authentication, event loop, message serialization/deserialization, and lifecycle teardown.
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- `VoiceRoutes` (`src/routes/voice/routes.rs`): Defines Axum router for `/voice`.
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- `VoiceService` (`src/voice/service.rs`): Coordinates `rustrtc::PeerConnection` instances, SFU track forwarding, and room management.
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- `GatewayManager` (`src/routes/gateway/mod.rs`): Retains responsibility solely for domain event broadcasting (chat, server, channel updates).
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### File Structure
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- `src/routes/mod.rs`: Register `voice::routes::router()` under `/ws`.
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- `src/routes/voice/mod.rs`: Public module definition.
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- `src/routes/voice/routes.rs`: Voice WebSocket route definition.
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- `src/routes/voice/handlers.rs`: Voice WebSocket connection handler and message loop.
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- `src/routes/voice/messages.rs`: Strongly typed signaling message definitions.
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- `src/routes/gateway/mod.rs`: Cleaned up to remove voice signaling logic.
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### Architecture Diagram
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```mermaid
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graph TD
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Client[Client App]
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Client -- Chat events / presence --> GW["/ws/gateway (GatewayManager)"]
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Client -- Voice signaling (SDP/ICE) --> VWS["/ws/voice (VoiceWsHandler)"]
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VWS -- Manage connections & rooms --> VS[VoiceService]
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VS -- SFU media fan-out --> RR[rustrtc / VoiceRoom]
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Client == SRTP media over single UDP port ==> RR
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```
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### Risks
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- **Socket Disconnect Race Conditions**: Ensure that client disconnection promptly removes the user from `VoiceRoom` without deadlocking `RwLock` guards.
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- **Message Deserialization Errors**: Malformed payloads must result in an `Error` response over the WebSocket rather than terminating the connection loop prematurely.
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@@ -0,0 +1,138 @@
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---
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sessionId: session-260912-101455-1pxv
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---
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# Requirements
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### Overview & Goals
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||||
Replace the current placeholder raw-UDP voice relay (`src/udp/*`) with a real WebRTC media stack built on the `rustrtc` crate (already declared in `Cargo.toml`). The server becomes a centralized **SFU** (Selective Forwarding Unit): each connected client opens **one `PeerConnection`** with the server, the server decrypts/receives that client's audio (and later video) track and re-encodes/forwards it to every other member of the same voice `Channel`.
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### Scope
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**In Scope**
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- Remove `src/udp/server.rs`, `src/udp/router.rs` raw-socket logic (metrics module is kept/adapted).
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- Add a `voice` module hosting: `PeerConnection` lifecycle management via `rustrtc`, SDP offer/answer exchange, ICE candidate exchange, and SFU-style track forwarding scoped by `channel_id`.
|
||||
- Reuse the existing WebSocket gateway (`GatewayManager` / `RealtimeRouter` / `on_message` in `src/routes/gateway/mod.rs`) as the signaling transport — add a `Voice` namespace to the existing JSON event envelope (`GatewayEvent { namespace, action, content }`) instead of introducing a new endpoint, per explicit request to avoid breaking the current structure.
|
||||
- Reuse `channel::ChannelType::Voice`, `channel_user`, and the `JOIN_VOICE` / `SPEAK` permission bits already defined in `src/permissions.rs` to authorize who may join/publish in a voice channel.
|
||||
- Update `config.toml` / `NetworkConfig` so the single `udp_port` continues to be the one UDP port used, now bound through `rustrtc`'s `ice_udp_mux` (single-port ICE) instead of a raw `UdpSocket`.
|
||||
- Update `AppMetrics`/`core::mod::App::run` wiring so the new voice service starts/stops alongside the HTTP server, replacing `UdpServer`.
|
||||
|
||||
**Out of Scope**
|
||||
- TURN/STUN server configuration (kept minimal per user's answer — LAN/local network scenario for now).
|
||||
- Video/screen-share specific handling (structure will allow it later, but only audio forwarding is implemented now).
|
||||
- Client-side (frontend) implementation details beyond the signaling contract needed by the server.
|
||||
|
||||
### User Stories
|
||||
- As a user with `JOIN_VOICE` permission on a voice channel, I want to connect and have my microphone audio heard by other members of that same channel.
|
||||
- As a user already in a voice channel, I want to hear every other member's audio mixed/forwarded by the server without opening a direct connection to each peer.
|
||||
- As a server operator, I want the previous ad-hoc UDP relay removed and replaced by a maintained WebRTC stack (rustrtc) so voice traffic is encrypted (SRTP) and NAT-traversal (ICE) works out of the box.
|
||||
|
||||
### Functional Requirements
|
||||
- A client sends an SDP offer through the gateway (`namespace: "Voice"`), scoped to a `channel_id`; the server validates channel membership/permissions, creates a `rustrtc::PeerConnection`, and returns an SDP answer through the same gateway connection.
|
||||
- ICE candidates generated by either side are exchanged as additional `Voice` gateway events.
|
||||
- Once connected, the server subscribes the client's inbound audio track to every other `PeerConnection` currently joined to the same voice channel (SFU fan-out), and adds new joiners' tracks to previously-connected peers as they arrive.
|
||||
- When a user disconnects (gateway socket closes) or leaves the voice channel, their `PeerConnection` is closed and their track is removed from all other peers' forwarding sets.
|
||||
- Existing `UdpMetrics`-style counters (packets/bytes in/out, errors) are preserved in spirit, adapted to whatever counters `rustrtc` exposes (or wrapped manually) for the periodic `reporter`.
|
||||
|
||||
# Technical Design
|
||||
|
||||
### Current Implementation
|
||||
- `src/udp/server.rs`: binds a raw `tokio::net::UdpSocket` on `NetworkConfig.udp_port`, loops on `recv_from`, and blindly forwards every datagram to all peers registered in a `RoutingTable` under a **hard-coded** `"default"` channel — there is no real protocol, no per-voice-channel isolation, no encryption.
|
||||
- `src/udp/router.rs`: `RoutingTable` maps `ChannelId -> Vec<SocketAddr>`; `join`/`leave`/`routing_table_mut()` exist but are **never called** anywhere in the codebase — dead placeholder API.
|
||||
- `src/udp/metrics.rs`: atomic counters (`packets_received`, `bytes_sent`, …) + `spawn_reporter` logging every interval. Plugged into `AppMetrics.udp` (`src/metrics/mod.rs`).
|
||||
- `src/core/mod.rs`: `App::build`/`App::run` construct `UdpServer::new(&config.network, udp_metrics)` and `tokio::spawn(udp_server.run())`, joined into the same `tokio::select!`/shutdown-broadcast pattern as `HttpServer`.
|
||||
- `src/routes/gateway/mod.rs`: `GatewayManager` keeps `ConnectionKey{user_id, connection_id} -> GatewayClient{sender: mpsc::UnboundedSender<Message>}`; events are pushed as JSON `GatewayEvent{namespace, action, content}`. `GatewayClient::on_message` currently just logs incoming text — this is the extension point for voice signaling.
|
||||
- `src/services/realtime_registry.rs`: already maintains `channel_id -> HashSet<user_id>` membership computed from `computed_permission` (READ_CHANNEL) — reusable to know who is allowed in a channel, but voice-specific "currently connected to voice" state does not exist yet and must be tracked separately (WebSocket presence in a channel != actively broadcasting audio).
|
||||
- `src/permissions.rs`: `JOIN_VOICE`, `SPEAK`, `STREAM`, `MOVE_OTHERS`, `DISCONNECT_OTHERS`, `MANAGE_VOICE_CHANNEL` bits already defined but unused by any voice logic today.
|
||||
- `src/models/channel.rs`: `ChannelType::Voice` variant already exists.
|
||||
|
||||
### Key Decisions
|
||||
1. **Topology: centralized SFU** (confirmed by user) — one `rustrtc::PeerConnection` per connected client per voice channel; the server forwards each publisher's decoded RTP track to every other subscriber in that channel. No client-to-client PeerConnections.
|
||||
2. **Signaling transport: existing WebSocket gateway** (confirmed by user, to avoid restructuring) — SDP offer/answer and ICE candidates are carried as new `Voice` namespace events inside the current `GatewayEvent` envelope, handled in `GatewayClient::on_message` (currently a stub) and dispatched to a new `VoiceService`.
|
||||
3. **Single UDP port reuse** — `rustrtc`'s `RtcConfiguration` will be set up with `ice_udp_mux = true` and bound to the existing `NetworkConfig.udp_port`, so the media/ICE traffic keeps using the same single port previously owned by the raw `UdpServer`, minimizing config/infra changes (firewall rules, `config.toml` stay compatible).
|
||||
4. **No TURN/STUN for now** (confirmed by user) — `RtcConfiguration.ice_servers` left empty/default; can be added later via `config.toml` without further architecture changes.
|
||||
5. **Voice presence tracked separately from `RealtimeRegistry`** — a new lightweight in-memory registry (`VoiceRoom` per `channel_id`) tracks which `PeerConnection`s are actively publishing/subscribing in a voice channel, since being subscribed to gateway events (`RealtimeRegistry`) is not the same as being connected to voice media.
|
||||
|
||||
### Proposed Changes
|
||||
- Replace the `src/udp` module with a new `src/voice` module:
|
||||
- `voice/mod.rs`: public API, exports `VoiceService`.
|
||||
- `voice/service.rs`: `VoiceService` owns a `rustrtc::RtcConfiguration` (built from `NetworkConfig`), a map `channel_id -> VoiceRoom`, and methods `handle_offer(user_id, channel_id, sdp) -> answer_sdp`, `handle_ice_candidate(...)`, `leave(user_id, channel_id)`.
|
||||
- `voice/room.rs`: `VoiceRoom` holds `HashMap<user_id, Arc<PeerConnection>>` for one channel; implements track fan-out — on receiving a remote track from peer A, it calls into every other peer's `PeerConnection` to add/forward that track (rustrtc `PeerConnection` API, per its `Usage` example: create connection, `set_remote_description`, `create_answer`, `set_local_description`, subscribe to `on_track`).
|
||||
- `voice/metrics.rs`: keep counters analogous to today's `UdpMetrics` (packets/bytes/errors), fed by hooks around track forwarding, still exposed through `AppMetrics.udp` (renamed `AppMetrics.voice` where feasible) and the existing `reporter::spawn_reporter` cadence.
|
||||
- Extend `GatewayEvent` handling: add `"Voice"` namespace with actions like `offer`, `answer`, `ice-candidate`, `leave`; `GatewayClient::on_message` parses these and calls `VoiceService` through `AppState`, then pushes the answer/ICE-candidate response back on the same `mpsc::UnboundedSender<Message>` used for all other gateway events (no protocol reinvention).
|
||||
- Permission check on `offer`: verify the user has `ChannelPermission::JOIN_VOICE` (and `SPEAK` to publish) on the target channel, reusing the same `computed_permission` lookups already used by `RealtimeRegistry`/`http/permissions.rs`.
|
||||
- Update `src/core/mod.rs`: drop `UdpServer::new` / `udp_server.run()` / `udp_shutdown_tx`; construct `VoiceService` instead and store it in `AppState` (alongside `gateway`, `services`) so gateway handlers can call it; no UDP socket bind/run task is spawned by `App::run` anymore — `rustrtc` manages its own I/O internally once configured.
|
||||
- Update `src/config.rs`: keep `NetworkConfig.udp_port` (renamed conceptually to "voice/media port" in comments) feeding `rustrtc`'s `ice_udp_mux_port`; update `DEFAULT_CONFIG_TOML` comment accordingly.
|
||||
- Remove `src/udp/` directory entirely once `voice/` fully replaces its responsibilities; update `Cargo.toml`/module declarations (`lib.rs`) accordingly.
|
||||
|
||||
### Data Models / Contracts
|
||||
```rust
|
||||
// Gateway JSON contract additions (namespace = "Voice")
|
||||
{ "namespace": "Voice", "action": "offer", "content": { "channel_id": Uuid, "sdp": String } }
|
||||
{ "namespace": "Voice", "action": "answer", "content": { "channel_id": Uuid, "sdp": String } }
|
||||
{ "namespace": "Voice", "action": "ice-candidate", "content": { "channel_id": Uuid, "candidate": String } }
|
||||
{ "namespace": "Voice", "action": "leave", "content": { "channel_id": Uuid } }
|
||||
```
|
||||
```rust
|
||||
pub struct VoiceService {
|
||||
config: rustrtc::RtcConfiguration,
|
||||
rooms: RwLock<HashMap<Uuid, VoiceRoom>>, // channel_id -> room
|
||||
}
|
||||
|
||||
impl VoiceService {
|
||||
pub async fn handle_offer(&self, user_id: Uuid, channel_id: Uuid, sdp: String) -> anyhow::Result<String>;
|
||||
pub async fn handle_ice_candidate(&self, user_id: Uuid, channel_id: Uuid, candidate: String) -> anyhow::Result<()>;
|
||||
pub fn leave(&self, user_id: Uuid, channel_id: Uuid);
|
||||
}
|
||||
```
|
||||
|
||||
### Architecture Diagram
|
||||
```mermaid
|
||||
graph TD
|
||||
ClientA[Client A - browser] -- WebSocket gateway: Voice offer/answer/ICE --> Gateway[GatewayManager / on_message]
|
||||
ClientB[Client B - browser] -- WebSocket gateway: Voice offer/answer/ICE --> Gateway
|
||||
Gateway -- dispatch --> VoiceService
|
||||
VoiceService -- creates/owns --> RoomA[VoiceRoom per channel_id]
|
||||
RoomA -- PeerConnection A --> PCA[rustrtc PeerConnection A]
|
||||
RoomA -- PeerConnection B --> PCB[rustrtc PeerConnection B]
|
||||
ClientA == RTP/SRTP media, single UDP port ==> PCA
|
||||
ClientB == RTP/SRTP media, single UDP port ==> PCB
|
||||
PCA -- forward audio track --> PCB
|
||||
PCB -- forward audio track --> PCA
|
||||
```
|
||||
|
||||
### Risks
|
||||
- `rustrtc` is a young/fast-moving crate (frequent point releases per the benchmark notes found) — API surface for track-forwarding/SFU usage should be validated against the pinned `0.3.133` docs during implementation; if a needed primitive (e.g., explicit track re-publishing helper) is missing, a thin adapter layer will be needed inside `voice/room.rs`.
|
||||
- Moving from "no real protocol" to full SDP/ICE negotiation is a larger surface than the previous placeholder; permission checks (`JOIN_VOICE`/`SPEAK`) must be enforced before any `PeerConnection` is created to avoid unauthorized channel joins.
|
||||
- Since signaling now flows through the same WebSocket used for all other realtime events, malformed/large SDP payloads must not block the `GatewayClient` message loop — the voice service calls will be dispatched without blocking other event types.
|
||||
|
||||
# Delivery Steps
|
||||
|
||||
### * Step 1: Scaffold the voice module and RtcConfiguration from NetworkConfig
|
||||
A new `src/voice` module exists with a `VoiceService` capable of building a `rustrtc::RtcConfiguration` from the app's network config, replacing the old `src/udp` module's role in `core/mod.rs`.
|
||||
- Create `src/voice/mod.rs`, `src/voice/service.rs`, `src/voice/room.rs`, `src/voice/metrics.rs`.
|
||||
- Port `UdpMetrics`-style counters into `voice/metrics.rs` (packets/bytes in/out, errors), keeping the `Metrics`/`MetricsSnapshot` trait implementations used by `crate::metrics::reporter`.
|
||||
- Build `RtcConfiguration`/`RtcConfigurationBuilder` in `VoiceService::new(&NetworkConfig, metrics)` using `ice_udp_mux = true` and `ice_udp_mux_port = network.udp_port`, no ICE servers configured.
|
||||
- Remove `src/udp/server.rs` and `src/udp/router.rs`; delete unused `RoutingTable` (dead code confirmed unused elsewhere).
|
||||
- Update `src/core/mod.rs` (`App::build`/`App::run`) to construct `VoiceService` instead of `UdpServer`, store it on `AppState`, and drop the now-unneeded `udp_shutdown_tx`/`udp_handle` spawn/join wiring.
|
||||
- Update `src/metrics/mod.rs` (`AppMetrics`) to reference the new voice metrics type.
|
||||
|
||||
### Step 2: Implement per-channel PeerConnection lifecycle in VoiceRoom
|
||||
`VoiceService`/`VoiceRoom` can accept an SDP offer for a given channel, create a `rustrtc::PeerConnection`, and return an SDP answer, tracking connections per `channel_id`.
|
||||
- Implement `VoiceRoom` (`channel_id -> HashMap<user_id, Arc<PeerConnection>>`).
|
||||
- Implement `VoiceService::handle_offer(user_id, channel_id, sdp)`: looks up/creates the room, creates a `PeerConnection` via `rustrtc`, calls `set_remote_description`/`create_answer`/`set_local_description`, returns the answer SDP.
|
||||
- Implement `VoiceService::handle_ice_candidate` and `VoiceService::leave`, closing and removing the `PeerConnection` from its room.
|
||||
- Enforce `ChannelPermission::JOIN_VOICE` (and `SPEAK` for publishing) before creating a `PeerConnection`, reusing the `computed_permission` lookup pattern from `src/services/realtime_registry.rs`.
|
||||
|
||||
### Step 3: Wire SFU track forwarding between peers in the same channel
|
||||
Audio published by one connected client in a voice channel is forwarded by the server to every other client connected to the same channel.
|
||||
- On `PeerConnection::on_track` for a given user's connection, register the inbound track on the owning `VoiceRoom`.
|
||||
- For every other `PeerConnection` already in that `VoiceRoom`, add/forward the new track (SFU fan-out) using `rustrtc`'s track/`MediaCapabilities` APIs.
|
||||
- When a new peer joins an existing room, subscribe it to all tracks already being forwarded by other room members.
|
||||
- On `leave`/disconnect, remove the peer's published track from all other peers' forwarding sets and close its `PeerConnection`.
|
||||
|
||||
### Step 4: Expose Voice signaling through the existing WebSocket gateway
|
||||
Clients can perform the full offer/answer/ICE handshake over the existing gateway WebSocket connection using a new `Voice` namespace, with no new HTTP/WS endpoint introduced.
|
||||
- Extend `GatewayClient::on_message` in `src/routes/gateway/mod.rs` to parse `GatewayEvent{namespace: "Voice", action, content}` messages (`offer`, `answer`, `ice-candidate`, `leave`).
|
||||
- Dispatch parsed messages to `VoiceService` (accessed via `AppState`) and send the resulting answer/ICE-candidate events back through the client's existing `mpsc::UnboundedSender<Message>`.
|
||||
- On gateway disconnect (`GatewayClient::on_disconnect`), call `VoiceService::leave` for any channel the user was actively connected to in voice.
|
||||
- Update `config.toml`'s `DEFAULT_CONFIG_TOML` comment for `udp_port` to reflect its new role as the rustrtc ICE/media mux port.
|
||||
Reference in New Issue
Block a user