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feat: fix voice join/leave, channel edit defaults, channel-update stream restart
Three bugs fixed across the full stack (proto/server/core/ABI/Win/macOS/iOS):

1. Join/Leave Voice now truly subscribes/unsubscribes from the voice plane.
   Previously the button only toggled the local mic — receiving was always on
   (gated by channel membership alone). Added a protocol-level voice subscription
   concept: new SubscribeVoiceRequest/UnsubscribeVoiceRequest/VoiceSubscriptionResult
   proto messages, User.voice_subscribed field, vc_join_voice/vc_leave_voice C ABI
   functions, VC_EVENT_VOICE_STATE event, server-side voice_subscribed flag checked
   by the SFU relay recipient filter, and core-client gating of remote-stream
   decoder setup. All three clients rewired to subscribe+mic on Join / unsubscribe
   on Leave. Text chat works regardless of voice subscription.

2. Channel edit dialog now shows the channel's actual current settings. The read
   struct vc_channel was missing sort_order and audio fields — only the write
   struct vc_channel_info had them. Extended vc_channel with both (additive, no
   ABI break), updated the session model and list_channels marshaling to populate
   them, and updated all three clients' edit callers to use actual channel info
   instead of hardcoded defaults.

3. Channel parameter updates now automatically restart everyone's streams.
   Previously editing a channel's audio config persisted and broadcast a
   ChannelEvent::UPDATED, but no layer restarted streams — encoders/decoders are
   frozen at announce time. handle_channel_event now detects audio-config changes
   on the user's current channel and stop->starts each active local stream. The
   server reads the updated config on re-announce; peers wire up fresh decoders
   at the new ssrc.

All 29 CTest tests pass; Windows DLL + C# client build clean. Apple clients not
yet compile-verified (Windows environment).
2026-06-24 14:29:39 +02:00
..

VoiceCat — Design Documentation

VoiceCat is a self-hosted, server-based voice and text chat system in the spirit of classic TeamSpeak / Mumble: channel-based voice, channel and private text chat, and a single server you own and run. It deliberately avoids WebRTC. The media path is plain UDP, the control path is plain TCP, and both are encrypted.

This folder is the design spec. No code yet — these documents define the architecture, the wire protocol, the audio pipeline, the security model, and the dependency list, so that implementation can start from a shared, agreed plan.

Decisions locked so far

Area Decision
Code architecture Shared C++ core (libvoicecat) consumed by native UIs over a C ABI. Server reuses the same core.
Native clients macOS/iOS in Swift (SwiftUI; Swift↔C++ interop), Windows in C# (LibraryImport P/Invoke).
Control transport TCP + TLS 1.3 (mbedTLS)
Media transport UDP secured by TLS-exported keys + ChaCha20-Poly1305 AEAD — mandatory, no plaintext mode (see security.md)
Crypto libraries mbedTLS (TLS 1.3) + libsodium (AEAD, Argon2id, Ed25519) — both permissive, no GPL/LGPL anywhere
Voice codec Opus (libopus 1.6), per-channel configurable mono/stereo, bitrate, frame size, FEC/DTX
Audio DSP webrtc-audio-processing (APM) was the plan for AEC/NS/AGC/VAD, but has no working Windows/MSVC build upstream — v1 ships a lightweight energy/RMS VAD instead, no AEC/NS/AGC yet (see tech-stack.md, roadmap.md §2). NR is two-sided: sender can denoise, and each listener can denoise a specific other user locally — this plumbing exists but is currently inert pending a real DSP backend. Input gate supports VAD and PTT, client-configurable.
Identity Guests + admin-provisioned local accounts (Argon2id, SQLite). No self-serve registration; guests toggleable per server.
Text Ephemeral — live relay, no server-side history in v1.
Serialization Protocol Buffers for the control plane; custom binary for voice frames
Deployment One docker run, one static binary, or cmake --build — zero-config, secured by default (see deployment.md)

Document index

  1. architecture.md — System layers, the shared core, threading model, the C ABI, server design.
  2. protocol.md — The control protocol: framing, connection lifecycle, the full message catalog, encoding, versioning/extensibility.
  3. voice.md — The UDP media protocol: voice frame format, Opus configuration, multi-stream model, jitter buffer, packet-loss handling.
  4. security.md — Mandatory encryption (TLS 1.3 + exported-key media AEAD), server identity (TOFU), authentication, accounts, anti-replay, threat model.
  5. tech-stack.md — Concrete libraries with versions and rationale, the permissive-license rule, build tooling, per-platform notes.
  6. deployment.md — The "set it up in a few minutes" story: Docker, single binary, source build, zero-config defaults.
  7. roadmap.md — Milestones, what ships when, and the list of open questions still to resolve.

Design principles

  • One core, many faces. Protocol, crypto, Opus, networking, jitter buffering, and mixing live once in C++. UIs are thin. This keeps behavior identical across platforms and the security-sensitive code reviewed in a single place.
  • Boringly simple transport. TCP for control, UDP for media. No ICE, no SDP, no TURN. A user opens a port (or port-forwards) and runs a server.
  • Extensible from day one. Every message rides in a versioned envelope; capabilities are negotiated at connect time; unknown fields are ignored. File transfer, screen-audio, and moderation slot in without breaking older clients.
  • Real-time correctness. The audio thread never blocks, never allocates, never takes a lock. Network and audio communicate through lock-free ring buffers.
  • Encrypted, always. There is no unencrypted mode to misconfigure. The server has no plaintext listener; encryption is on because it can't be turned off. And it's free to the operator — the server self-provisions its key/cert on first run.
  • Stupid-easy to self-host. The target reaction is "oh, I (or my agent) can stand this up in a few minutes." One docker run, or one static binary, or a plain cmake --build — no certificate wrangling, no external database, sane defaults out of the box. Permissive licenses only (no GPL/LGPL) so it can be redistributed freely, including closed-source.

Glossary

  • Corelibvoicecat, the shared C++ library.
  • Control channel — the TCP/TLS connection carrying protobuf messages.
  • Media channel — the UDP connection carrying voice frames.
  • Stream — one audio source from one user (e.g. mic, screen audio, second device). A user may publish several streams at once; each is independently controllable.
  • Channel — a room in the channel tree. Voice is scoped to a channel.
  • Session — an authenticated connection; ties a TCP control channel to a UDP 5-tuple.
  • SFU relay — the server forwards Opus packets between channel members without decoding them (selective forwarding, no transcoding).