Files
voice-cat/docs
Talon 118ca5129f fix(protocol): deliver self-initiated state changes to the actor too
A connected Windows client would randomly snap from its joined channel
back to Lobby. Root cause was a state-sync inconsistency, not a drop:
the server delivered self-initiated state changes (channel join/leave,
stream announce/stop) only as a private *Result to the actor and
broadcast the authoritative UserEvent::UPDATED to everyone else. The
core never applied the result to its SessionModel, so vc_list_users()
kept self in the old channel; the Windows HandleUserUpdated rebuilds
_currentChannelId from vc_list_users() on any user's UPDATED event, so
the next unrelated event surfaced the stale self-channel.

Fix, per the response-vs-broadcast contract now documented in
docs/protocol.md §6: the *Result is pure ack/correlation/actor-private
payload; the resulting state change is broadcast to every client
INCLUDING the actor, and clients apply it to their local model rather
than re-deriving own state from a *Result.

- server: join/leave/stream announce+stop broadcast with exclude=0
- server: text fan-out includes the sender (channel + private echo)
- core: response handlers no longer mutate session_model_
- windows: drop optimistic text echo; render own message via the relay
- docs/protocol.md §6: document the response-vs-broadcast contract

Registry-level admin broadcasts (move/mute/kick/channel CRUD) already
used exclude=0 and were correct. ctest build/m1-dev 18/18 green;
VoiceCat.App builds 0 warnings.

Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-17 20:48:50 +02:00
..
2026-06-17 00:52:02 +02:00
2026-06-17 00:52:02 +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).