Files
voice-cat/docs
Talon 6071c8e238 fix(media): stop permanent voice loss after bad-network blip (protocol v2)
A bad UDP packet on a flaky link could permanently wedge the voice path,
unrecoverable even across app restarts. Three defects:

1. Anti-replay window was advanced from the UNAUTHENTICATED header seq
   before the AEAD tag was checked, and not rolled back on failure. One
   corrupted/forged frame shoved recv_highest_ far ahead, after which every
   legitimate frame was rejected as "too old" forever. Reorder to
   replay-check -> authenticate -> update (RFC 3711 3.3); the window now
   moves only after a successful tag check.

2. The wire seq was only the low 16 bits of the nonce counter (zero-extended
   on receive). After 65,536 frames the nonce desynced and all frames failed
   auth. Widen the voice frame seq u16 -> u64 (header 14 -> 20 bytes). The
   core owns all UDP framing, so Swift/C# clients need only a rebuild. This
   is a versioned wire change: VOICECAT_PROTOCOL_VERSION 1 -> 2, handshake
   rejects on mismatch.

3. Server leaked per-session UDP state on disconnect; unregister_session now
   frees udp_endpoints_/udp_tokens_/ssrc_to_session_.

Also add rate-limited dropped-frame logging to MediaRelay so a wedged media
path is observable. New regression tests in test_media_aead.cpp cover the
poison (fails on old code) and the 16-bit wrap. ctest --preset dev
-E external_pcm: 22/22 pass (external_pcm aborts on a pre-existing CoreAudio
shutdown race, unrelated).
2026-06-21 17:45:28 +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).