📞 WaCalls (Go)

Native WhatsApp voice calls in pure Go, straight from the browser. Built for native VoIP media, multi-account (multi-session) operation, and a modern browser client.

Overview · Architecture · Quick Start · API · Security

Overview

WaCalls pairs one or more WhatsApp accounts via QR code and lets you place and receive 1:1 voice calls from any browser on the LAN. The browser microphone is sent as raw 16 kHz PCM over a WebRTC data channel to the Go server, which encodes it with Meta's MLow codec and injects the media into WhatsApp's SRTP relay mesh — and the reverse path brings the peer's audio back to the browser.

The entire VoIP stack runs natively in pure Go: the MLow voice codec, RTP/SRTP packetization, STUN, the WebRTC/SCTP relay transport and the <call> signaling, integrated with whatsmeow and served to a React 19 client. There is no cgo and no native DLL — the MLow codec is a vendored pure-Go package, so a plain go build produces a self-contained binary with live audio.

Multiple WhatsApp accounts can be paired and operated side by side, each with its own pairing QR, connection status, and history. A single account can also run several concurrent 1:1 calls at once — one per browser operator — routed independently by call ID.

Status: stable. Outgoing and incoming 1:1 calls reach ACTIVE with bidirectional audio, and a single account can hold several of them concurrently. Sessions persist in wacalls.db (pure-Go SQLite).

Architecture

┌──────────────────────────────────────────────────────────────────────────┐
│                          BROWSER (React client)                            │
│   mic + speaker  ·  WebRTC data channel (16 kHz PCM)  ·  HTTP + SSE         │
└───────────────────────────────┬──────────────────────────────────────────┘
                                 │  POST /api/sessions/{sid}/calls/{id}/webrtc  (SDP)
                                 │  GET  /api/events                            (SSE)
                                 ▼
┌──────────────────────────── GO SERVER (cmd/server) ────────────────────────┐
│  SessionManager   registry of accounts (client + CallManager + bridge)     │
│  Broker           SSE hub (sessions, auth, call lifecycle fan-out)          │
│  Bridge           pion WebRTC bridge (16 kHz PCM data channel ⇄ call core)  │
│                                                                            │
│  internal/wa      VoipSocket adapter over whatsmeow                        │
│  internal/voip    call · signaling · media · transport · core · wanode     │
└───────────────┬──────────────────────────────────────┬────────────────────┘
                │ <call> signaling (Signal/USync)       │ SRTP media
                ▼                                        ▼
        ┌───────────────┐                    ┌──────────────────────┐
        │  WhatsApp WS  │                    │   WhatsApp relay      │
        │  (whatsmeow)  │                    │  (SRTP over SCTP/DC)  │
        └───────────────┘                    └──────────────────────┘

Layout

How a call flows

The core is internal/voip/call.CallManager, which drives a call end to end. Outgoing call sequence:

1. POST .../calls            → CallManager.StartCall(peerJid)
                               generates a callID, builds the <call> offer, sends it

2. Browser opens WebRTC      → POST .../calls/{id}/webrtc (SDP offer)
                               the bridge answers with an SDP answer (pion)

3. Peer accepts              → events.CallAccept → HandleCallAccept
                               server receives <relay> + hop-by-hop keys

4. Relay transport           → STUN binding/allocate on WhatsApp relays
                               ICE + DTLS + SCTP DataChannel connect (pion)

5. SRTP media flowing        → state goes ACTIVE
   ├── uplink   (you → peer): browser 16 kHz PCM (data channel) → MLow encode → SRTP → relay
   └── downlink (peer → you): relay → SRTP → MLow decode → 16 kHz PCM (data channel) → browser

6. Teardown                  → DELETE .../calls/{id} or events.CallTerminate
                               CallManager.EndCall + bridge cleanup

Each protocol step (hop-by-hop SRTP key derivation, RTP packetization at PT=120/16 kHz, STUN relay registration, relay-ack and <call> stanza parsing) is implemented and covered by tests in internal/voip (go test ./...).

Requirements

• Go 1.26+
• Node 22+ and npm (only to build/run the React client)

No C compiler, cgo, or native libraries are required — the MLow codec is vendored pure Go (internal/voip/media/mlow).

Quick Start

# clone and enter the project
git clone <repo-url> wacalls-go
cd wacalls-go

# Go dependencies
go mod download

# React client dependencies
cd client && npm install && cd ..

Run

go run ./cmd/server -addr :8080          # add -debug for verbose logs

Live audio works out of the box — the MLow codec is pure Go, so a plain build includes it. No build tags, no CGO_ENABLED, no DLLs.

Open http://localhost:8080, click New session, and scan the QR shown in the browser (it is also printed in the terminal) with WhatsApp → Linked devices. Add more accounts the same way and switch between them in the sidebar.

React client in dev mode

cd client
npm run dev      # Vite on :5173, proxies /api → http://localhost:8080

For production, build the static client and serve it from the Go server:

cd client && npm run build && cd ..
go run ./cmd/server -static client/dist -addr :8080

Server flags

API

All routes are session-scoped. Events stream over a single SSE channel, tagged with the originating sessionId.

Tests

go test ./...                 # media stack: SRTP, STUN, RTP, relay-ack, codec, state
cd client && npm run build    # client type-check + production build

Security

The API has no authentication — anyone with HTTP access can create accounts, place calls, and read history. Run it only on a trusted LAN. wacalls.db holds WhatsApp session credentials (secrets): do not commit it and keep it protected.

Contributors

This project builds on the work of:

@jotadev66 · @jobasfernandes · @edgardmessias · @w3nder

Acknowledgements

• whatsmeow — Go WhatsApp Web protocol library
• pion/webrtc — pure-Go WebRTC stack (ICE + DTLS + SCTP)
• whatsapp-rust — reference MLow codec implementation (ported to the vendored pure-Go internal/voip/media/mlow)
• zapo — VoIP media-stack reference

License

MIT