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@zapo-js/voip

v1.0.0

Published

WhatsApp VOIP (calling) plugin for zapo-js — MLow/WASM codec, SRTP, STUN and WebRTC/SCTP relay transport

Readme

@zapo-js/voip

WhatsApp VOIP / calling plugin for zapo-js.

Registers on WaClient via the plugin system and exposes everything at client.voip: MLow voice codec (WhatsApp's Opus variant through libmlow-wasm), RTP/SRTP, STUN, WebRTC/SCTP relay transport, and <call> signaling (offer / accept / preaccept / transport / relaylatency / mute / terminate).

Incoming <call>, call-class <ack>, and call <receipt> stanzas are handled automatically (prepend handlers return true so the core client does not double-ack).

Calls flow over WhatsApp relay servers using the MLow codec. This package handles audio calls with pre-recorded files or live 16 kHz mono PCM. Video is offered in signaling but not encoded.

Install

npm install zapo-js @zapo-js/voip libmlow-wasm

Peer dependencies:

| Package | Required | Purpose | | -------------- | -------------- | ----------------------------------------------- | | zapo-js | yes | WaClient and plugin host | | libmlow-wasm | yes | MLow encode/decode (WASM, no native build step) | | @roamhq/wrtc | for real calls | SCTP relay transport | | ffmpeg (CLI) | optional | Decode pre-recorded audio files (loadAudio) |

npm install @roamhq/wrtc

Node 20.9+. libmlow-wasm is ESM-only; the codec loads it via dynamic import().

Quick start

Importing from @zapo-js/voip applies WaClient type extensions (client.voip and voip_* events):

import { WaClient } from 'zapo-js'
import { voipPlugin, EndCallReason } from '@zapo-js/voip'

const client = new WaClient({
    store,
    sessionId: 'main',
    plugins: [voipPlugin()]
})

await client.connect()

client.on('voip_call_incoming', async (call) => {
    await client.voip.acceptCall(call.callId)
})

client.on('voip_call_state', (call) => {
    console.log(call.callId, call.stateData.state)
})

client.on('voip_call_inbound_audio', ({ call, pcm }) => {
    // Float32Array @ 16 kHz mono from the peer for this call
})

client.on('voip_call_outbound_audio_finished', (call) => {
    // preloaded file finished sending on this call
})

Multi-call (maxConcurrentCalls)

By default only one non-ended call is allowed at a time (maxConcurrentCalls: 1). Additional incoming offers are tracked with canAccept: false (no preaccept sent) until a slot frees; use call.canReject to decline manually.

Increase the limit explicitly to enable parallel calls (each with isolated relay/codec/audio):

plugins: [voipPlugin({ maxConcurrentCalls: 2 })]

Every audio/control API is scoped by callId. To mirror the same microphone into two active calls, call feedLiveAudio(callId, chunk) for each call.

Outgoing call – pre-recorded audio

loadAudio shells out to the ffmpeg binary (must be on PATH) to decode the file to 16 kHz mono PCM before encoding.

const callId = await client.voip.startCall({
    peerJid: '[email protected]'
})

await client.voip.loadAudio(callId, './hello.mp3')

// optional: react when the file finishes playing out
client.on('voip_call_outbound_audio_finished', (call) => {
    console.log('outbound audio done', call.callId)
})

// ... later
await client.voip.endCall(callId, EndCallReason.UserEnded)

Outgoing call – live audio

const callId = await client.voip.startCall({ peerJid: '[email protected]' })

client.voip.setExternalAudioMode(callId, true)

// feed 16 kHz mono Float32 chunks as they arrive;
// feedLiveAudio returns the buffered ms still queued to send
const bufferedMs = client.voip.feedLiveAudio(callId, pcmChunk)

// backpressure: pause your source above pauseMs, resume below resumeMs
const { pauseMs, resumeMs } = client.voip.getFeedWatermarksMs()

Incoming calls

The plugin registers incoming handlers; you only need to react to events:

client.on('voip_call_incoming', (call) => {
    console.log('ringing from', call.peerJid, call.callId)
})

// accept / reject / end
await client.voip.acceptCall(callId)
await client.voip.rejectCall(callId)
await client.voip.endCall(callId)

getCalls() returns every tracked call. getCall(callId) returns one call or null.

Events

Emitted on WaClient:

| Event | Payload | When | | ----------------------------------- | --------------------------------------- | ----------------------------------------- | | voip_call_incoming | CallInfo | Remote offer received | | voip_call_state | CallInfo | State transition | | voip_call_ended | CallInfo | Call finished | | voip_call_inbound_audio | { call: CallInfo; pcm: Float32Array } | Decoded peer audio received (16 kHz) | | voip_call_outbound_audio_finished | CallInfo | Preloaded outbound audio finished sending | | voip_call_error | Error | Engine error |

You can also use client.voip.on('call_state', ...) etc. for the manager-level events (CallManagerEvents).

client.voip API

| Method | Description | | --------------------------------------- | ------------------------------------------- | | startCall({ peerJid, isVideo?, audioFile?, peerDevices? }) | Place an outgoing call; returns callId | | acceptCall(callId) | Accept an incoming call | | rejectCall(callId, reason?) | Reject | | endCall(callId, reason?) | Hang up | | loadAudio(callId, path) | Load a file for outbound audio on that call | | setExternalAudioMode(callId, enabled) | Switch to live PCM input for that call | | feedLiveAudio(callId, Float32Array) | Push a capture chunk (external mode); returns buffered ms | | getLiveBufferMs(callId) | Buffered live-audio ms not yet sent | | getFeedWatermarksMs() | { pauseMs, resumeMs } backpressure thresholds | | setMute(callId, muted) | Mute/unmute local capture for that call | | getCall(callId) | One call or null | | getCalls() | All tracked calls | | on / off / once | Manager-level events |

Plugin options: maxConcurrentCalls?: number (default 1), logLevel?: LogLevel (caps VOIP diagnostics; defaults to the host client's level).

Codec

MLow runs through libmlow-wasm (≥ 0.1.1): 16 kHz, mono, 960-sample frames (60 ms), useSmpl: true, DTX enabled. No koffi, no bundled native libraries.

The signaling and media stack (RTP/SRTP, SCTP relay, codec, audio engine) is internal to the package; use client.voip and the events above.

Credits

The VOIP plugin was built by:

License

MIT