Audio

Call sessions emit received audio and can stream a Buffer to the remote peer. The raw buffer format depends on the codec selected when constructing the Softphone.

Choose a codec

Codec	Raw input and decoded output	Playback example
OPUS/16000 (default)	16-bit signed little-endian PCM, 16 kHz, mono	ffplay -autoexit -f s16le -ar 16000 -ac 1 audio.raw
OPUS/48000/2	16-bit signed little-endian PCM, 48 kHz, stereo	ffplay -autoexit -f s16le -ar 48000 -ac 2 audio.raw
PCMU/8000	8-bit mu-law, 8 kHz, mono	ffplay -autoexit -f mulaw -ar 8000 -ac 1 audio.raw

Set the codec during construction:

const softphone = new Softphone({
  // SIP credentials...
  codec: "PCMU/8000",
});

Receive audio

For Opus codecs, audioPacket.payload contains decoded PCM. For PCMU, it contains mu-law bytes.

import fs from "node:fs";

const output = fs.createWriteStream(`${callSession.callId}.raw`);

callSession.on("audioPacket", (packet) => {
  output.write(packet.payload);
});

callSession.once("disposed", () => output.close());

Stream audio

Pass a complete raw audio buffer in the selected codec's input format. Streaming begins immediately.

import fs from "node:fs";

const streamer = callSession.streamAudio(fs.readFileSync("audio.raw"));
streamer.once("finished", () => console.log("Audio sent"));

Control the stream synchronously:

streamer.pause();
streamer.resume();
streamer.stop();
streamer.start(); // restart from the beginning

Pause streaming while the call is on hold, then resume after unholding.

The repository contains small test buffers used by the maintained demos. See the Examples page.