Unity

com.aurix.voice package for Unity 2021.3+ — C# client, AudioSource-based playback, desktop, mobile and WebGL.

Install#

Window → Package Manager → + → Add package from git URL:

https://github.com/aurix-voice/aurix.git?path=/sdk/unity#v1.13.0

The package (com.aurix.voice) contains the managed client, the native aurix-client library for Windows, macOS, Linux and Android, a WebGL bridge, and samples (VoiceQuickstart, WebGLQuickstart, MirrorProximity, NetcodeProximity, Concentus fallback codec) under Package Manager → Samples.

Connect and join#

using Aurix;

var voice = GetComponent<AurixVoiceBehaviour>();   // on a persistent object
voice.WebSocketUrl = token.Endpoint.WsUrl;
voice.Token = token.Token;
voice.ChannelId = token.Channels[0].ChannelId;     // joined right after connect
await voice.Connect();                              // microphone, playback and PTT are handled by the behaviour

AurixVoiceBehaviour owns capture (Unity's Microphone API), playback and push-to-talk and exposes the underlying AurixVoiceClient as voice.Client for everything else — JoinChannelAsync(Guid), LeaveChannelAsync, TransmitToChannelAsync, SetParticipantVolumeAsync, chat, moderation. Use AurixVoiceClient directly only when you bring your own audio pipeline.

Playback and spatial audio#

Every remote participant gets an AurixParticipantAudioSource — an ordinary AudioSource you can parent to the player's avatar. Unity's spatial blend, roll-off and mixer groups apply. AurixListenerTap reports who is speaking for UI rings; AurixGameAudioDucker lowers your music/SFX mixer group while someone talks; AurixLipSync exposes visemes for avatar mouths.

For positional channels send the local player's pose:

await voice.Client.UpdatePositionAsync(channelId, myUserId, new Position3D { X = p.x, Y = p.y, Z = p.z },
    new Orientation3D { ForwardX = f.x, ForwardY = f.y, ForwardZ = f.z, UpX = u.x, UpY = u.y, UpZ = u.z });   // 10–20 Hz

Push-to-talk, VAD, mute#

voice.PushToTalkKey = KeyCode.V;        // or PushToTalkAction (Input System); PushToTalkToggle for toggle mode
voice.Vad.Threshold = 0.02f; voice.Vad.HangoverFrames = 8;
voice.SetInputGain(1f);
voice.SetMuted(false);

Mobile#

• Android: RECORD_AUDIO is requested at first capture (PermissionState, RetryMicrophonePermission()); capture goes through Unity's Microphone API.
• iOS: add NSMicrophoneUsageDescription to the player settings.
• Both platforms suspend capture when the app goes to the background and resume on return.

Reconnects#

The client keeps reconnecting with its token while valid (60 minutes). Set voice.TokenRefresher = async ct => await FetchTokenFromYourBackend(); and it fetches a fresh one when a reconnect needs it.

Channel patterns#

The transmit channel is a client-side choice: join several, send to one, hear all with receive. Positional audio needs positions roughly 10 times a second; between updates the node interpolates.

Latency and quality#

Expect 30–60 ms mouth-to-ear on the same continent over QUIC, more over TURN. Opus runs at 32 kbps with DTX by default; the spending ladder may lower it to 16 kbps when your forecast exceeds the limit. Server-side noise suppression is on by default and costs nothing on the client.

Testing without a full build#

• Two players: the rooms demo in a browser plus your build — they share a cell, so a token from your backend joins the same channel by name.
• Mic path: an echo channel plays your own voice back through the real pipeline.
• Network: throttle UDP on one machine and watch the transport fall back to WebRTC / WebSocket in the SDK's connection state.

Troubleshooting#