Voice

Voice connections operate in a similar fashion to the Gateway connection. However, they use a different set of payloads and a separate UDP-based connection for voice data transmission. Because UDP is used for both receiving and transmitting voice data, your client must be able to receive UDP packets, even through a firewall or NAT (see UDP Hole Punching for more information). The Discord voice servers implement functionality (see IP Discovery) for discovering the local machine's remote UDP IP/Port, which can assist in some network configurations. While video from a camera feed is sent in the same connection as voice, streaming (the "Go Live" feature) uses seperate Gateway Opcodes and a second below voice connection.

To ensure that you have the most up-to-date information, please use version 7. Otherwise, we cannot guarantee that the events and commands documented here will reflect what you receive over the socket. Video is only fully supported on gateway v5 and above.

The first step in connecting to a voice server (and in turn, a guild's voice channel or private channel) is formulating a request that can be sent to the Gateway, which will return information about the voice server we will connect to. Because Discord's voice platform is widely distributed, users should never cache or save the results of this call. To inform the gateway of our intent to establish voice connectivity, we first send an Update Voice State payload.

If our request succeeded, the gateway will respond with two events—a Voice State Update event and a Voice Server Update event—meaning you must properly wait for both events before continuing. The first will contain a new key, session_id, and the second will provide voice server information we can use to establish a new voice connection.

With this information, we can move on to establishing a voice WebSocket connection.

Once we retrieve a session_id, token, and endpoint information, we can connect and handshake with the voice server over another secure WebSocket. Unlike the gateway endpoint we receive in an HTTP Get Gateway request, the endpoint received from our Voice Server Update payload does not contain a URL protocol, so some libraries may require manually prepending it with "wss://" before connecting. Once connected to the voice WebSocket endpoint, we can send an Opcode 0 Identify payload with our server_id, user_id, session_id, and token:

{
  "op": 0,
  "d": {
    "server_id": "41771983423143937",
    "user_id": "104694319306248192",
    "session_id": "my_session_id",
    "token": "my_token",
    "video": true,
    "streams": []
  }
}

The voice server should respond with an Opcode 2 Ready payload, which informs us of the SSRC, connection IP/port, supported encryption modes, and experiments the voice server expects:

{
  "op": 2,
  "d": {
    "ssrc": 1,
    "ip": "127.0.0.1",
    "port": 1234,
    "modes": [
      "aead_aes256_gcm_rtpsize",
      "aead_aes256_gcm",
      "aead_xchacha20_poly1305_rtpsize",
      "xsalsa20_poly1305_lite_rtpsize",
      "xsalsa20_poly1305_lite",
      "xsalsa20_poly1305_suffix",
      "xsalsa20_poly1305"
    ],
    "experiments": ["fixed_keyframe_interval"],
    "streams": []
  }
}

In order to maintain your WebSocket connection, you need to continuously send heartbeats at the interval determined in Opcode 8 Hello:

{
  "heartbeat_interval": 41250
}

{
  "op": 8,
  "d": {
    "v": 7,
    "heartbeat_interval": 41250
  }
}

This is sent at the start of the connection. Be warned that the Opcode 8 Hello structure differs by gateway version as shown in the above examples. Versions below v3 do not have op or d fields. v3 and above was updated to be structured like other payloads. Be sure to expect this different format based on your version.

This heartbeat interval is the minimum interval you should heartbeat at. You can heartbeat at a faster interval if you wish. For example, the web client uses a heartbeat interval of min(heartbeat_interval, 5000) if the gateway version is v4 or above, and heartbeat_interval * 0.1 otherwise. The desktop client uses the provided heartbeat interval if the gateway version is v4 or above, and heartbeat_interval * 0.25 otherwise.

After receiving Opcode 8 Hello, you should send Opcode 3 Heartbeat—which contains an integer nonce—every elapsed interval:

The gateway may request a heartbeat from the client in some situations by sending an Opcode 3 Heartbeat. When this occurs, the client should immediately send an Opcode 3 Heartbeat without waiting the remainder of the current interval.

{
  "op": 3,
  "d": 1501184119561
}

In return, you will be sent back an Opcode 6 Heartbeat ACK that contains the previously sent nonce:

{
  "op": 6,
  "d": 1501184119561
}

Once we receive the properties of a voice server from our Ready payload, we can proceed to the final step of voice connections, which entails establishing and handshaking a connection for voice data. First, we connect to the IP and port provided in the Ready payload. We then send an Opcode 1 Select Protocol with details about our connection:

¹ These fields are only used to receive voice data. If you do not care about receiving voice data, you can randomize these values.

{
  "op": 1,
  "d": {
    "protocol": "udp",
    "data": {
      "address": "127.0.0.1",
      "port": 1337,
      "mode": "xsalsa20_poly1305_lite"
    },
    "experiments": ["fixed_keyframe_interval"]
  }
}

Finally, the voice server will respond with an Opcode 4 Session Description that includes the mode and secret_key, a 32 byte array used for sending and receiving voice data:

{
    "op": 4,
    "d": {
        "audio_codec": "opus",
        "media_session_id": "89f1d62f166b948746f7646713d39dbb",
        "mode": "xsalsa20_poly1305_lite",
        "secret_key": [ ...251, 100, 11...],
        "video_codec": "H264"
    }
}

Sometimes, the voice server will later send an Opcode 14 Session Update to indicate an update to the voice session:

We can now start encrypting/decrypting and sending/receiving voice data over the previously established connection.

UDP is the most likely protocol that clients will use. First, we open a UDP connection to the IP and port provided in the Ready payload. If required, we can now perform an IP Discovery using this connection. Once we've fully discovered our external IP and UDP port, we can then tell the voice WebSocket what it is by sending a Select Protocol as outlined above, and receive our Session Description to begin sending/receiving voice data.

Generally routers on the Internet mask or obfuscate UDP ports through a process called NAT. Most users who implement voice will want to utilize IP discovery to find their external IP and port which will then be used for receiving voice communications. To retrieve your external IP and port, send the following UDP packet to your voice port (all numeric are big endian):

Voice data sent to and received from Discord should be encoded with Opus, using two channels (stereo) and a sample rate of 48kHz. Voice Data is sent using a RTP Header, followed by encrypted Opus audio data. Voice encryption uses the key passed in Session Description and the nonce formed with the 12 byte header appended with 12 null bytes, if required. Discord encrypts with the libsodium encryption library.

When receiving data, the user who sent the voice packet is identified by caching the SSRC and user IDs received from Speaking events. At least one Speaking event for the user is received before any voice data is received, so the user ID should always be available.

WebRTC allows for direct peer-to-peer voice connections, and is most commonly used in browsers. To use WebRTC, you must first send a Select Protocol payload as outlined above, with the protocol field set to webrtc, and data set to the client's WebRTC SDP. The voice server will respond with a Session Description payload, with the sdp field set to the server's WebRTC SDP. The client can then use this SDP to establish a WebRTC connection.

To notify the voice server that you are speaking or have stopped speaking, send an Opcode 5 Speaking payload:

¹ For gateway v4 and below, this field is a boolean.

² Only sent by the voice server.

³ Not sent by the voice server.

{
  "op": 5,
  "d": {
    "speaking": 5,
    "delay": 0,
    "ssrc": 1
  }
}

When a different user's speaking state is updated, the voice server will send an Opcode 5 Speaking payload:

{
  "op": 5,
  "d": {
    "speaking": 5,
    "ssrc": 2,
    "user_id": "852892297661906993"
  }
}

The following flags can be used as a bitwise mask. For example 5 would be priority and voice.

When there's a break in the sent data, the packet transmission shouldn't simply stop. Instead, send five frames of silence (0xF8, 0xFF, 0xFE) before stopping to avoid unintended Opus interpolation with subsequent transmissions.

Likewise, when you receive these five frames of silence, you know that the user has stopped speaking.

When your client detects that its connection has been severed, it should open a new WebSocket connection. Once the new connection has been opened, your client should send an Opcode 7 Resume payload:

{
  "op": 7,
  "d": {
    "server_id": "41771983423143937",
    "session_id": "my_session_id",
    "token": "my_token"
  }
}

If successful, the voice server will respond with an Opcode 9 Resumed to signal that your client is now resumed:

{
  "op": 9,
  "d": null
}

If the resume is unsuccessful—for example, due to an invalid session—the WebSocket connection will close with the appropriate close code. You should then follow the Connecting flow to reconnect.

When a user disconnects from voice, the voice server will send an Opcode 13 Client Disconnect payload:

When received, the SSRC of the user should be discarded.

{
  "op": 13,
  "d": {
    "user_id": "852892297661906993"
  }
}

For analytics, the client may want to receive information about the voice backend's current version. This is only available on gateway v6 and above. To do so, send an Opcode 16 Voice Backend Version with an empty payload:

{
  "op": 16,
  "d": {}
}

In response, the voice server will send an Opcode 16 Voice Backend Version payload with the versions:

{
  "op": 16,
  "d": {
    "voice": "0.9.1",
    "rtc_worker": "0.3.35"
  }
}