Add Messaging, Voice, and Authentication to your apps with Vonage Communications APIs
Visit the Vonage API Developer PortalAudio Connector lets you send raw audio (PCM 16 khz/16bit) streams from a live Vonage Video session to external services such as AWS, GCP, Azure, etc., through your own servers for further processing and analysis.
Using Audio Connector, you can send audio streams individually or mixed. You can identify the speaker by sending the audio streams individually by opening multiple WS connections.
The further processing of audio streams in real-time and offline enables building capabilities such as captions, transcriptions, translations, search and index, content moderation, media intelligence, Electronic Health Records, sentiment analysis, etc.
You can also use Audio Connector to use a WebSocket connection to publish audio to an OpenTok session.
Audio Connector is enabled by default for all projects, and it is a usage-based product. Audio Connector usage is charged based on the number of audio streams of participants (or stream IDs) that are sent to the WebSocket server. The Audio Connector feature is only supported in routed sessions (sessions that use the OpenTok Media Router). You can send up to 50 audio streams from a single session at a time.
Notes:
This page includes the following sections:
To start an Audio Connector WebSocket connection, use the OpenTok REST API.
You can also you can also start an Audio Connector WebSocket connection using the OpenTok server SDKs:
OpenTok.connectAudioStream() method.opentok.websocketConnect() method.OpenTok->connectAudio()] method.opentok.connect_audio_to_websocket() method.opentok.websocket.connect() methodOpenTok.StartBroadcast() methodFor more details, see the Audio Connector REST API documentation.
The initial message sent on the established WebSocket connection is text-based, containing
a JSON payload. The JSON details the audio format in content-type, along with any other metadata
that you put in the headers property of the body in the POST request to start the WebSocket
connection:
{
"content-type":"audio/l16;rate=16000",
"CUSTOM-HEADER-1": "value-1",
"CUSTOM-HEADER-2": "value-2"
}
Messages that are binary represent the audio of the call. The audio codec supported on the WebSocket interface is Linear PCM 16-bit, with a 16kHz sample rate. Each message includes one 640-byte frame of data (20ms of audio) at 50 frames (messages) per second.
When audio in the streams included in the WebSocket is muted, a text message is sent with the
following JSON payload (with active set to false):
{
"content-type":"audio/l16;rate=16000",
"method": "update",
"event": "websocket:media:update",
"active": false,
"CUSTOM-HEADER-1": "value-1",
"CUSTOM-HEADER-2": "value-2"
}
(The CUSTOM-HEADER properties in this example represent metadata that you include
in the headers property of the body in the POST request to start the WebSocket
connection.)
Audio may be muted because all clients stop publishing audio or as a result of a force mute moderation event.
When audio of one of the streams resumes, a text message is sent with the
following JSON payload (with active set to true):
{
"content-type":"audio/l16;rate=16000",
"method": "update",
"event": "websocket:media:update",
"active": true,
"CUSTOM-HEADER-1": "value-1",
"CUSTOM-HEADER-2": "value-2"
}
When the Audio Connector WebSocket stops because of a call to the force disconnect REST method or because the 6-hour time limit is reached (see Stopping a WebSocket connection), a text message is sent with the following JSON payload:
{
"content-type":"audio/l16;rate=16000",
"method": "delete",
"event": "websocket:disconnected",
"CUSTOM-HEADER-1": "value-1",
"CUSTOM-HEADER-2": "value-2"
}
This message marks the termination of the WebSocket connection.
(The CUSTOM-HEADER properties in this example represent metadata that you include
in the headers property of the body in the POST request to start the WebSocket
connection.)
When your WebSocket server closes the connection, the OpenTok connection for the call also ends. In each client connected to the session, the OpenTok client-side SDK dispatches events indicating the connection ended (just as it would when other clients disconnect from the session).
You can disconnect the Audio Connector WebSocket connection using the force disconnect REST method. Use the connection ID of the Audio Connector WebSocket connection with this method.
As a security measure, the WebSocket will be closed automatically after 6 hours.
Audio Connector will make a few attempts to re-establish a WebSocket connection that closes unexpectedly (for example, if the WebSocket closes without resulting from a call to the force disconnect REST method).
You can use the Audio Connector WebSocket connection to send audio data from the WebSocket connection to a stream published in an OpenTok session (in addition to having WebSocket connection receive audio from the session). Set the bidirectional property to true in the data you send with the REST API method to start the Audio Connector.
See Binary audio messages for details on the format of the audio data to send via the WebSocket connection.
When creating the token used by the Audio Connector, you can add token data to identify the the Audio Connector stream. (The OpenTok client libraries include methods for inspecting the connection data for the connection of a stream in session.)
Note: The bidirectional option is only available in the REST API. It is not currently supported in the OpenTok server SDKs.
See the demo-video-node-audio_connector project for a sample Node application that uses Audio Connector.
See this blog post.