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The listen verb sends real-time audio streams to your external server over a WebSocket connection for processing. The Call Recording feature relies on this verb. This table outlines the properties related to the audio streams sent by the listen verb:
PropertiesDescription
Format16-bit
EncodingPCM
Sample rateuser-specified
Connection typewebsocket
One text frame is sent immediately after the WebSocket connection is established to send a JSON string with call attributes over an HTTP request. Additional metadata can also be added to this payload using the metadata parameter. The listen verb can also be nested in a dial or config verb, allowing the audio for a call between two parties to be sent to a remote WebSocket server. 
{
  "verb": "listen",
  "url": "wss://myrecorder.example.com/calls",
  "mixType" : "stereo"
}

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Configuration

The full set of configuration parameters:
ParametersDescriptionRequired
actionHookA webhook to receive an HTTP POST when the listen operation ends. The information will include the duration of the audio stream, and also a digits property if the recording was terminated by a DTMF key.Yes
bidirectionalAudio.enabledSends audio over the WebSocket connection back to the call. The default value is true.No
bidirectionalAudio.sampleRateIs required when streaming is enabled. This parameter specifies the sampling rate of the audio in Hz that is sent over the WebSocket connection back to the call.No
bidirectionalAudio.streamingBreaks the raw audio into segments (chunks) and sends them over the WebSocket connection back to the call. The audio is expected to be in linear PCM format (uncompressed raw audio format without headers). The default value is false. This parameter works if bidirectionalAudio.enabled is activated.No
disableBidirectionalAudioThis parameter is deprecated. If set to true, it will disable bidirectional audio, which is equivalent to setting bidirectionalAudio.enabled = false. The default value is false.No
finishOnKeyThe set of digits that can end the listen action.No
maxLengthThe maximum length of the listened audio stream in seconds.No
metadataArbitrary data to add to the JSON payload sent to the remote server when the WebSocket connection is first established.No
mixTypeThe following types can be specified:
- mono — sends a single channel,
- stereo — sends a dual channel of both calls in a bridge,
- mixed — sends audio from both calls in a bridge in a single mixed audio stream. The default value is mono.		No
passDtmfIf this parameter is true, detected DTMF digits will be sent over WebSocket as JSON text frames. The default value is false.No
playBeepEnable a beep sound when the listen operation starts. The default value is false.No
sampleRateThe sample rate of audio to send. Allowable values: 8000, 16000, 24000, 48000, or 64000. The default value is 8000.No
timeoutThe number of seconds of silence that terminates the listen operation.No
transcribeA nested transcribe verb.No
urlThe URL of the remote server to connect to.Yes
wsAuth.usernameThe HTTP basic auth username to use on the WebSocket connection.No
wsAuth.passwordThe HTTP basic auth password to use on the WebSocket connection.No

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Passing DTMF

Any DTMF digits entered by the far end party on the call can optionally be passed to the WebSocket server as JSON text frames by setting the passDtmf property to true. Each DTMF entry is reported separately in a payload containing the specific DTMF key that was entered, along with its duration reported in RTP timestamp units. The payload example: 
{
  "event": "dtmf",
  "dtmf": "2",
  "duration": "1600"
}

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Bidirectional Audio

Audio can also be sent back over the WebSocket. This audio, if supplied, will be played out to the caller.
Bidirectional audio is not supported when the listen is nested in the context of a dial verb.The far-end WebSocket server supplies bidirectional audio by sending a JSON text frame over the WebSocket connection:
{
  "type": "playAudio",
  "data": {
"audioContent": "base64-encoded content..",
"audioContentType": "raw",
"sampleRate": "16000"
  }
}
In the example above, raw (headerless) audio is sent. The audio must comply with the standard properties of encoding and format, with a configurable sample rate of either 8000, 16000, 24000, 32000, 48000, or 64000 Hz.Alternatively, a wave file format can be supplied by using type wav (or wave), and in this case, no sampleRate property is needed. In all cases, the audio must be base64 encoded when sent over the socket.If multiple playAudio verbs are sent before the first has finished playing, they will be queued and played in order. You may have up to 10 queued playAudio verbs at any time.Once a playAudio verb has finished playing out the audio, a playDone JSON text frame will be sent over the WebSocket connection for confirmation.
{
  "type": "playDone"
}
A killAudio verb can be sent by the WebSocket server to stop the playback of audio that was started via a previous playAudio verb:
{
  "type": "killAudio"
}
If the WebSocket connection wishes to end the listen, it can send a disconnect verb:
{
  "type": "disconnect"
}