CallClient¶
Initiate an outbound call to a WebRTC client and connect it to the target connection.
Properties¶
Property | Type | Description |
---|---|---|
Required Properties | ||
client_name | string | The WebRTC name to call. The name should match the one in the token used to initialize a Gridspace web client. |
Optional Properties | ||
event_callback | string | URL to receive Conversation Event Callback requests for this conversation. |
live_talk_times | boolean | If true, the event_callback will receive
talk time events. Defaults to false . |
live_transcripts | boolean | If true, real-time transcription will be
enabled for the conversation and event_callback will receive
transcript events.
Defaults to false . |
live_topics | boolean | If true, real-time topic detection will be enabled for the
conversation and event_callback will receive
topic events. Defaults to false . |
do_record | boolean | If true, audio recordings of this call will be saved. A
recording of the full call will be made available via the
audio property of the
Conversation object. Each
member’s voice will be recorded in isolation as well and made
available via the channels property of the Conversation.
Defaults to true . |
processors | list | A list of strings indicating which Conversation Processors should be
run on the conversation once finished. The do_record
parameter must be true (the default) to process the
conversation, otherwise, this parameter is ignored. |
Example¶
{
"name": "CallClient",
"client_name": "user10548",
"event_callback": "http://myserver.com/callevents",
"live_talk_times": true,
"live_transcripts": true,
"live_topics": true,
"do_record": true,
"processors": ["transcribe", "findtopics"]
}