Accounts & API Keys
Calls, Messages, Stages
Webhooks
Schema
List Calls
Returns details for all calls
API key
The pagination cursor value.
Number of results to return per page.
Authorizations
API key
Query Parameters
The pagination cursor value.
Number of results to return per page.
Response
The version of the client that joined this call.
The reason the call ended.
unjoined- Client never joinedhangup- Client hung uptimeout- Call timed outconnection_error- Connection error
unjoined, hangup, timeout, connection_error Who was supposed to talk first when the call started. Typically set to FIRST_SPEAKER_USER for outgoing calls and left as the default (FIRST_SPEAKER_AGENT) otherwise.
FIRST_SPEAKER_AGENT, FIRST_SPEAKER_USER Messages spoken by the agent when the user is inactive for the specified duration. Durations are cumulative, so a message m > 1 with duration 30s will be spoken 30 seconds after message m-1.
The duration after which the message should be spoken.
The message to speak.
The behavior to exhibit when the message is finished being spoken.
END_BEHAVIOR_UNSPECIFIED, END_BEHAVIOR_HANG_UP_SOFT, END_BEHAVIOR_HANG_UP_STRICT The medium used initially by the agent. May later be changed by the client.
MESSAGE_MEDIUM_VOICE, MESSAGE_MEDIUM_TEXT BCP47 language code that may be used to guide speech recognition.
16Details about a call's protocol. By default, calls occur over WebRTC using the Ultravox client SDK. Setting a different call medium will prepare the server for a call using a different protocol. At most one call medium may be set.
The call will use WebRTC with the Ultravox client SDK. This is the default.
The call will use Twilio's "Media Streams" protocol. Once you have a join URL from starting a call, include it in your TwiML like so: <Connect><Stream url=${your-join-url} /></Connect> This works for both inbound and outbound calls.
The call will use a plain websocket connection. This is unlikely to yield an acceptable user experience if used from a browser or mobile client, but may be suitable for a server-to-server connection. This option provides a simple way to connect your own server to an Ultravox inference instance.
The sample rate for input (user) audio. Required.
The desired sample rate for output (agent) audio. If unset, defaults to the input_sample_rate.
The size of the client-side audio buffer in milliseconds. Smaller buffers allow for faster interruptions but may cause audio underflow if network latency fluctuates too greatly. For the best of both worlds, set this to some large value (e.g. 30000) and implement support for playback_clear_buffer messages. Defaults to 60.
The call will use Telnyx's media streaming protocol. Once you have a join URL from starting a call, include it in your TexML like so: <Connect><Stream url=${your-join-url} bidirectionalMode="rtp" /></Connect> This works for both inbound and outbound calls.
The call will use Plivo's AudioStreams protocol. Once you have a join URL from starting a call, include it in your Plivo XML like so: <Stream keepCallAlive="true" bidirectional="true" contentType="audio/x-l16;rate=16000">${your-join-url}</Stream> This works for both inbound and outbound calls.
0 < x < 1Indicates whether a transcript is optional for the call.
The number of errors in this call.
VAD settings for the call.
The delay to wait after the last speech frame before ending the user's "turn". Increasing this value makes the agent wait longer before speaking after the user stops speaking.
The minimum duration of user speech required to be considered a user turn.
The minimum duration of user speech required to interrupt the agent.