Create Call

If set, the agent should speak first.

If set, the user should speak first.

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.

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 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 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 WebRTC with the Ultravox client SDK. This is the default.

The minimum duration of user speech required to interrupt the agent. This works the same way as minimumTurnDuration, but allows for a higher threshold for interrupting the agent. (This value will be ignored if it is less than minimumTurnDuration.)

Defaults to "0.09s" (90ms) as a starting point, but there's nothing special about this value.

The minimum duration of user speech required to be considered a user turn. Increasing this value will cause the agent to ignore short user audio. This may be useful in particularly noisy environments, but it comes at the cost of possibly ignoring very short user responses such as "yes" or "no".

Defaults to "0s" meaning the agent considers all user audio inputs (that make it through built-in noise cancellation).

The minimum amount of time the agent will wait to respond after the user seems to be done speaking. Increasing this value will make the agent less eager to respond, which may increase perceived response latency but will also make the agent less likely to jump in before the user is really done speaking.

Built-in VAD currently operates on 32ms frames, so only multiples of 32ms are meaningful. (Anything from 1ms to 31ms will produce the same result.)

Defaults to "0.384s" (384ms) as a starting point, but there's nothing special about this value aside from it corresponding to 12 VAD frames.

If set, the agent should speak first.

If set, the user should speak first.

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.

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 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 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 WebRTC with the Ultravox client SDK. This is the default.

The minimum duration of user speech required to interrupt the agent. This works the same way as minimumTurnDuration, but allows for a higher threshold for interrupting the agent. (This value will be ignored if it is less than minimumTurnDuration.)

Defaults to "0.09s" (90ms) as a starting point, but there's nothing special about this value.

The minimum duration of user speech required to be considered a user turn. Increasing this value will cause the agent to ignore short user audio. This may be useful in particularly noisy environments, but it comes at the cost of possibly ignoring very short user responses such as "yes" or "no".

Defaults to "0s" meaning the agent considers all user audio inputs (that make it through built-in noise cancellation).

The minimum amount of time the agent will wait to respond after the user seems to be done speaking. Increasing this value will make the agent less eager to respond, which may increase perceived response latency but will also make the agent less likely to jump in before the user is really done speaking.

Built-in VAD currently operates on 32ms frames, so only multiples of 32ms are meaningful. (Anything from 1ms to 31ms will produce the same result.)

Defaults to "0.384s" (384ms) as a starting point, but there's nothing special about this value aside from it corresponding to 12 VAD frames.