Create session
Create an ephemeral API token for use in client-side applications with the Realtime API. Can be configured with the same session parameters as the session.update
client event.
It responds with a session object, plus a client_secret
key which contains a usable ephemeral API token that can be used to authenticate browser clients for the Realtime API.
Request body
modalities
object
The set of modalities the model can respond with. To disable audio, set this to ["text"].model
string
The Realtime model used for this session.gpt-4o-realtime-preview
string
gpt-4o-realtime-preview-2024-10-01
string
gpt-4o-realtime-preview-2024-12-17
string
gpt-4o-mini-realtime-preview
string
gpt-4o-mini-realtime-preview-2024-12-17
string
instructions
string
The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.created
event at the start of the session.voice
string
The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy
,ash
,ballad
,coral
,echo
,fable
,onyx
,nova
,sage
,shimmer
, andverse
.voice
string
voice
string
alloy
string
ash
string
ballad
string
coral
string
echo
string
fable
string
onyx
string
nova
string
sage
string
shimmer
string
verse
string
input_audio_format
string
Defaults: pcm16
The format of input audio. Options are
pcm16
,g711_ulaw
, org711_alaw
. Forpcm16
, input audio must be 16-bit PCM at a 24kHz sample rate, single channel (mono), and little-endian byte order.pcm16
string
g711_ulaw
string
g711_alaw
string
output_audio_format
string
Defaults: pcm16
The format of output audio. Options are
pcm16
,g711_ulaw
, org711_alaw
. Forpcm16
, output audio is sampled at a rate of 24kHz.pcm16
string
g711_ulaw
string
g711_alaw
string
input_audio_transcription
object
Configuration for input audio transcription, defaults to off and can be set to
null
to turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.model
string
The model to use for transcription, current options are
gpt-4o-transcribe
,gpt-4o-mini-transcribe
, andwhisper-1
.language
string
The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en
) format will improve accuracy and latency.prompt
string
An optional text to guide the model's style or continue a previous audio segment. For
whisper-1
, the prompt is a list of keywords. Forgpt-4o-transcribe
models, the prompt is a free text string, for example "expect words related to technology".
turn_detection
object
Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
null
to turn off, in which case the client must manually trigger model response. Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech. Semantic VAD is more advanced and uses a turn detection model (in conjuction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.type
string
Defaults: server_vad
Type of turn detection.server_vad
string
semantic_vad
string
eagerness
string
Defaults: auto
Used only for
semantic_vad
mode. The eagerness of the model to respond.low
will wait longer for the user to continue speaking,high
will respond more quickly.auto
is the default and is equivalent tomedium
.low
string
medium
string
high
string
auto
string
threshold
number
Used only for
server_vad
mode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.prefix_padding_ms
integer
Used only for
server_vad
mode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.silence_duration_ms
integer
Used only for
server_vad
mode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user.create_response
boolean
Defaults: true
Whether or not to automatically generate a response when a VAD stop event occurs.interrupt_response
boolean
Defaults: true
Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversation
ofauto
) when a VAD start event occurs.
input_audio_noise_reduction
object
Defaults: null
Configuration for input audio noise reduction. This can be set to
null
to turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.type
string
Type of noise reduction.
near_field
is for close-talking microphones such as headphones,far_field
is for far-field microphones such as laptop or conference room microphones.near_field
string
far_field
string
tools
array
Tools (functions) available to the model.items
object
type
string
The type of the tool, i.e.
function
.function
string
name
string
The name of the function.description
string
The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).parameters
object
Parameters of the function in JSON Schema.
tool_choice
string
Defaults: auto
How the model chooses tools. Options are
auto
,none
,required
, or specify a function.temperature
number
Defaults: 0.8
Sampling temperature for the model, limited to [0.6, 1.2]. For audio models a temperature of 0.8 is highly recommended for best performance.max_response_output_tokens
integer or string
Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inf
for the maximum available tokens for a given model. Defaults toinf
.max_response_output_tokens
integer
max_response_output_tokens
string
inf
string
Response
The created Realtime session object, plus an ephemeral key
1 curl -X POST https://api.openai.com/v1/realtime/sessions \2 -H "Authorization: Bearer $OPENAI_API_KEY" \3 -H "Content-Type: application/json" \4 -d '{5 "model": "gpt-4o-realtime-preview",6 "modalities": ["audio", "text"],7 "instructions": "You are a friendly assistant."8 }'
1 {2 "id": "sess_001",3 "object": "realtime.session",4 "model": "gpt-4o-realtime-preview",5 "modalities": ["audio", "text"],6 "instructions": "You are a friendly assistant.",7 "voice": "alloy",8 "input_audio_format": "pcm16",9 "output_audio_format": "pcm16",10 "input_audio_transcription": {11 "model": "whisper-1"12 },13 "turn_detection": null,14 "tools": [],15 "tool_choice": "none",16 "temperature": 0.7,17 "max_response_output_tokens": 200,18 "client_secret": {19 "value": "ek_abc123",20 "expires_at": 123456789021 }22 }