Choose your API to see its connection steps and session events.
Connect incoming and outgoing phone calls to GPT-Live through a SIP trunk. Your provider exchanges call audio with OpenAI, while your backend handles session configuration, call controls, and business logic. SIP signaling uses TLS, and GPT-Live requires SRTP for call audio.
- Handle an inbound call : receive an incoming-call webhook, then accept or reject the call.
- Place an outbound call : create a session to dial a phone number through your provider. No incoming-call webhook or accept request is needed.
Use a sideband connection when your backend needs to receive session events or send commands. It attaches to the existing conversation while SIP carries the audio. Assign one handler to each action so that duplicate webhook deliveries or events observed on multiple connections don’t execute tools twice.
Handle an inbound call #
Route incoming phone calls from your SIP provider to your OpenAI project. Your backend receives a webhook and decides whether to accept each call.
Configure your phone number and trunk
Use a SIP trunking provider, such as Twilio, to connect a phone number to GPT-Live. After purchasing a number from your provider:
- Open your project in OpenAI settings . UnderWebhooks , create awebhook endpoint subscribed to
live.transport.incoming. - Find your project ID under General . It starts with
proj_; use the same project that owns the webhook. - Point your provider’s SIP trunk at
sip:$PROJECT_ID@sip.api.openai.com;transport=tls. For European data residency, usesip:$PROJECT_ID@sip-eu.api.openai.com;transport=tlsinstead.
Confirm that GPT-Live SIP support is enabled for your project and that your provider’s SIP trunk is routed to that project before using this flow.
Receive the incoming call
When OpenAI receives a call for your project, it sends a live.transport.incoming event to your webhook endpoint. Verify the webhook signature and deduplicate deliveries, then accept or reject the call.
The webhook identifies a SIP call with data.type: "sip" and provides data.session_id. Use that session ID unchanged for every Live call action. Treat data.sip_headers as untrusted caller metadata, not authorization.
For example, the webhook body contains:
{
"object": "event",
"id": "evt_abc123",
"type": "live.transport.incoming",
"created_at": 1750287018,
"data": {
"type": "sip",
"session_id": "live_123",
"sip_headers": [
{ "name": "From", "value": "sip:+14155550123@sip.example.com" },
{ "name": "To", "value": "sip:+18005550100@sip.example.com" },
{ "name": "Call-ID", "value": "call-123@example.com" }
]
}
}
The created_at field is a Unix timestamp in seconds. Delivery headers include webhook-id for deduplication, webhook-timestamp for the delivery attempt, and webhook-signature for signature verification. See webhook verification examples for server-side handlers.
Existing integrations may still receive the deprecated live.call.incoming event, which has no data.type. During migration, handle both names and retain the old subscription until legacy deliveries and retries have drained. The same pending call can also emit a Realtime webhook; assign one handler to the accept/reject decision rather than accepting through both APIs.
Accept or reject the call
Apply your application’s authorization and routing rules. To accept the call, send an authenticated POST /v1/live/sessions/{session_id}/accept request with a top-level session object:
{
"session": {
"type": "live",
"model": "gpt-live-1",
"instructions": "You are answering an inbound support call.",
"audio": { "output": { "voice": "marin" } },
"delegation": { "type": "client" }
}
}
Use Authorization: Bearer $OPENAI_API_KEY from your trusted backend for call-control requests. Choose the voice and delegation mode at acceptance. SIP negotiates the audio format, so omit audio.format. The example selects client delegation; your backend must handle delegated work. See Delegation and tools for client and Responses configurations.
A successful acceptance returns 200 OK with an empty body after session initialization. Handle HTTP errors before treating the call as accepted.
To reject the call, send POST /v1/live/sessions/{session_id}/reject with a SIP status, such as { "status_code": 486 } for busy. The status must be an integer from 300 through 699. The first accept or reject decision wins; a later competing decision returns decision_already_made.
Attach your backend
After acceptance, connect a sideband WebSocket at wss://api.openai.com/v1/live/sessions/{session_id}/attach. Use the accepted session ID and the same project authentication and connection headers. Do not send session.start again.
SIP carries the call audio. Use the sideband for transcripts, delegation, tools, commands, and reflected audio. Choose one owner for each side effect, even if multiple connections observe an event.
Once connected, use the active call controls to send keypad tones, transfer the call, or hang up.
Place an outbound call #
Call a phone number through your SIP provider with Create session. Your provider handles the phone network connection while GPT-Live carries the conversation.
Outbound SIP calling is available to all organizations through the Live API, not the Realtime API call-creation endpoint.
Configure your trunk
Use a trunk that supports TLS signaling and SDES-SRTP media. Enable SRTP in your provider’s settings before placing a call.
Supply the trunk configuration with each request:
| Field | Value |
|---|---|
transport.destination |
The phone number to call, in E.164 format, such as +14155550123 . SIP URI destinations aren’t supported. |
transport.trunk.provider_url |
A provider endpoint such as sips:sip.example.com:5061 . The default port is5061 ;;transport=tcp is optional. |
transport.trunk.auth.type |
digest for SIP Digest authentication. |
transport.trunk.auth.username |
Your provider’s SIP username. |
transport.trunk.auth.password |
Your provider’s SIP password. |
transport.trunk.caller_number |
The caller phone number to send to your provider, in E.164 format. |
The provider endpoint must use sips: for TLS signaling. Don’t include credentials, paths, URI headers, or other URI parameters in the URL. Local hostnames and literal private or local IP addresses are rejected. Keep your OpenAI API key and SIP credentials on your server.
Create the session
Send POST /v1/live/sessions with your session configuration and transport.type: "sip". Choose the voice and delegation mode when creating the session. Omit audio.format because SIP negotiates the audio format.
This example uses curl and jq. Set OPENAI_API_KEY, SIP_USERNAME, and SIP_PASSWORD in your server environment, and replace the example provider endpoint and phone numbers with your own values. The example selects client delegation; your backend must handle delegated work.
jq -n \
--arg username "$SIP_USERNAME" \
--arg password "$SIP_PASSWORD" \
'{
"session": {
"model": "gpt-live-1",
"instructions": "Help the user schedule an appointment.",
"audio": { "output": { "voice": "marin" } },
"delegation": { "type": "client" }
},
"transport": {
"type": "sip",
"destination": "+14155550123",
"trunk": {
"provider_url": "sips:sip.example.com:5061",
"auth": {
"type": "digest",
"username": $username,
"password": $password
},
"caller_number": "+14155550100"
}
}
}' | curl https://api.openai.com/v1/live/sessions \
-H "Authorization: Bearer $OPENAI_API_KEY" \
-H "Content-Type: application/json" \
--data-binary @-
The request returns 200 OK after the session initializes:
{
"session": { "id": "live_123" },
"transport": { "type": "sip" }
}
This response doesn’t mean the call has been answered. It contains no SDP or trunk credentials. Preserve session.id unchanged for the sideband connection and call controls. You don’t need an incoming-call webhook or an accept request for an outbound call.
Monitor and end the call
Attach your backend to wss://api.openai.com/v1/live/sessions/{session_id}/attach using your OpenAI API key. Don’t send session.start again. The sideband connection carries conversation events, delegated work, and call progress while SIP carries the audio.
| Event | Meaning |
|---|---|
transport.ringing |
The provider reports ringing or early media. |
transport.answered |
The call has been answered and media is established. |
transport.failed |
Call setup failed after session initialization. Inspect error.code anderror.message . |
Each call-progress event includes event_id and session_id. Attach immediately after creation: the sideband only replays events from the preceding 3 seconds, so a later attachment can miss earlier call progress. Replayed events retain their original event IDs. Deduplicate events by event_id.
Once connected, use the active call controls to send keypad tones, transfer the call, or hang up. Keep the sideband open for session.closed and final usage as described in Usage and graceful close.
Handle limits and errors
Ringing is limited to 3 minutes, and a connected call is limited to 2 hours. These limits aren’t configurable in the creation request.
Invalid session configuration is returned on the creation request. Transport setup failures can return 502, and initialization timeouts return 504. After creation succeeds, monitor transport.failed for asynchronous setup failures.
Manage an active call #
These controls apply to both inbound and outbound SIP calls.
Send and receive keypad events
The sideband receives transport.dtmf.received when the caller presses a key. To send a DTMF tone to the SIP trunk, send a transport.dtmf.send client event over the sideband WebSocket. After the tone is successfully sent, the sideband receives transport.dtmf.sent.
If you supply an event_id, the notification includes it as client_event_id; failures return an error event with error.client_event_id.
Transfer or end the call
To transfer the call, send POST /v1/live/sessions/{session_id}/refer with { "target_uri": "sip:agent@example.com" } for your destination. To hang up, send POST /v1/live/sessions/{session_id}/hangup with no request body. Both return 200 OK with an empty body on success.
The transfer target becomes the SIP Refer-To header and can be a telephone URI such as tel:+14155550123 or a SIP URI such as sip:agent@example.com. Your provider handles the remaining transfer flow.
Keep the sideband open until session.closed supplies final usage. See Usage and graceful close for usage reporting and close reasons.
SIP network requirements #
GPT-Live SIP calls use separate network paths for signaling and media. Configure your network to allow both:
- SIP signaling:
sip.api.openai.comandsip-eu.api.openai.comare GeoIP-routed endpoints. Allow outbound TCP/TLS traffic to the addresses returned by DNS on port5061. - SRTP media: The API specifies a separate media IP address and UDP port in the negotiated SDP. Allow bidirectional SRTP traffic over UDP to and from the following CIDRs:
13.79.45.80/2823.98.140.64/2840.67.149.176/2840.83.204.240/28
Next steps with GPT-Live #
- Prompting guide : write instructions for your voice agent.
- Webhooks and server-side controls : manage a session from your backend.
- Delegation and tools : connect speech to your reasoning and tool backend.
- Managing sessions : handle transcripts, session state, and close.
- Latency and cost : understand latency and usage.
- GPT-Live partner integrations : find provider-specific setup guides and examples.
SIP is a protocol used to make phone calls over the internet. With SIP and the Realtime API you can direct incoming phone calls to the API.
Overview #
If you want to connect a phone number to the Realtime API, use a SIP trunking provider (e.g., Twilio). This is a service that converts your phone call to IP traffic. After you purchase a phone number from your SIP trunking provider, follow the instructions below.
Start by creating a webhook for incoming calls, through your platform.openai.com settings > Project > Webhooks.
Then, point your SIP trunk at the OpenAI SIP endpoint, using the project ID
for which you configured the webhook, e.g., sip:$PROJECT_ID@sip.api.openai.com;transport=tls.
For European data residency, use sip:$PROJECT_ID@sip-eu.api.openai.com;transport=tls instead.
To find your $PROJECT_ID, visit settings > Project > General. That page will display the project ID, which
will have a proj_ prefix.
When OpenAI receives SIP traffic associated with your project,
your webhook will be fired. The event fired will be a
realtime.call.incoming event,
like the example below:
POST https://my_website.com/webhook_endpoint
user-agent: OpenAI/1.0 (+https://platform.openai.com/docs/webhooks)
content-type: application/json
webhook-id: wh_685342e6c53c8190a1be43f081506c52 # unique id for idempotency
webhook-timestamp: 1750287078 # timestamp of delivery attempt
webhook-signature: v1,K5oZfzN95Z9UVu1EsfQmfVNQhnkZ2pj9o9NDN/H/pI4= # signature to verify authenticity from OpenAI
{
"object": "event",
"id": "evt_685343a1381c819085d44c354e1b330e",
"type": "realtime.call.incoming",
"created_at": 1750287018, // Unix timestamp
"data": {
"call_id": "some_unique_id",
"sip_headers": [
{ "name": "From", "value": "sip:+142555512112@sip.example.com" },
{ "name": "To", "value": "sip:+18005551212@sip.example.com" },
{ "name": "Call-ID", "value": "03782086-4ce9-44bf-8b0d-4e303d2cc590"}
]
}
}
From this webhook, you can accept or reject the call, using the call_id value from the webhook.
When accepting the call, you’ll provide the needed configuration
(instructions, voice, etc) for the Realtime API session.
Once established, you can set up a WebSocket and monitor the session as usual. The APIs to
accept, reject, monitor, refer, and hangup the call are documented below.
Accept the call #
Use the Accept call endpoint to
approve the inbound call and configure the realtime session that will answer it.
Send the same parameters you would send in a
create client secret
request, i.e., ensure the realtime model, voice, tools, or instructions are set before bridging the
call to the model.
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8curl -X POST "https://api.openai.com/v1/realtime/calls/$CALL_ID/accept" \
-H "Authorization: Bearer $OPENAI_API_KEY" \
-H "Content-Type: application/json" \
-d '{
"type": "realtime",
"model": "gpt-realtime-2.1",
"instructions": "You are Alex, a friendly concierge for Example Corp."
}'
The request path must include the call_id from the
realtime.call.incoming
webhook, and every request requires the Authorization header shown above. The
endpoint returns 200 OK once the SIP leg is ringing and the realtime session
is being established.
Reject the call #
Use the Reject call endpoint to
decline an invite when you do not want to handle the incoming call, (e.g., from
an unsupported country code.) Supply the call_id path parameter
and an optional SIP status_code (e.g., 486 to indicate “busy”) in the JSON
body to control the response sent back to the carrier.
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4curl -X POST "https://api.openai.com/v1/realtime/calls/$CALL_ID/reject" \
-H "Authorization: Bearer $OPENAI_API_KEY" \
-H "Content-Type: application/json" \
-d '{"status_code": 486}'
If no status code is supplied the API uses 603 Decline by default. A
successful request responds with 200 OK after OpenAI delivers the SIP
response.
Monitor call events #
After you accept a call, open a WebSocket connection to the same session to
stream events and issue realtime commands. Note that when connecting to an existing
call using the call_id parameter, the model argument is not used (as it has already been configured
via the accept endpoint).
WebSocket request
GET wss://api.openai.com/v1/realtime?call_id={call_id}
Query parameters
| Parameter | Type | Description |
|---|---|---|
call_id |
string | Identifier from the realtime.call.incoming webhook. |
Headers
Authorization: Bearer YOUR_API_KEY
The WebSocket behaves exactly like any other Realtime API connection. Send
response.create,
and other client events to control the call, and listen for server events to
track progress. See Webhooks and server-side controls
for more information.
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16import WebSocket from "ws";
const callId = "rtc_u1_9c6574da8b8a41a18da9308f4ad974ce";
const ws = new WebSocket(`wss://api.openai.com/v1/realtime?call_id=${callId}`, {
headers: {
Authorization: `Bearer ${process.env.OPENAI_API_KEY}`,
},
});
ws.on("open", () => {
ws.send(
JSON.stringify({
type: "response.create",
})
);
});
Redirect the call #
Transfer an active call using the
Refer call endpoint. Provide the
call_id as well as the target_uri that should be placed in the SIP Refer-To
header (for example tel:+14155550123 or sip:agent@example.com).
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4curl -X POST "https://api.openai.com/v1/realtime/calls/$CALL_ID/refer" \
-H "Authorization: Bearer $OPENAI_API_KEY" \
-H "Content-Type: application/json" \
-d '{"target_uri": "tel:+14155550123"}'
OpenAI returns 200 OK once the REFER is relayed to your SIP provider. The
downstream system handles the rest of the call flow for the caller.
Hang up the call #
End the session with the Hang up endpoint when your application should disconnect the caller. This endpoint can be used to terminate both SIP and WebRTC realtime sessions.
curl -X POST "https://api.openai.com/v1/realtime/calls/$CALL_ID/hangup" \
-H "Authorization: Bearer $OPENAI_API_KEY"
The API responds with 200 OK when it starts tearing down the call.
SIP signaling and media IP ranges #
Realtime SIP calls use separate network paths for signaling and media. To ensure proper operation, configure your network to allow signaling and media traffic as described below.
SIP signaling
sip.api.openai.com and sip-eu.api.openai.com are GeoIP-routed endpoints. Your network must allow
outbound TCP/TLS traffic to the addresses returned by DNS on port 5061.
SRTP media
The API specifies a separate media IP address and UDP port in the negotiated SDP. Your network must allow bidirectional SRTP traffic over UDP to and from the following CIDRs:
13.79.45.80/2823.98.140.64/2840.67.149.176/2840.83.204.240/28
Server examples #
The following is an example of a realtime.call.incoming handler. It accepts the call and then logs all the events from
the Realtime API.
For the Ruby example, set the OPENAI_API_KEY and OPENAI_WEBHOOK_SECRET
environment variables, then install the required dependencies with
gem install openai webrick async-websocket.
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69from flask import Flask, request, Response, jsonify, make_response
from openai import OpenAI, InvalidWebhookSignatureError
import asyncio
import json
import os
import requests
import time
import threading
import websockets
app = Flask(__name__)
client = OpenAI(webhook_secret=os.environ["OPENAI_WEBHOOK_SECRET"])
AUTH_HEADER = {"Authorization": "Bearer " + os.environ["OPENAI_API_KEY"]}
call_accept = {
"type": "realtime",
"instructions": "You are a support agent.",
"model": "gpt-realtime-2.1",
}
response_create = {
"type": "response.create",
"response": {
"instructions": ("Say to the user 'Thank you for calling, how can I help you'")
},
}
async def websocket_task(call_id):
try:
async with websockets.connect(
"wss://api.openai.com/v1/realtime?call_id=" + call_id,
additional_headers=AUTH_HEADER,
) as websocket:
await websocket.send(json.dumps(response_create))
while True:
response = await websocket.recv()
print(f"Received from WebSocket: {response}")
except Exception as e:
print(f"WebSocket error: {e}")
@app.route("/", methods=["POST"])
def webhook():
try:
event = client.webhooks.unwrap(request.data, request.headers)
except (InvalidWebhookSignatureError, ValueError) as e:
print("Invalid signature", e)
return Response("Invalid signature", status=400)
if event.type == "realtime.call.incoming":
requests.post(
"https://api.openai.com/v1/realtime/calls/"
+ event.data.call_id
+ "/accept",
headers={**AUTH_HEADER, "Content-Type": "application/json"},
json=call_accept,
).raise_for_status()
threading.Thread(
target=lambda: asyncio.run(websocket_task(event.data.call_id)),
daemon=True,
).start()
return Response(status=200)
if __name__ == "__main__":
app.run(port=8000)
Next steps #
Now that you’ve connected over SIP, use the left navigation or click into these pages to start building your realtime application.