***
id: c9602999-48ee-454f-99e8-95eccb283cb9
title: connectSip
slug: /node/reference/voice/call/connect-sip
description: connectSip method for the Call class.
max-toc-depth: 3
----------------
[call]: /docs/server-sdk/v4/node/reference/voice/call
[callstate]: /docs/server-sdk/v4/node/reference/voice/call-state
[disconnected]: /docs/server-sdk/v4/node/reference/voice/call/disconnected
[sipcodec]: /docs/server-sdk/v4/node/reference/voice/types#sipcodec
[sipheader]: /docs/server-sdk/v4/node/reference/voice/types#sipheader
[voiceplaylist]: /docs/server-sdk/v4/node/reference/voice/playlist
### connectSip
* **connectSip**(`params`): `Promise`\<[`Call`][call]>
Attempt to connect an existing call to a new outbound SIP call. The two devices will hear each other.
You can wait until the new peer is disconnected by calling [disconnected][disconnected].
#### Parameters
Object containing the parameters for connecting the call to a SIP endpoint.
The party the call is coming from. Must be a SignalWire number or SIP endpoint that you own.
The party you are attempting to call.
The time, in seconds, the call will ring before it is considered unanswered.
Array of [`SipHeader`][sipheader] objects. Must be X- headers only, see example below.
Array of desired codecs in order of preference. Supported values are `PCMU`, `PCMA`, `OPUS`, `G729`, `G722`, `VP8`, `H264`. Default is parent leg codec(s). See [`SipCodec`][sipcodec].
If `true`, WebRTC media is negotiated. Default is parent leg setting.
Non-negative value, in seconds, to use for the SIP `Session-Expires` header. If `0` or unset, SignalWire will pick the default (typically `600`).
The maximum price in USD acceptable for the call to be created. If the rate for the call is greater than this value, the call will not be created. If not set, all calls will be created. Price can have a maximum of four decimal places, i.e. `0.0075`.
Ringback audio to play to the call leg. You can play audio, TTS, silence or ringtone. See [`VoicePlaylist`][voiceplaylist].
Webhook URL to which SignalWire will send call status change notifications. See the payload specifications under [`CallState`][callstate].
An array of event names to be notified about. Allowed values are `created`, `ringing`, `answered`, and `ended`.
#### Returns
`Promise`\<[`Call`][call]>
A promise that resolves to a [`Call`][call] object that you can use to control the new peer.
The promise resolves only after the new peer picks up the call.
#### Example
In this example, we connect a inbound call to a internal SIP endpoint. We play a ringback tone to the inbound call leg
while waiting for the SIP endpoint to answer. Once the SIP endpoint answers, we wait for the peer to hangup and then hangup the inbound call.
```js
import { SignalWire, Voice} from "@signalwire/realtime-api";
const client = await SignalWire({ project: "ProjectID Here", token: "Token Here" })
const voiceClient = client.voice;
// Ringback tone
const ringback = new Voice.Playlist().add(
Voice.Playlist.Ringtone({
name: "it"
})
);
await voiceClient.listen({
topics: ["office"],
onCallReceived: async (call) => {
// Answer the call
call.answer();
// Connect the call to the device builder plan
let peer = await call.connectSip({
// Replace the to and from with valid SIP endpoint domains
to: "sip:example@example.sip.signalwire.com",
from: `sip:${call.from}@example.sip.signalwire.com`,
ringback: ringback
});
// Listen to peer state changes
await peer.listen({
onStateChanged: (state) => {
console.log("Peer state changed:", state.state);
}
})
// wait for peer to hangup
await peer.disconnected();
console.log("The peer hungup");
// hangup the call
call.hangup();
}
});
```