This is an experimental technology
Because this technology's specification has not stabilized, check the compatibility table for usage in various browsers. Also note that the syntax and behavior of an experimental technology is subject to change in future versions of browsers as the specification changes.

The createOffer() method of the RTCPeerConnection interface initiates the creation of an SDP offer which includes information about any MediaStreamTracks already attached to the WebRTC session, codec and options supported by the browser, and any candidates already gathered by the ICE agent, for the purpose of being sent over the signaling channel to a potential peer to request a connection or to update the configuration of an existing connection.

The return value is a Promise which, when the offer has been created, is resolved with a RTCSessionDescription object containing the newly-created offer.


aPromise = myPeerConnection.createOffer([options]);

myPeerConnection.createOffer(successCallback, failureCallback[, options]) <span title="This deprecated API should no longer be used, but will probably still work.">class="icon-thumbs-down-alt"> </span>


options Optional
An RTCOfferOptions dictionary providing options requested for the offer.

RTCOfferOptions dictionary

The RTCOfferOptions dictionary is used to customize the offer created by this method.

iceRestart Optional
To restart ICE on an active connection, set this to true. This will cause the returned offer to have different credentials than those already in place. If you then apply the returned offer, ICE will restart. Specify false to keep the same credentials and therefore not restart ICE. The default is false.
voiceActivityDetection Optional
Some codecs and hardware are able to detect when audio begins and ends by watching for "silence" (or relatively low sound levels) to occur. This reduces network bandwidth used for audio by only sending audio data when there's actually something to broadcast. However, in some cases, this is unwanted. For example, in the case of music or other non-voice transmission, this can cause loss of important low-volume sounds. Also, emergency calls should never cut audio when quiet. This option defaults to true (voice activity detection enabled).

Deprecated parameters

In older code and documentation, you may see a callback-based version of this function. This has been deprecated and its use is strongly discouraged. You should update any existing code to use the Promise-based version of createOffer() instead. The parameters for this form of createOffer() are described below, to aid in updating existing code.

An RTCSessionDescriptionCallback which will be passed a single RTCSessionDescription object describing the newly-created offer.
An RTCPeerConnectionErrorCallback which will be passed a single DOMError object explaining why the request to create an offer failed.
options Optional
An optional RTCOfferOptions dictionary providing options requested for the offer.

Return value

A Promise whose fulfillment handler will receive an RTCSessionDescription which contains the SDP describing the generated offer. That received offer should be delivered through the signaling server to a remote peer.


Here we see a handler for the negotiationneeded event which creates the offer and sends it to the remote system over a signaling channel.

Keep in mind that this is part of the signaling process, the transport layer for which is an implementation detail that's entirely up to you. In this case, a WebSocket connection is used to send a JSON message with a type field with the value "video-offer" to the other peer. The contents of the object being passed to the sendToServer() function, along with everything else in the promise fulfillment handler, depend entirely on your design.

  myPeerConnection.createOffer().then(function(offer) {
    return myPeerConnection.setLocalDescription(offer);
  .then(function() {
      name: myUsername,
      target: targetUsername,
      type: "video-offer",
      sdp: myPeerConnection.localDescription
  .catch(function(reason) {
    // An error occurred, so handle the failure to connect

In this code, the offer is created, and once successful, the local end of the RTCPeerConnection is configured to match by calling setLocalDescription(). Once that's done, the offer is sent to the remote system over the signaling channel; in this case, by using a custom function called sendToServer(). The implementation of the signaling server is independent from the WebRTC specification, so it doesn't matter how the offer is sent as long as both the caller and potential receiver are using the same one.

Use Promise.catch() to trap and handle errors.

See Signaling and video calling for the complete example from which this snippet is derived; this will help you to understand how the signaling code here works.


Specification Status Comment
WebRTC 1.0: Real-time Communication Between Browser
The definition of 'createOffer()' in that specification.
Working Draft Initial definition.

Browser compatibility

Feature Chrome Firefox (Gecko) Internet Explorer Opera Safari (WebKit)
Basic support (Yes) [1] 22 (22) [2] ? ? ?
Promise-based version 52.0 ? ? ? ?
Feature Android Android Webview Firefox Mobile (Gecko) Firefox OS IE Mobile Opera Mobile Safari Mobile Chrome for Android
Basic support No support No support [1] 24.0 (24) [2] ? [2] ? ? ? (Yes) [1]
Promise-based version No support 52.0 ? ? ? ? ? 52.0

[1] The callback-based version of this method was deprecated in Chrome 49.

[2] The callback-based version of this method was deprecated in Firefox 37.


© 2016 Mozilla Contributors
Licensed under the Creative Commons Attribution-ShareAlike License v2.5 or later.

API createOffer Experimental Method Reference RTCPeerConnection WebRTC