RTCPeerConnection: createAnswer() method
The createAnswer()
method on the
RTCPeerConnection
interface creates an SDP answer to an
offer received from a remote peer during the offer/answer negotiation of a WebRTC
connection. The answer contains information about any media already attached to the
session, codecs and options supported by the browser, and any ICE
candidates already gathered. The answer is delivered to the returned
Promise
, and should then be sent to the source of the offer to continue
the negotiation process.
Syntax
js
createAnswer()
createAnswer(options)
createAnswer(successCallback, failureCallback) // deprecated
createAnswer(successCallback, failureCallback, options) // deprecated
Parameters
options
Optional-
An optional object providing options requested for the answer. Currently, there are no available options.
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
createAnswer()
instead. The parameters for the older form of
createAnswer()
are described below, to aid in updating existing code.
successCallback
Deprecated-
A callback function which will be passed a single
RTCSessionDescription
object describing the newly-created answer. failureCallback
Deprecated-
A callback function which will be passed a single
DOMException
object explaining why the request to create an answer failed. options
Optional-
An optional object providing options requested for the answer.
Exceptions
NotReadableError
-
The identity provider wasn't able to provide an identity assertion.
OperationError
-
Generation of the SDP failed for some reason; this is a general failure catch-all exception.
Return value
A Promise
whose fulfillment handler is called with an object conforming
to the RTCSessionDescriptionInit
dictionary which contains the SDP answer
to be delivered to the other peer.
Examples
Here is a segment of code taken from the code that goes with the article Signaling and video calling. This code comes from the handler for the message sent to carry an offer to another peer across the signaling channel.
Note: 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-answer" to the other peer, carrying the answer to the device which sent the
offer to connect. 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
js
pc.createAnswer()
.then((answer) => pc.setLocalDescription(answer))
.then(() => {
// Send the answer to the remote peer through the signaling server.
})
.catch(handleGetUserMediaError);
This asks RTCPeerConnection
to create and return a new answer. In our
promise handler, the returned answer is set as the description of the local end of the
connection by calling setLocalDescription()
.
Once that succeeds, the answer is sent to the signaling server using whatever protocol you see fit.
Promise.catch()
is used to trap and handle errors.
See Handling the invitation in our WebRTC chat example to view the complete code from which this snippet is derived; that will help you understand the signaling process and how answers work.
Specifications
Specification |
---|
WebRTC: Real-Time Communication in Browsers # dom-rtcpeerconnection-createanswer |
Browser compatibility
BCD tables only load in the browser