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• 句法
• 范例
• 规范
• 浏览器兼容性
• 相关话题

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.

句法

aPromise = RTCPeerConnection.createAnswer([options]);
RTCPeerConnection.createAnswer(successCallback, failureCallback[, options]);  
					

参数

选项 可选

An object which contains options which customize the answer; this is based on the RTCAnswerOptions 字典。

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 this form of createAnswer() are described below, to aid in updating existing code.

successCallback

RTCSessionDescriptionCallback which will be passed a single RTCSessionDescription object describing the newly-created answer.

failureCallback

RTCPeerConnectionErrorCallback which will be passed a single DOMException object explaining why the request to create an answer failed.

选项 可选

An optional RTCOfferOptions object providing options requested for the answer.

异常

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.

返回值

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.

范例

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.

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

pc.createAnswer().then(function(answer) {
  return pc.setLocalDescription(answer);
})
.then(function() {
  // 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.

见 Handling the invitation in Signaling and video calling to see the complete code, in context, from which this snippet is derived; that will help you understand the signaling process and how answers work.

规范

规范	状态	注释
WebRTC 1.0: Real-time Communication Between Browsers The definition of 'createAnswer()' in that specification.	候选推荐	初始定义。

浏览器兼容性

The compatibility table on this page is generated from structured data. If you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. 更新 GitHub 上的兼容性数据

	桌面						移动
	Chrome	Edge	Firefox	Internet Explorer	Opera	Safari	Android webview	Chrome for Android	Firefox for Android	Opera for Android	Safari on iOS	Samsung Internet
createAnswer	Chrome 51 51 Promise-based version. 24	Edge 15	Firefox 22	IE No	Opera 43 43 Promise-based version. 不支持 37 — 43	Safari 11	WebView Android 51 51 Promise-based version. Yes	Chrome Android 51 51 Promise-based version. Yes	Firefox Android 44	Opera Android 43 43 Promise-based version. 不支持 37 — 43	Safari iOS Yes	Samsung Internet Android 6.0 6.0 Promise-based version and unprefixed. 不支持 5.0 — 6.0 Promise-based version.

图例

完整支持

完整支持

不支持

不支持

见实现注意事项。

元数据

• 最后修改： Mar 18, 2019

相关话题

1. WebRTC API
2. RTCPeerConnection
3. 构造函数

   1. RTCPeerConnection()
4. 特性

   1. canTrickleIceCandidates
   2. connectionState
   3. currentLocalDescription
   4. currentRemoteDescription
   5. getDefaultIceServers()
   6. iceConnectionState
   7. iceGatheringState
   8. localDescription
   9. onaddstream
   10. onconnectionstatechange
   11. ondatachannel
   12. onicecandidate
   13. oniceconnectionstatechange
   14. onicegatheringstatechange
   15. onidentityresult
   16. onidpassertionerror
   17. onidpvalidationerror
   18. onnegotiationneeded
   19. onpeeridentity
   20. onremovestream
   21. onsignalingstatechange
   22. ontrack
   23. peerIdentity
   24. pendingLocalDescription
   25. pendingRemoteDescription
   26. remoteDescription
   27. sctp
   28. signalingState
5. 方法

   1. addIceCandidate()
   2. addStream()
   3. addTrack()
   4. close()
   5. createAnswer()
   6. createDataChannel()
   7. createOffer()
   8. generateCertificate()
   9. getConfiguration()
   10. getIdentityAssertion()
   11. getReceivers()
   12. getSenders()
   13. getStats()
   14. getStreamById()
   15. getTransceivers()
   16. removeStream()
   17. removeTrack()
   18. restartIce()
   19. setConfiguration()
   20. setIdentityProvider()
   21. setLocalDescription()
   22. setRemoteDescription()
6. 事件

   1. addstream
   2. identityresult
   3. negotiationneeded
   4. removestream
   5. signalingstatechange
7. 继承：

   1. EventTarget
8. Related pages for WebRTC

   1. MediaDevices.getUserMedia()
   2. Navigator.mediaDevices
   3. RTCCertificate
   4. RTCDTMFSender
   5. RTCDTMFToneChangeEvent
   6. RTCDataChannel
   7. RTCDataChannelEvent
   8. RTCDtlsTransport
   9. RTCErrorEvent
   10. RTCIceCandidate
   11. RTCIceTransport
   12. RTCPeerConnectionIceErrorEvent
   13. RTCPeerConnectionIceEvent
   14. RTCRtpReceiver
   15. RTCRtpSender
   16. RTCRtpTransceiver
   17. RTCSctpTransport
   18. RTCSessionDescription
   19. RTCStatsEvent
   20. RTCStatsReport
   21. RTCTrackEvent