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  4. RTCPeerConnection.createOffer()

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createOffer() 方法在 RTCPeerConnection interface initiates the creation of an SDP offer for the purpose of starting a new WebRTC connection to a remote peer. The SDP offer includes information about any MediaStreamTrack s 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.

返回值为 Promise which, when the offer has been created, is resolved with a RTCSessionDescription object containing the newly-created offer.

句法 

aPromise = myPeerConnection.createOffer([options]);
myPeerConnection.createOffer(successCallback, failureCallback, [options])

参数

选项 可选
An RTCOfferOptions dictionary providing options requested for the offer.

RTCOfferOptions dictionary

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

iceRestart 可选
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. 默认为 false .
offerToReceiveAudio 可选 (Legacy)
A legacy Boolean option which used to control whether or not to offer to the remote peer the opportunity to try to send audio. If this value is false , the remote peer will not be offered to send audio data, even if the local side will be sending audio data. If this value is true , the remote peer will be offered to send audio data, even if the local side will not be sending audio data. The default behavior is to offer to receive audio only if the local side is sending audio, not otherwise.
To emulate this behavior in modern implementations, the presence of this member with a value false , will set the direction of all existing audio transceivers to exclude reception (i.e. set to either "sendonly" or "inactive" ).
In modern implementations, the presence of this member with a value true , will ensure there is at least one transceiver set to receive audio that has not been stopped, and if there isn't one, one will be created.
注意： You shouldn't use this legacy property. Instead, use RTCRtpTransceiver to control whether or not to accept incoming audio.
offerToReceiveVideo 可选 (Legacy)
A legacy Boolean option which used to control whether or not to offer to the remote peer the opportunity to try to send video. If this value is false , the remote peer will not be offered to send video data, even if the local side will be sending video data. If this value is true , the remote peer will be offered to send video data, even if the local side will not be sending video data. The default behavior is to offer to receive video only if the local side is sending video, not otherwise.
To emulate this behavior in modern implementations, the presence of this member with a value false , will set the direction of all existing video transceivers to exclude reception (i.e. set to either "sendonly" or "inactive" ).
In modern implementations, the presence of this member with a value true , will ensure there is at least one transceiver set to receive video that has not been stopped, and if there isn't one, one will be created.
注意： You shouldn't use this legacy property. Instead, use RTCRtpTransceiver to control whether or not to accept incoming video.
voiceActivityDetection 可选
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.

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

返回值

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

异常

These exceptions are returned by rejecting the returned promise. Your rejection handler should examine the received exception to determine which occurred.

InvalidStateError
RTCPeerConnection is closed.
NotReadableError
No certificate or set of certificates was provided for securing the connection, and createOffer() was unable to create a new one. Since all WebRTC connections are required to be secured, that results in an error.
OperationError

Examining the state of the system to determine resource availability in order to generate the offer failed for some reason.

范例

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

注意： 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() {
    sendToServer({
      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 passing the offer (which is represented using an object conforming to RTCSessionDescriptionInit ) into 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.

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

见 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.

规范

规范 状态 注释
WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'createOffer()' 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
createOffer Chrome 51 注意事项
51 注意事项
Promise-based version.
24
Edge 15 Firefox 22 IE 不支持 No Opera 43 注意事项
43 注意事项
Promise-based version.
37 — 43
Safari 11 WebView Android 51 注意事项
51 注意事项
Promise-based version.
Yes
Chrome Android 51 注意事项
51 注意事项
Promise-based version.
Yes
Firefox Android 44 Opera Android 43 注意事项
43 注意事项
Promise-based version.
37 — 43
Safari iOS Yes Samsung Internet Android 6.0 注意事项
6.0 注意事项
Promise-based version and unprefixed.
5.0 — 6.0 注意事项
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：
  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