在此页

• 句法
• 范例
• 规范
• 浏览器兼容性
• 另请参阅
• 相关话题

这是 实验性技术
检查 浏览器兼容性表格 要小心谨慎在生产中使用这之前。

只读 sctp property on the RTCPeerConnection interface returns an RTCSctpTransport describing the SCTP transport over which SCTP data is being sent and received. If SCTP hasn't been negotiated, this value is null .

The SCTP transport is used for transmitting and receiving data for any and all RTCDataChannel s on the peer connection.

句法

var sctp = RTCPeerConnection.sctp;
					

值

A RTCSctpTransport object describing the SCTP transport being used by the RTCPeerConnection for transmitting and receiving on its data channels, or null if SCTP negotiation hasn't happened.

范例

var pc = new RTCPeerConnection();
var channel = pc.createDataChannel("Mydata");
channel.onopen = function(event) {
  channel.send('sending a message');
}
channel.onmessage = function(event) {
  console.log(event.data);
}
// Determine the largest message size that can be sent
var sctp = pc.sctp;
var maxMessageSize = sctp.maxMessageSize;
					

规范

规范	状态	注释
WebRTC 1.0: Real-time Communication Between Browsers The definition of 'RTCPeerConnection.sctp' in that specification.	候选推荐	最初的规范。

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
sctp	Chrome 76	Edge 79	Firefox No 不支持 No 见 bug 1278299 .	IE No	Opera No	Safari No	WebView Android 76	Chrome Android 76	Firefox Android No 不支持 No 见 bug 1278299 .	Opera Android 54	Safari iOS No	Samsung Internet Android 12.0

图例

完整支持

完整支持

不支持

不支持

见实现注意事项。

另请参阅

• WebRTC
• RTCPeerConnection
• RTCDataChannel
• SCTP

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