在此页

• 句法
• 范例
• 规范
• 浏览器兼容性
• 另请参阅
• 相关话题

RTCPeerConnection 方法 getSenders() returns an array of RTCRtpSender objects, each of which represents the RTP sender responsible for transmitting one track's data. A sender object provides methods and properties for examining and controlling the encoding and transmission of the track's data.

句法

var senders = rtcPeerConnection.getSenders();
					

返回值

An array of RTCRtpSender objects, one for each track on the connection. The array is empty if there are no RTP senders on the connection.

The order of the returned RTCRtpSender s is not defined by the specification, and may change from one call to getSenders() to the next.

范例

In this example, a setMuting() function is shown. This function takes as input an RTCPeerConnection , pc , and a Boolean, muting . The function gets the list of the peer connection's senders and iterates over every sender, setting the corresponding media track's enabled to the inverse of the specified muting .

function setMuting(pc, muting) {
  let senderList = pc.getSenders();
  senderList.forEach(sender) {
    sender.track.enabled = !muting;
  }
}
					

规范

规范	状态	注释
WebRTC 1.0: Real-time Communication Between Browsers The definition of 'RTCPeerConnection.getSenders()' 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
getSenders	Chrome 64	Edge ≤18	Firefox 22	IE No	Opera 51	Safari 11	WebView Android 64	Chrome Android 64	Firefox Android 44	Opera Android 47	Safari iOS Yes	Samsung Internet Android 6.0

图例

完整支持

完整支持

不支持

不支持

另请参阅

• WebRTC
• RTCRtpSender

元数据

• 最后修改： Jul 2, 2020

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