Webrtc learning transfer rate control [four]

I learned about RTCPeerConnection(null) before, so I won’t recall it anymore.

RTP Media

• Receiver
• Sender
• getReceivers();
• getSenders ();

Receiver and Sender properties

• MediaStreamTrack media track can get audio or video
• RTCDtlsTransport media data transmission properties

Sender's method

• getParameters Get RTCPtpParameter object
  Codec, h264, vk8
• getContributingSources will be used in general mixing

var local = new RTCPeerConnection(null);
var senders = local.getSenders(); //数组
var sender;
//遍历数组
senders.forEach(function(item){
    
    
	item.track.kind //区分音视频
    sender = item; //根据需求去获取sender
});

var parms = sender.getParameters();

if(parms.encodeings){
    
    
	parms.encodeings[0].maxBitrate = 2048*1000 //设置传输速率
}

await sender.setParameters(parms);

Generally, changing the transmission rate should be set by the sending end, and at the same time, it will be equipped with a network speed monitoring service to switch the rate flexibly;