无法限制WebRTC P2P多参与者接收带宽

Question

我正试图通过将此示例与我现有的多参与者视频呼叫代码相结合来更改WebRTC P2P视频呼叫的"动态带宽"：

示例：https://webrtc.github.io/samples/src/content/peerconnection/bandwidth/

当我通过Chrome、查看WebRTC内部时

发送(ssrc((视频(的bitsReceivedPerSecond已被丢弃到所选带宽。但是，recv(ssrc((视频(的bitsReceivedPerSecond仍然保持不变。我可以知道如何使带宽更改同时适用于发送和接收吗？

下面是我的代码，如果你能帮我指出错误就太好了，提前谢谢。

2018年12月14日更新：在代码中添加了接收器的第一个选项

问题：未捕获类型错误：receiver.getParameters不是函数

const bandwidthSelector = document.querySelector('select#bandwidth');
bandwidthSelector.disabled = false;
// renegotiate bandwidth on the fly.
bandwidthSelector.onchange = () => {
bandwidthSelector.disabled = true;
const bandwidth = bandwidthSelector.options[bandwidthSelector.selectedIndex].value;
// In Chrome, use RTCRtpSender.setParameters to change bandwidth without
// (local) renegotiation. Note that this will be within the envelope of
// the initial maximum bandwidth negotiated via SDP.
if ((adapter.browserDetails.browser === 'chrome' ||
(adapter.browserDetails.browser === 'firefox' &&
adapter.browserDetails.version >= 64)) &&
'RTCRtpSender' in window &&
'setParameters' in window.RTCRtpSender.prototype) {
$.each(peers, function( index, value ) {
const sender = value.getSenders()[0];
const parameters = sender.getParameters();
if (!parameters.encodings) {
parameters.encodings = [{}];
}
if (bandwidth === 'unlimited') {
delete parameters.encodings[0].maxBitrate;
} else {
parameters.encodings[0].maxBitrate = bandwidth * 1000;
}
sender.setParameters(parameters)
.then(() => {
bandwidthSelector.disabled = false;
})
.catch(e => console.error(e));
/* 1st Option - Start */
const receiver = value.getReceivers()[0];
const recParameters = receiver.getParameters();
if (!recParameters.encodings) {
recParameters.encodings = [{}];
}
if (bandwidth === 'unlimited') {
delete recParameters.encodings[0].maxBitrate;
} else {
recParameters.encodings[0].maxBitrate = bandwidth * 1000;
}
receiver.setParameters(recParameters)
.then(() => {
bandwidthSelector.disabled = false;
})
.catch(e => console.error(e));
/* 1st Option - End */
return;
});             
}
// Fallback to the SDP munging with local renegotiation way of limiting
// the bandwidth.
function onSetSessionDescriptionError(error) {
console.log('Failed to set session description: ' + error.toString());
}
};
function updateBandwidthRestriction(sdp, bandwidth) {
let modifier = 'AS';
if (adapter.browserDetails.browser === 'firefox') {
bandwidth = (bandwidth >>> 0) * 1000;
modifier = 'TIAS';
}
if (sdp.indexOf('b=' + modifier + ':') === -1) {
// insert b= after c= line.
sdp = sdp.replace(/c=IN (.*)rn/, 'c=IN $1rnb=' + modifier + ':' + bandwidth + 'rn');
} else {
sdp = sdp.replace(new RegExp('b=' + modifier + ':.*rn'), 'b=' + modifier + ':' + bandwidth + 'rn');
}
return sdp;
}
function removeBandwidthRestriction(sdp) {
return sdp.replace(/b=AS:.*rn/, '').replace(/b=TIAS:.*rn/, '');
}

2018年12月14日更新：第二个选项createOffer

问题：未能设置会话描述：InvalidStateError:未能在"RTCPeerConnection"上执行"setRemoteDescription"：未能设置远程应答sdp:在错误状态下调用：kStable

const bandwidthSelector = document.querySelector('select#bandwidth');
bandwidthSelector.disabled = false;
// renegotiate bandwidth on the fly.
bandwidthSelector.onchange = () => {
bandwidthSelector.disabled = true;
const bandwidth = bandwidthSelector.options[bandwidthSelector.selectedIndex].value;
// In Chrome, use RTCRtpSender.setParameters to change bandwidth without
// (local) renegotiation. Note that this will be within the envelope of
// the initial maximum bandwidth negotiated via SDP.
if ((adapter.browserDetails.browser === 'chrome' ||
(adapter.browserDetails.browser === 'firefox' &&
adapter.browserDetails.version >= 64)) &&
'RTCRtpSender' in window &&
'setParameters' in window.RTCRtpSender.prototype) {
$.each(peers, function( index, value ) {
const sender = value.getSenders()[0];
const parameters = sender.getParameters();
if (!parameters.encodings) {
parameters.encodings = [{}];
}
if (bandwidth === 'unlimited') {
delete parameters.encodings[0].maxBitrate;
} else {
parameters.encodings[0].maxBitrate = bandwidth * 1000;
}
sender.setParameters(parameters)
.then(() => {
bandwidthSelector.disabled = false;
})
.catch(e => console.error(e));
/* 2nd option - Start */
value.createOffer(
function (local_description) {
console.log("Local offer description is: ", local_description);
value.setLocalDescription(local_description,
function () {
signaling_socket.emit('relaySessionDescription', {
'peer_id': index,
'session_description': local_description
});
console.log("Offer setLocalDescription succeeded");
},
function () {
Alert("Offer setLocalDescription failed!");
}
);
},
function (error) {
console.log("Error sending offer: ", error);
}).then(() => {
const desc = {
type: value.remoteDescription.type,
sdp: bandwidth === 'unlimited'
? removeBandwidthRestriction(value.remoteDescription.sdp)
: updateBandwidthRestriction(value.remoteDescription.sdp, bandwidth)
};
console.log('Applying bandwidth restriction to setRemoteDescription:n' +
desc.sdp);
return value.setRemoteDescription(desc);
})
.then(() => {
bandwidthSelector.disabled = false;
})
.catch(onSetSessionDescriptionError);
/* 2nd option - End */
return;
});             
}
// Fallback to the SDP munging with local renegotiation way of limiting
// the bandwidth.
function onSetSessionDescriptionError(error) {
console.log('Failed to set session description: ' + error.toString());
}
};
function updateBandwidthRestriction(sdp, bandwidth) {
let modifier = 'AS';
if (adapter.browserDetails.browser === 'firefox') {
bandwidth = (bandwidth >>> 0) * 1000;
modifier = 'TIAS';
}
if (sdp.indexOf('b=' + modifier + ':') === -1) {
// insert b= after c= line.
sdp = sdp.replace(/c=IN (.*)rn/, 'c=IN $1rnb=' + modifier + ':' + bandwidth + 'rn');
} else {
sdp = sdp.replace(new RegExp('b=' + modifier + ':.*rn'), 'b=' + modifier + ':' + bandwidth + 'rn');
}
return sdp;
}
function removeBandwidthRestriction(sdp) {
return sdp.replace(/b=AS:.*rn/, '').replace(/b=TIAS:.*rn/, '');
}

Answer 1

RTCRtpSender仅控制发送带宽。如果要限制接收带宽，则需要使用b=AS/b=TIAS方式或使接收器使用setParameters。