How to use sdpTransform in Peer.js? - webrtc

Question

I am developing a voice call app for android using peer.js. It is working but there is a lag in audio (around 1-5 seconds). So I was searching for a fix and found out that I can enable DTX Audio to reduce the number of packets sent. But I have no idea How to access and change SDP object in peer.js. Here is my code,
let audioConstraints = {
channelCount: 1,
sampleRate: 16000,
sampleSize: 8,
volume: 1,
latency: 0.003,
echoCancellation: true,
noiseSuppression: true,
autoGainControl: true,
};
constraints = { audio: audioConstraints, video: false };
let localStream;
//listen for calls
function listen() {
peer.on("call", (call) => {
navigator.mediaDevices
.getUserMedia(constraints)
.then(function (stream) {
localVideo.srcObject = stream;
localStream = stream;
call.answer(stream);
call.on("stream", (remoteStream) => {
Android.onPeerStream();
remoteVideo.srcObject = remoteStream;
});
})
.catch(function (err) {
Android.logEvent("getUserMedia error: " + err);
});
});
}
//start call
function startCall(otherUserId) {
navigator.mediaDevices
.getUserMedia(constraints)
.then(function (stream) {
localVideo.srcObject = stream;
localStream = stream;
const call = peer.call(otherUserId, stream);
call.on("stream", (remoteStream) => {
Android.onPeerStream();
remoteVideo.srcObject = remoteStream;
});
})
.catch(function (err) {
Android.logEvent("getUserMedia error: " + err);
});
}
could someone please share a working example of how to access and modify SDP in peer.js? Also, how can I tune the audioConstraints to achieve low network lag while keeping decent voice quality?
UPDATE
Found the way to access SDP in peer.js but still can't figure it out how to enable DTX audio.