I'm currently working on a webrtc project that was written before from another developer. While I'm currently new to webrtc and I tried hard to make it work after deleting obsolete functions and other things, now can I identify what is wrong here in my steps?
I'm following steps in here
my code here is about trigger .call button
$(dod)
.find(".call")
and I run throw signaling
wbsc.emit("SEND_EVENT_EMIT_CALL_AUDIO", {
data: { type: "login", id: id },
});
$(dod).hide();
//call*donecallProccess 1
setTimeout(() => {
wbsc.emit("SEND_EVENT_EMIT_CALL_AUDIO", {
data: { type: "doneoif", id: id },
});
}, 2e3);
that can trigger and process here
case "donecall":
call(data.id);
break;
case "showcall":
handleLogin(data.success, data.id);
break;
case "offercall":
handleOffer(data.offer, data.name);
break;
case "answercall":
handleAnswer(data.answer);
break;
case "candidatecall":
handleCandidate(data.candidate);
break;
case "leavecall":
handleLeave();
then this code run one after each handle login and getUserMediaSuccess gather permission from media stream and create a new RTCPeerConnection(servers) get tracks from my streams with addtrack if I implement it correctly and if ontrack happened can I collect streaming to add to remote peer like this way or should just add
yourConn.ontrack = (event) => {
if (event.candidate !== null) {
remoteVideo.srcObject = event.streams[0];
} else {
console.log("there is an error with on trackevent", event);
}
};
complete code for previous handlelogin and call is
let handleLogin = async (success) => {
try {
if (success) {
localVideo = document.getElementById("wbrtclocal");
remoteVideo = document.getElementById("wbrtcremote");
var getUserMedia = navigator.mediaDevices.getUserMedia|| navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia ;
if (getUserMedia) {
getUserMedia({
audio: { noiseSuppression: false },
})
.then(getUserMediaSuccess)
.catch(errorHandler);
} else {
alert("Your browser does not support getUserMedia API");
}
} else {
alert("Ooops...try a different username");
}
} catch (err) {
errorHandler(error);
}
};
let getUserMediaSuccess = async (streams) => {
try {
yourConn = new RTCPeerConnection(servers);
if(streams){
localStream = streams;
localVideo.srcObject = streams;
streams.getTracks().forEach((track) => yourConn.addTrack(track, streams));
}
yourConn.onicecandidate = (event) => {
if (event.candidate) {
wbsc.emit("SEND_EVENT_EMIT_CALL_AUDIO", {
data: {
type: "candidate",
candidate: event.candidate,
id: connectedUser,
},
});
}
console.log("nwe ice candidate is", event.candidate);
console.log("nwe stream is", streams);
};
remoteStream = new MediaStream();
remoteVideo.srcObject = remoteStream;
yourConn.ontrack = (event) => {
if (event.candidate !== null) {
event.streams[0].getTracks().forEach((track) => {
remoteStream.addTrack(track);
});
} else {
console.log("there is an error with on trackevent", event);
}
};
} catch (err) {
errorHandler(error);
}
console.log("stream is", streams);
};
call function
async function call(id) {
$("#videoCall").show();
if (id.length > 0) {
connectedUser = id;
try {
RTCPeerConnection.createOffer().
offer.
await yourConn
.createOffer()
.then((offer) => successCallback)
.catch((e) => {
fl(e);
});
let successCallback = async (offer) => {
try {
yourConn
.setLocalDescription(offer)
.then(
wbsc.emit("SEND_EVENT_EMIT_CALL_AUDIO", {
data: { type: "offer", offer: offer, id: connectedUser },
})
)
.catch((e) => {
fl(e);
});
} catch (e) {
fl(e);
}
};
} catch (e) {
fl(e);
}
const user = U_CASH[id];
if (user) {
$("#videoCall")
.find(".u-pic")
.css("background-image", "url(" + removegifpic(user.pic + ")"));
$("#videoCall").find(".u-topic").text(user.topic);
}
$(".statecall").text("جاري الإتصال");
hl($(".statecall"), "warning");
} else {
alert("username can't be blank!");
}
console.log("connectedUser", id);
console.log("offer", offer);
console.log();
console.log();
console.log();
}
and this about each one for those
[handleOffer , handleAnswer , handleCandidate , handleLeave]
let handleOffer = async (offer, name) => {
$("#callvideonot").show();
const user = U_CASH[name];
if (user) {
$("#callvideonot")
.find(".u-pic")
.css("background-image", "url(" + removegifpic(user.pic + ")"));
$("#callvideonot").find(".u-topic").text(user.topic);
}
$(".callvideoaccept").on("click", async () => {
connectedUser = name;
await yourConn.setRemoteDescription(offer);
yourConn
.createAnswer()
.then((answer) => t.setLocalDescription(answer))
.then(() => {
wbsc.emit("SEND_EVENT_EMIT_CALL_AUDIO", {
data: { type: "answer", answer: answer, id: connectedUser },
});
})
.catch(fl);
const user = U_CASH[name];
if (user) {
$(".statecall").text("متصل");
hl($(".statecall"), "success");
$("#videoCall")
.find(".u-pic")
.css("background-image", "url(" + removegifpic(user.pic + ")"));
$("#videoCall").find(".u-topic").text(user.topic);
}
$("#callvideonot").hide();
$("#videoCall").show();
});
$(".callvideodeny").on("click", function () {
wbsc.emit("SEND_EVENT_EMIT_CALL_AUDIO", {
data: { type: "leave", id: name },
});
});
};
let handleAnswer = async (answer) => {
try {
$(".statecall").text("متصل");
hl($(".statecall"), "success");
//here we delete new RTCSessionDescription because constructor is deprecated.
await yourConn.setRemoteDescription(answer);
} catch (e) {
fl(e);
}
};
let handleCandidate = async (candidate) => {
try {
var NewlyIceCandidate = new RTCIceCandidate(candidate)
.setRemoteDescription().
await yourConn.addIceCandidate(NewlyIceCandidate);
} catch (e) {
fl(e);
}
};
function handleLeave() {
$("#callvideonot").hide();
$(".statecall").text("رفض");
hl($(".statecall"), "danger");
$(".vloumemic").removeClass("fa-volume-off");
$(".vloumemic").addClass("fa-volume-up");
$(".mutemic").removeClass("fa-microphone-slash");
$(".mutemic").addClass("fa-microphone");
setTimeout(() => {
$("#videoCall").hide();
}, 1e3);
if (localStream) {
localStream.getTracks().forEach((e) => e.stop());
}
if (connectedUser) {
connectedUser = null;
}
remoteVideo.src = null;
if (yourConn) {
yourConn.close();
yourConn.onicecandidate = null;
yourConn.ontrack = null;
localStream = null;
}
}
in here here number 7 instruction they said should I add Wait for an incoming remote SDP description from the signaling service and set it using RTCPeerConnection.setRemoteDescription(). as the caller where can I add it? in handleCandidate function?
and in the callee side number 1 instruction the said that I should Create a new RTCPeerConnection instance with the appropriate ICE configuration.
can I reuse
yourConn = new RTCPeerConnection(servers);
or should instantiate a new one to prevent conflict in website server process as a not yourConn it's global value and in top level of this file and reuse it over all the connection, is peers 2 or the callee should have another new RTCPeerConnection?
and for remoteVideo.srcObject = remoteStream; the remoteStream value is global and I overwrite it. can I here add new media stream or just it's good to dealing with the present one which is remoteVideo element?
// remoteStream = new MediaStream();
remoteVideo.srcObject = remoteStream;
the issue that was here is to split the RTCPeerConnection object and just create new one for each peer local and remote that what i did and it working corectly after i remove addtrack completly and replace it with addtranceiver and gettranceiver u can find it here from previous issue's answer and i fix it here
and no need to overwrite or reset the srcObect value. The track transition on receiver side in the same MediaStream should be "seamless" RTCRtpSender.replaceTrack This allows you to seamlessly change which track is being sent without having to renegotiate at the expense of another offer/answer cycle