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10 Commits
4.01 ... 4.02

41
app.js
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@ -67,8 +67,8 @@ const peers = io.of('/');
const createWorker = async () => {
try {
worker = await mediasoup.createWorker({
rtcMinPort: process.env.RTC_MIN_PORT,
rtcMaxPort: process.env.RTC_MAX_PORT,
rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
})
console.log(`[createWorker] worker pid ${worker.pid}`);
@ -109,7 +109,7 @@ const mediaCodecs = [
const closeCall = (callId) => {
try {
if (videoCalls[callId]) {
if (callId && videoCalls[callId]) {
videoCalls[callId].producer?.close();
videoCalls[callId].consumer?.close();
videoCalls[callId]?.consumerTransport?.close();
@ -124,16 +124,6 @@ const closeCall = (callId) => {
}
}
const getRtpCapabilities = (callId, callback) => {
try {
console.log('[getRtpCapabilities] callId', callId);
const rtpCapabilities = videoCalls[callId].router.rtpCapabilities;
callback({ rtpCapabilities });
} catch (error) {
console.log(`ERROR | getRtpCapabilities | callId ${callId} | ${error.message}`);
}
}
/*
- Handlers for WS events
- These are created only when we have a connection with a peer
@ -160,7 +150,9 @@ peers.on('connection', async socket => {
- If the room already exists, it will not create it, but will only return rtpCapabilities
*/
socket.on('createRoom', async ({ callId }, callback) => {
let callbackResponse = null;
try {
// We can continue with the room creation process only if we have a callId
if (callId) {
console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
if (!videoCalls[callId]) {
@ -169,12 +161,19 @@ peers.on('connection', async socket => {
console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
}
socketDetails[socket.id] = callId;
getRtpCapabilities(callId, callback);
// rtpCapabilities is set for callback
console.log('[getRtpCapabilities] callId', callId);
callbackResponse = {
rtpCapabilities :videoCalls[callId].router.rtpCapabilities
};
} else {
console.log(`[createRoom] missing callId ${callId}`);
}
} catch (error) {
console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
} finally {
callback(callbackResponse);
}
});
@ -195,16 +194,19 @@ peers.on('connection', async socket => {
videoCalls[callId].producerTransport = await createWebRtcTransportLayer(callId, callback);
} else {
console.log(`producerTransport has already been defined | callId ${callId}`);
callback(null);
}
} else if (!sender) {
if (!videoCalls[callId].consumerTransport) {
videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
} else {
console.log(`consumerTransport has already been defined | callId ${callId}`);
callback(null);
}
}
} catch (error) {
console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
callback(error);
}
});
@ -248,9 +250,9 @@ peers.on('connection', async socket => {
});
// Send back to the client the Producer's id
// callback({
// id: videoCalls[callId].producer.id
// });
callback && callback({
id: videoCalls[callId].producer.id
});
} catch (error) {
console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
}
@ -299,14 +301,14 @@ peers.on('connection', async socket => {
videoCalls[callId].consumer.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport close from consumer', callId);
closeCall();
closeCall(callId);
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
videoCalls[callId].consumer.on('producerclose', () => {
const callId = socketDetails[socket.id];
console.log('producer of consumer closed', callId);
closeCall();
closeCall(callId);
});
// From the consumer extract the following params to send back to the Client
@ -321,6 +323,7 @@ peers.on('connection', async socket => {
callback({ params });
} else {
console.log(`[canConsume] Can't consume | callId ${callId}`);
callback(null);
}
} catch (error) {
console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)