Comments (8)
@Blancduman Thank you for your time. π
I solved it by changing announcedIp to my public IP on the server-side.
listenIps: [
{
ip: '0.0.0.0',
announcedIp: 'public IP',
}
],
from mediasoup-client-flutter.
from mediasoup-client-flutter.
from mediasoup-client-flutter.
Here is my flutter code.
import 'dart:async';
import 'dart:io';
import 'package:flutter/foundation.dart';
import 'package:flutter/material.dart';
import 'package:flutter_webrtc/flutter_webrtc.dart';
import 'package:mediasoup_client_flutter/mediasoup_client_flutter.dart';
import 'package:permission_handler/permission_handler.dart';
import 'package:video_call_mediasoup/socket.dart';
void main() {
HttpOverrides.global = MyHttpOverrides();
runApp(MyApp());
}
class MyHttpOverrides extends HttpOverrides{
@override
HttpClient createHttpClient(SecurityContext? context){
return super.createHttpClient(context)
..badCertificateCallback = (X509Certificate cert, String host, int port)=> true;
}
}
class MyApp extends StatelessWidget {
// This widget is the root of your application.
@override
Widget build(BuildContext context) {
return MaterialApp(
title: 'Flutter Demo',
theme: ThemeData(
primarySwatch: Colors.blue,
),
home: MyHomePage(title: 'Flutter Demo Home Page'),
);
}
}
class MyHomePage extends StatefulWidget {
MyHomePage({Key? key, required this.title}) : super(key: key);
final String title;
@override
_MyHomePageState createState() => _MyHomePageState();
}
class _MyHomePageState extends State<MyHomePage> {
RTCVideoRenderer _localRenderer = new RTCVideoRenderer();
RTCVideoRenderer _remoteRenderer = new RTCVideoRenderer();
late MediaStream _localStream;
late Transport _producerTransport;
late Transport _consumerTransport;
late Device device;
@override
void initState() {
SocketUtils.connect();
_localRenderer.initialize();
_remoteRenderer.initialize();
super.initState();
}
@override
deactivate() {
_localRenderer.dispose();
_remoteRenderer.dispose();
super.deactivate();
}
@override
dispose() {
_localRenderer.dispose();
_remoteRenderer.dispose();
super.dispose();
}
getUserMedia() async {
}
connectAndProduce() async {
var isHasPermission = true;
if (!kIsWeb) {
var status = await Permission.camera.request();
isHasPermission = status.isGranted;
}
if (isHasPermission) {
final Map<String, dynamic> constraints = {
'audio': false,
'video': true
};
_localStream = await navigator.mediaDevices.getUserMedia(constraints);
this.setState(() {
_localRenderer.srcObject = _localStream;
});
} else {
print('PERMISSION NOT ALLOWED');
}
_producerTransport.produce(
track: _localStream.getVideoTracks().first,
stream: _localStream,
source: 'webcam',
codecOptions: ProducerCodecOptions(
videoGoogleStartBitrate: 1000,
),
encodings: [
// RtpEncodingParameters(rid: 'r0', scalabilityMode: 'S1T3', maxBitrate: 100000),
// RtpEncodingParameters(rid: 'r1', scalabilityMode: 'S1T3', maxBitrate: 300000),
// RtpEncodingParameters(rid: 'r2', scalabilityMode: 'S1T3', maxBitrate: 900000)
],
);
_producerTransport.on('trackended', () {
print('Video track ended');
});
}
connectAndReceive() async {
SocketUtils.connection.emitWithAck('consume', {
'rtpCapabilities': device.rtpCapabilities.toMap(),
}, ack: (response) async {
_consumerTransport.consume(
id: response['id'],
producerId: response['producerId'],
peerId: response['producerId'],
kind: RTCRtpMediaTypeExtension.fromString(response['kind']),
rtpParameters: RtpParameters.fromMap(response['rtpParameters']),
accept: (param) {
}
);
});
}
_startCall() async {
await getUserMedia();
await createDevice();
await createWebRtcTransport();
await connectAndProduce();
}
_answerCall() async {
await createRecvTransport();
await connectAndReceive();
}
createDevice () async {
// server side will create router and return RtpCapabilities then we need to load it to device
device = Device();
var routerRtpCapabilities = await getRtpCapabilities();
print(routerRtpCapabilities);
await device.load(routerRtpCapabilities: routerRtpCapabilities);
}
createWebRtcTransport() async {
var webRtcTransport = await getWebRtcTransport(true);
_producerTransport = device.createSendTransportFromMap(webRtcTransport);
_producerTransport.on('connect', (Map data) {
var callback = data['callback'];
var errback = data['errback'];
try {
SocketUtils.connection.emit('transport-connect', {
'transportId': _producerTransport.id,
'dtlsParameters': data['dtlsParameters'].toMap()
});
callback();
} catch(error) {
errback(error);
}
});
_producerTransport.on('produce', (Map data) {
var callback = data['callback'];
var errback = data['errback'];
try {
var rtpParameters = data['rtpParameters'];
var kind = data['kind'];
var appData = data['appData'];
SocketUtils.connection.emitWithAck('transport-produce', {
'transportId': _producerTransport.id,
'kind': kind,
'rtpParameters': rtpParameters.toMap(),
'appData': appData
}, ack: (response) {
callback(response['id']);
});
} catch (error) {
errback(error);
}
});
}
createRecvTransport() async {
var webRtcTransport = await getWebRtcTransport(false);
_consumerTransport = device.createRecvTransportFromMap(
webRtcTransport,
consumerCallback: (Consumer consumer, [dynamic accept]) {
this.setState(() {
_remoteRenderer.srcObject = consumer.stream;
});
accept({});
SocketUtils.connection.emit('consumer-resume');
},
dataConsumerCallback: (data) {
}
);
_consumerTransport.on('connect', (Map data) {
var callback = data['callback'];
var errback = data['errback'];
try {
SocketUtils.connection.emit('transport-recv-connect', {
'transportId': _consumerTransport.id,
'dtlsParameters': data['dtlsParameters'].toMap()
});
callback();
} catch(error) {
print(error);
errback(error);
}
});
}
getRtpCapabilities() {
Completer completer = new Completer();
SocketUtils.connection.emitWithAck('getRtpCapabilities', null, ack: (response) {
var rtpCapabilities = RtpCapabilities.fromMap(response['rtpCapabilities']);
completer.complete(rtpCapabilities);
});
return completer.future;
}
getWebRtcTransport(bool sender) async {
Completer completer = new Completer();
SocketUtils.connection.emitWithAck('createWebRtcTransport', { 'sender': sender }, ack: (response) {
completer.complete(response);
});
return completer.future;
}
SizedBox videoRenderers() => SizedBox(
height: 200,
child: Row(
children: [
Flexible(
child: Container(
key: Key('local'),
margin: EdgeInsets.fromLTRB(5.0, 5.0, 5.0, 5.0),
decoration: BoxDecoration(color: Colors.grey),
child: RTCVideoView(_localRenderer),
)
),
Flexible(
child: Container(
key: Key('remote'),
margin: EdgeInsets.fromLTRB(5.0, 5.0, 5.0, 5.0),
decoration: BoxDecoration(color: Colors.grey),
child: RTCVideoView(_remoteRenderer),
)
),
],
),
);
@override
Widget build(BuildContext context) {
return Scaffold(
appBar: AppBar(
title: Text(widget.title),
),
body: Container(
child: Column(
children: [
videoRenderers(),
ElevatedButton(onPressed: _startCall, child: Text('Call')),
ElevatedButton(onPressed: _answerCall, child: Text('Answer'))
],),
),
);
}
}
from mediasoup-client-flutter.
And this is serverside code:
import express from 'express'
const app = express()
import https from 'httpolyglot'
import fs from 'fs'
import path from 'path'
const __dirname = path.resolve()
import { Server } from 'socket.io'
import mediasoup from 'mediasoup'
app.get('/', (req, res) => {
res.send('Hello from mediasoup app!')
})
app.use('/sfu', express.static(path.join(__dirname, 'public')))
// SSL cert for HTTPS access
const options = {
key: fs.readFileSync('./server/ssl/key.key', 'utf-8'),
cert: fs.readFileSync('./server/ssl/cert.crt', 'utf-8')
}
const httpsServer = https.createServer(options, app)
httpsServer.listen(3000, () => {
console.log('listening on port: ' + 3000)
})
const io = new Server(httpsServer)
// socket.io namespace (could represent a room?)
const peers = io.of('/mediasoup')
let worker
let router
let producerTransport
let consumerTransport
let producer
let consumer
const createWorker = async () => {
worker = await mediasoup.createWorker({
rtcMinPort: 2000,
rtcMaxPort: 2020,
})
console.log(`worker pid ${worker.pid}`)
worker.on('died', error => {
// This implies something serious happened, so kill the application
console.error('mediasoup worker has died')
setTimeout(() => process.exit(1), 2000) // exit in 2 seconds
})
return worker
}
// We create a Worker as soon as our application starts
worker = createWorker()
// This is an Array of RtpCapabilities
// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
// list of media codecs supported by mediasoup ...
// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
const mediaCodecs = [
{
kind: 'audio',
mimeType: 'audio/opus',
clockRate: 48000,
channels: 2,
},
{
kind: 'video',
mimeType: 'video/vp8',
clockRate: 90000,
parameters: {
'x-google-start-bitrate': 1000,
},
},
]
peers.on('connection', async socket => {
console.log(socket.id)
socket.emit('connection-success', {
socketId: socket.id
})
socket.on('disconnect', () => {
// do some cleanup
console.log('peer disconnected')
})
// worker.createRouter(options)
// options = { mediaCodecs, appData }
// mediaCodecs -> defined above
// appData -> custom application data - we are not supplying any
// none of the two are required
router = await worker.createRouter({ mediaCodecs, })
// Client emits a request for RTP Capabilities
// This event responds to the request
socket.on('getRtpCapabilities', (callback) => {
const rtpCapabilities = router.rtpCapabilities
console.log('rtp Capabilities', rtpCapabilities)
// call callback from the client and send back the rtpCapabilities
callback({ rtpCapabilities })
})
// Client emits a request to create server side Transport
// We need to differentiate between the producer and consumer transports
socket.on('createWebRtcTransport', async ({ sender }, callback) => {
console.log(`Is this a sender request? ${sender}`)
// The client indicates if it is a producer or a consumer
// if sender is true, indicates a producer else a consumer
if (sender)
producerTransport = await createWebRtcTransport(callback)
else
consumerTransport = await createWebRtcTransport(callback)
})
// see client's socket.emit('transport-connect', ...)
socket.on('transport-connect', async ({ dtlsParameters }) => {
console.log('DTLS PARAMS... ', { dtlsParameters })
await producerTransport.connect({ dtlsParameters })
})
// see client's socket.emit('transport-produce', ...)
socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
// call produce based on the prameters from the client
producer = await producerTransport.produce({
kind,
rtpParameters,
})
console.log('Producer ID: ', producer.id, producer.kind)
producer.on('transportclose', () => {
console.log('transport for this producer closed ')
producer.close()
})
// Send back to the client the Producer's id
callback({
id: producer.id
})
})
// see client's socket.emit('transport-recv-connect', ...)
socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
console.log(`DTLS PARAMS: ${dtlsParameters}`)
await consumerTransport.connect({ dtlsParameters })
})
socket.on('consume', async ({ rtpCapabilities }, callback) => {
console.log('CONS');
try {
// check if the router can consume the specified producer
if (router.canConsume({
producerId: producer.id,
rtpCapabilities
})) {
// transport can now consume and return a consumer
consumer = await consumerTransport.consume({
producerId: producer.id,
rtpCapabilities,
paused: true,
})
consumer.on('transportclose', () => {
console.log('transport close from consumer')
})
consumer.on('producerclose', () => {
console.log('producer of consumer closed')
})
// from the consumer extract the following params
// to send back to the Client
const params = {
id: consumer.id,
producerId: producer.id,
kind: consumer.kind,
rtpParameters: consumer.rtpParameters,
}
// send the parameters to the client
callback(params)
}
} catch (error) {
console.log(error.message)
callback({
params: {
error: error
}
})
}
})
socket.on('consumer-resume', async () => {
console.log('consumer resume')
await consumer.resume()
})
})
const createWebRtcTransport = async (callback) => {
try {
// https://mediasoup.org/documentation/v3/mediasoup/api/#WebRtcTransportOptions
const webRtcTransport_options = {
listenIps: [
{
ip: '0.0.0.0', // replace with relevant IP address
announcedIp: '127.0.0.1',
}
],
enableUdp: true,
enableTcp: true,
preferUdp: true,
}
// https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
let transport = await router.createWebRtcTransport(webRtcTransport_options)
console.log(`transport id: ${transport.id}`)
transport.on('dtlsstatechange', dtlsState => {
if (dtlsState === 'closed') {
transport.close()
}
})
transport.on('close', () => {
console.log('transport closed')
})
// send back to the client the following prameters
callback({
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
id: transport.id,
iceParameters: transport.iceParameters,
iceCandidates: transport.iceCandidates,
dtlsParameters: transport.dtlsParameters,
}
)
return transport
} catch (error) {
console.log(error)
callback( {
error: error
}
)
}
}
from mediasoup-client-flutter.
- try to debug your signaling's messages.
- check your transports seperatly
Let me know what you found out
from mediasoup-client-flutter.
Hi @un-hongly,
I am aslo new to webrtc and mediasoup. I have also followed the same tutorial on youtube as yours and I am also developing the app on flutter. I have used your above code, it is working fine for producer but camera is not rendering remote video. Can you help me to solve this issue?
from mediasoup-client-flutter.
Hi un-hongly...I have tried your code its working fine on wifi but not work with mobile hotspot and also not work on ec2 server where I have used nginx as a reverse proxy and allowed all the relevant ports....though I have changed announced ip to public ip....
from mediasoup-client-flutter.
Related Issues (20)
- Click to close the camera to report an error
- Voice calls on iOS are too low HOT 2
- The rear camera image is flipped 180 degrees HOT 2
- Remote video is blurry
- iOS audio always on speaker and can't switch to earpiece HOT 2
- how to broadcast one to many HOT 3
- load.device error
- Video images display problems
- Android phone sharing screen crash
- Transport.Produce(): type 'String' is not a subtype of type 'int' HOT 3
- transport.produce(): Null check operator used on a null value
- Why "maxRetransmits" is "required" in Transport.produceData()? HOT 2
- Transport connection state changed to Failed HOT 11
- Info regarding architecture and the mediasoup server compatability
- Send Transport "produce" listener not running on .produce() HOT 3
- update ..
- issue regarding not able to produce
- createSendTransportFromMap not connecting
- video stop after passing after sometime
- If producer transport close automatic consumer transport also closing
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from mediasoup-client-flutter.