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Autres articles (26)
-
Use, discuss, criticize
13 avril 2011, parTalk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
A discussion list is available for all exchanges between users. -
Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...) -
Les vidéos
21 avril 2011, parComme les documents de type "audio", Mediaspip affiche dans la mesure du possible les vidéos grâce à la balise html5 .
Un des inconvénients de cette balise est qu’elle n’est pas reconnue correctement par certains navigateurs (Internet Explorer pour ne pas le nommer) et que chaque navigateur ne gère en natif que certains formats de vidéos.
Son avantage principal quant à lui est de bénéficier de la prise en charge native de vidéos dans les navigateur et donc de se passer de l’utilisation de Flash et (...)
Sur d’autres sites (3233)
-
record mediasoup RTP stream using FFmpeg for Firefox
30 juillet 2024, par Hadi AghandehI am trying to record WebRTC stream using mediasoup. I could record successfully on chrome and safari 13/14/15. However on Firefox the does not work.


Client side code is a vue js component which gets rtp-compabilities using socket.io and create producers after the server creates the transports. This works good on chrome and safari.


const { connect , createLocalTracks } = require('twilio-video');
const SocketClient = require("socket.io-client");
const SocketPromise = require("socket.io-promise").default;
const MediasoupClient = require("mediasoup-client");

export default {
 data() {
 return {
 errors: [],
 isReady: false,
 isRecording: false,
 loading: false,
 sapio: {
 token: null,
 connectionId: 0
 },
 server: {
 host: 'https://rtc.test',
 ws: '/server',
 socket: null,
 },
 peer: {},
 }
 },
 mounted() {
 this.init();
 },
 methods: {
 async init() {
 await this.startCamera();

 if (this.takeId) {
 await this.recordBySapioServer();
 }
 },
 startCamera() {
 return new Promise( (resolve, reject) => {
 if (window.videoMediaStreamObject) {
 this.setVideoElementStream(window.videoMediaStreamObject);
 resolve();
 } else {
 // Get user media as required
 try {
 this.localeStream = navigator.mediaDevices.getUserMedia({
 audio: true,
 video: true,
 }).then((stream) => {
 this.setVideoElementStream(stream);
 resolve();
 })
 } catch (err) {
 console.error(err);
 reject();
 }
 }
 })
 },
 setVideoElementStream(stream) {
 this.localStream = stream;
 this.$refs.video.srcObject = stream;
 this.$refs.video.muted = true;
 this.$refs.video.play().then((video) => {
 this.isStreaming = true;
 this.height = this.$refs.video.videoHeight;
 this.width = this.$refs.video.videoWidth;
 });
 },
 // first thing we need is connecting to websocket
 connectToSocket() {
 const serverUrl = this.server.host;
 console.log("Connect with sapio rtc server:", serverUrl);

 const socket = SocketClient(serverUrl, {
 path: this.server.ws,
 transports: ["websocket"],
 });
 this.socket = socket;

 socket.on("connect", () => {
 console.log("WebSocket connected");
 // we ask for rtp-capabilities from server to send to us
 socket.emit('send-rtp-capabilities');
 });

 socket.on("error", (err) => {
 this.loading = true;
 console.error("WebSocket error:", err);
 });

 socket.on("router-rtp-capabilities", async (msg) => {
 const { routerRtpCapabilities, sessionId, externalId } = msg;
 console.log('[rtpCapabilities:%o]', routerRtpCapabilities);
 this.routerRtpCapabilities = routerRtpCapabilities;

 try {
 const device = new MediasoupClient.Device();
 // Load the mediasoup device with the router rtp capabilities gotten from the server
 await device.load({ routerRtpCapabilities });

 this.peer.sessionId = sessionId;
 this.peer.externalId = externalId;
 this.peer.device = device;

 this.createTransport();
 } catch (error) {
 console.error('failed to init device [error:%o]', error);
 socket.disconnect();
 }
 });

 socket.on("create-transport", async (msg) => {
 console.log('handleCreateTransportRequest() [data:%o]', msg);

 try {
 // Create the local mediasoup send transport
 this.peer.sendTransport = await this.peer.device.createSendTransport(msg);
 console.log('send transport created [id:%s]', this.peer.sendTransport.id);

 // Set the transport listeners and get the users media stream
 this.handleSendTransportListeners();
 this.setTracks();
 this.loading = false;
 } catch (error) {
 console.error('failed to create transport [error:%o]', error);
 socket.disconnect();
 }
 });

 socket.on("connect-transport", async (msg) => {
 console.log('handleTransportConnectRequest()');
 try {
 const action = this.connectTransport;

 if (!action) {
 throw new Error('transport-connect action was not found');
 }

 await action(msg);
 } catch (error) {
 console.error('ailed [error:%o]', error);
 }
 });

 socket.on("produce", async (msg) => {
 console.log('handleProduceRequest()');
 try {
 if (!this.produce) {
 throw new Error('produce action was not found');
 }
 await this.produce(msg);
 } catch (error) {
 console.error('failed [error:%o]', error);
 }
 });

 socket.on("recording", async (msg) => {
 this.isRecording = true;
 });

 socket.on("recording-error", async (msg) => {
 this.isRecording = false;
 console.error(msg);
 });

 socket.on("recording-closed", async (msg) => {
 this.isRecording = false;
 console.warn(msg)
 });

 },
 createTransport() {
 console.log('createTransport()');

 if (!this.peer || !this.peer.device.loaded) {
 throw new Error('Peer or device is not initialized');
 }

 // First we must create the mediasoup transport on the server side
 this.socket.emit('create-transport',{
 sessionId: this.peer.sessionId
 });
 },
 handleSendTransportListeners() {
 this.peer.sendTransport.on('connect', this.handleTransportConnectEvent);
 this.peer.sendTransport.on('produce', this.handleTransportProduceEvent);
 this.peer.sendTransport.on('connectionstatechange', connectionState => {
 console.log('send transport connection state change [state:%s]', connectionState);
 });
 },
 handleTransportConnectEvent({ dtlsParameters }, callback, errback) {
 console.log('handleTransportConnectEvent()');
 try {
 this.connectTransport = (msg) => {
 console.log('connect-transport action');
 callback();
 this.connectTransport = null;
 };

 this.socket.emit('connect-transport',{
 sessionId: this.peer.sessionId,
 transportId: this.peer.sendTransport.id,
 dtlsParameters
 });

 } catch (error) {
 console.error('handleTransportConnectEvent() failed [error:%o]', error);
 errback(error);
 }
 },
 handleTransportProduceEvent({ kind, rtpParameters }, callback, errback) {
 console.log('handleTransportProduceEvent()');
 try {
 this.produce = jsonMessage => {
 console.log('handleTransportProduceEvent callback [data:%o]', jsonMessage);
 callback({ id: jsonMessage.id });
 this.produce = null;
 };

 this.socket.emit('produce', {
 sessionId: this.peer.sessionId,
 transportId: this.peer.sendTransport.id,
 kind,
 rtpParameters
 });
 } catch (error) {
 console.error('handleTransportProduceEvent() failed [error:%o]', error);
 errback(error);
 }
 },
 async recordBySapioServer() {
 this.loading = true;
 this.connectToSocket();
 },
 async setTracks() {
 // Start mediasoup-client's WebRTC producers
 const audioTrack = this.localStream.getAudioTracks()[0];
 this.peer.audioProducer = await this.peer.sendTransport.produce({
 track: audioTrack,
 codecOptions :
 {
 opusStereo : 1,
 opusDtx : 1
 }
 });


 let encodings;
 let codec;
 const codecOptions = {videoGoogleStartBitrate : 1000};

 codec = this.peer.device.rtpCapabilities.codecs.find((c) => c.kind.toLowerCase() === 'video');
 if (codec.mimeType.toLowerCase() === 'video/vp9') {
 encodings = { scalabilityMode: 'S3T3_KEY' };
 } else {
 encodings = [
 { scaleResolutionDownBy: 4, maxBitrate: 500000 },
 { scaleResolutionDownBy: 2, maxBitrate: 1000000 },
 { scaleResolutionDownBy: 1, maxBitrate: 5000000 }
 ];
 }
 const videoTrack = this.localStream.getVideoTracks()[0];
 this.peer.videoProducer =await this.peer.sendTransport.produce({
 track: videoTrack,
 encodings,
 codecOptions,
 codec
 });

 },
 startRecording() {
 this.Q.answer.recordingId = this.peer.externalId;
 this.socket.emit("start-record", {
 sessionId: this.peer.sessionId
 });
 },
 stopRecording() {
 this.socket.emit("stop-record" , {
 sessionId: this.peer.sessionId
 });
 },
 },

}






console.log of my ffmpeg process :


// sdp string
[sdpString:v=0
 o=- 0 0 IN IP4 127.0.0.1
 s=FFmpeg
 c=IN IP4 127.0.0.1
 t=0 0
 m=video 25549 RTP/AVP 101 
 a=rtpmap:101 VP8/90000
 a=sendonly
 m=audio 26934 RTP/AVP 100 
 a=rtpmap:100 opus/48000/2
 a=sendonly
 ]

// ffmpeg args
commandArgs:[
 '-loglevel',
 'debug',
 '-protocol_whitelist',
 'pipe,udp,rtp',
 '-fflags',
 '+genpts',
 '-f',
 'sdp',
 '-i',
 'pipe:0',
 '-map',
 '0:v:0',
 '-c:v',
 'copy',
 '-map',
 '0:a:0',
 '-strict',
 '-2',
 '-c:a',
 'copy',
 '-f',
 'webm',
 '-flags',
 '+global_header',
 '-y',
 'storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm',
 [length]: 26
]
// ffmpeg log
ffmpeg::process::data [data:'ffmpeg version n4.4']
ffmpeg::process::data [data:' Copyright (c) 2000-2021 the FFmpeg developers']
ffmpeg::process::data [data:'\n']
ffmpeg::process::data [data:' built with gcc 11.1.0 (GCC)\n']
ffmpeg::process::data [data:' configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-shared --enable-version3\n']
ffmpeg::process::data [data:' libavutil 56. 70.100 / 56. 70.100\n' +
 ' libavcodec 58.134.100 / 58.134.100\n' +
 ' libavformat 58. 76.100 / 58. 76.100\n' +
 ' libavdevice 58. 13.100 / 58. 13.100\n' +
 ' libavfilter 7.110.100 / 7.110.100\n' +
 ' libswscale 5. 9.100 / 5. 9.100\n' +
 ' libswresample 3. 9.100 / 3. 9.100\n' +
 ' libpostproc 55. 9.100 / 55. 9.100\n' +
 'Splitting the commandline.\n' +
 "Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.\n" +
 "Reading option '-protocol_whitelist' ..."]
ffmpeg::process::data [data:" matched as AVOption 'protocol_whitelist' with argument 'pipe,udp,rtp'.\n" +
 "Reading option '-fflags' ..."]
ffmpeg::process::data [data:" matched as AVOption 'fflags' with argument '+genpts'.\n" +
 "Reading option '-f' ... matched as option 'f' (force format) with argument 'sdp'.\n" +
 "Reading option '-i' ... matched as input url with argument 'pipe:0'.\n" +
 "Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:v:0'.\n" +
 "Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.\n" +
 "Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:a:0'.\n" +
 "Reading option '-strict' ...Routing option strict to both codec and muxer layer\n" +
 " matched as AVOption 'strict' with argument '-2'.\n" +
 "Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'.\n" +
 "Reading option '-f' ... matched as option 'f' (force format) with argument 'webm'.\n" +
 "Reading option '-flags' ... matched as AVOption 'flags' with argument '+global_header'.\n" +
 "Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.\n" +
 "Reading option 'storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm' ... matched as output url.\n" +
 'Finished splitting the commandline.\n' +
 'Parsing a group of options: global .\n' +
 'Applying option loglevel (set logging level) with argument debug.\n' +
 'Applying option y (overwrite output files) with argument 1.\n' +
 'Successfully parsed a group of options.\n' +
 'Parsing a group of options: input url pipe:0.\n' +
 'Applying option f (force format) with argument sdp.\n' +
 'Successfully parsed a group of options.\n' +
 'Opening an input file: pipe:0.\n' +
 "[sdp @ 0x55604dc58400] Opening 'pipe:0' for reading\n" +
 '[sdp @ 0x55604dc58400] video codec set to: vp8\n' +
 '[sdp @ 0x55604dc58400] audio codec set to: opus\n' +
 '[sdp @ 0x55604dc58400] audio samplerate set to: 48000\n' +
 '[sdp @ 0x55604dc58400] audio channels set to: 2\n' +
 '[udp @ 0x55604dc6c500] end receive buffer size reported is 425984\n' +
 '[udp @ 0x55604dc6c7c0] end receive buffer size reported is 425984\n' +
 '[sdp @ 0x55604dc58400] setting jitter buffer size to 500\n' +
 '[udp @ 0x55604dc6d900] end receive buffer size reported is 425984\n' +
 '[udp @ 0x55604dc6d2c0] end receive buffer size reported is 425984\n' +
 '[sdp @ 0x55604dc58400] setting jitter buffer size to 500\n']
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] Before avformat_find_stream_info() pos: 210 bytes read:210 seeks:0 nb_streams:2\n']
 **mediasoup:Consumer resume() +1s**
 **mediasoup:Channel request() [method:consumer.resume, id:12] +1s**
 **mediasoup:Channel request succeeded [method:consumer.resume, id:12] +0ms**
 **mediasoup:Consumer resume() +1ms**
 **mediasoup:Channel request() [method:consumer.resume, id:13] +0ms**
 **mediasoup:Channel request succeeded [method:consumer.resume, id:13] +0ms**
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] Could not find codec parameters for stream 0 (Video: vp8, 1 reference frame, yuv420p): unspecified size\n' +
 "Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options\n"]
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] After avformat_find_stream_info() pos: 210 bytes read:210 seeks:0 frames:0\n' +
 "Input #0, sdp, from 'pipe:0':\n" +
 ' Metadata:\n' +
 ' title : FFmpeg\n' +
 ' Duration: N/A, bitrate: N/A\n' +
 ' Stream #0:0, 0, 1/90000: Video: vp8, 1 reference frame, yuv420p, 90k tbr, 90k tbn, 90k tbc\n' +
 ' Stream #0:1, 0, 1/48000: Audio: opus, 48000 Hz, stereo, fltp\n' +
 'Successfully opened the file.\n' +
 'Parsing a group of options: output url storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm.\n' +
 'Applying option map (set input stream mapping) with argument 0:v:0.\n' +
 'Applying option c:v (codec name) with argument copy.\n' +
 'Applying option map (set input stream mapping) with argument 0:a:0.\n' +
 'Applying option c:a (codec name) with argument copy.\n' +
 'Applying option f (force format) with argument webm.\n' +
 'Successfully parsed a group of options.\n' +
 'Opening an output file: storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm.\n' +
 "[file @ 0x55604dce5bc0] Setting default whitelist 'file,crypto,data'\n"]
ffmpeg::process::data [data:'Successfully opened the file.\n' +
 '[webm @ 0x55604dce0fc0] dimensions not set\n' +
 'Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument\n' +
 'Error initializing output stream 0:1 -- \n' +
 'Stream mapping:\n' +
 ' Stream #0:0 -> #0:0 (copy)\n' +
 ' Stream #0:1 -> #0:1 (copy)\n' +
 ' Last message repeated 1 times\n' +
 '[AVIOContext @ 0x55604dc6dcc0] Statistics: 0 seeks, 0 writeouts\n' +
 '[AVIOContext @ 0x55604dc69380] Statistics: 210 bytes read, 0 seeks\n']
ffmpeg::process::close




FFmpeg says
dimensions not set
andCould not write header for output file
when I use Firefox. This might be enough for understanding the problem, but if you need more information you can read how server side is performing.
Server-Side in summary can be something like this :
lets say we initialized worker and router at run time using following functions.

// Start the mediasoup workers
module.exports.initializeWorkers = async () => {
 const { logLevel, logTags, rtcMinPort, rtcMaxPort } = config.worker;

 console.log('initializeWorkers() creating %d mediasoup workers', config.numWorkers);

 for (let i = 0; i < config.numWorkers; ++i) {
 const worker = await mediasoup.createWorker({
 logLevel, logTags, rtcMinPort, rtcMaxPort
 });

 worker.once('died', () => {
 console.error('worker::died worker has died exiting in 2 seconds... [pid:%d]', worker.pid);
 setTimeout(() => process.exit(1), 2000);
 });

 workers.push(worker);
 }
};



module.exports.createRouter = async () => {
 const worker = getNextWorker();

 console.log('createRouter() creating new router [worker.pid:%d]', worker.pid);

 console.log(`config.router.mediaCodecs:${JSON.stringify(config.router.mediaCodecs)}`)

 return await worker.createRouter({ mediaCodecs: config.router.mediaCodecs });
};



We pass
router.rtpCompatibilities
to the client. clients get thertpCompatibilities
and create a device and loads it. after that a transport must be created at server side.

const handleCreateTransportRequest = async (jsonMessage) => {

 const transport = await createTransport('webRtc', router);

 var peer;
 try {peer = peers.get(jsonMessage.sessionId);}
 catch{console.log('peer not found')}
 
 peer.addTransport(transport);

 peer.socket.emit('create-transport',{
 id: transport.id,
 iceParameters: transport.iceParameters,
 iceCandidates: transport.iceCandidates,
 dtlsParameters: transport.dtlsParameters
 });
};



Then after the client side also created the transport we listen to connect event an at the time of event, we request the server to create connection.


const handleTransportConnectRequest = async (jsonMessage) => {
 var peer;
 try {peer = peers.get(jsonMessage.sessionId);}
 catch{console.log('peer not found')}

 if (!peer) {
 throw new Error(`Peer with id ${jsonMessage.sessionId} was not found`);
 }

 const transport = peer.getTransport(jsonMessage.transportId);

 if (!transport) {
 throw new Error(`Transport with id ${jsonMessage.transportId} was not found`);
 }

 await transport.connect({ dtlsParameters: jsonMessage.dtlsParameters });
 console.log('handleTransportConnectRequest() transport connected');
 peer.socket.emit('connect-transport');
};



Similar thing happen on produce event.


const handleProduceRequest = async (jsonMessage) => {
 console.log('handleProduceRequest [data:%o]', jsonMessage);

 var peer;
 try {peer = peers.get(jsonMessage.sessionId);}
 catch{console.log('peer not found')}

 if (!peer) {
 throw new Error(`Peer with id ${jsonMessage.sessionId} was not found`);
 }

 const transport = peer.getTransport(jsonMessage.transportId);

 if (!transport) {
 throw new Error(`Transport with id ${jsonMessage.transportId} was not found`);
 }

 const producer = await transport.produce({
 kind: jsonMessage.kind,
 rtpParameters: jsonMessage.rtpParameters
 });

 peer.addProducer(producer);

 console.log('handleProducerRequest() new producer added [id:%s, kind:%s]', producer.id, producer.kind);

 peer.socket.emit('produce',{
 id: producer.id,
 kind: producer.kind
 });
};



For Recording, first I create plain transports for audio and video producers.


const rtpTransport = router.createPlainTransport(config.plainRtpTransport);



then rtp transport must be connected to ports :


await rtpTransport.connect({
 ip: '127.0.0.1',
 port: remoteRtpPort,
 rtcpPort: remoteRtcpPort
 });



Then the consumer must also be created.


const rtpConsumer = await rtpTransport.consume({
 producerId: producer.id,
 rtpCapabilities,
 paused: true
 });



After that we can start recording using following code :


this._rtpParameters = args;
 this._process = undefined;
 this._observer = new EventEmitter();
 this._peer = args.peer;

 this._sdpString = createSdpText(this._rtpParameters);
 this._sdpStream = convertStringToStream(this._sdpString);
 // create dir
 const dir = process.env.REOCRDING_PATH ?? 'storage/recordings';
 if (!fs.existsSync(dir)) shelljs.mkdir('-p', dir);
 
 this._extension = 'webm';
 // create file path
 this._path = `${dir}/${args.peer.sessionId}.${this._extension}`
 let loop = 0;
 while(fs.existsSync(this._path)) {
 this._path = `${dir}/${args.peer.sessionId}-${++loop}.${this._extension}`
 }

this._recordingnModel = await Recording.findOne({sessionIds: { $in: [this._peer.sessionId] }})
 this._recordingnModel.files.push(this._path);
 this._recordingnModel.save();

let proc = ffmpeg(this._sdpStream)
 .inputOptions([
 '-protocol_whitelist','pipe,udp,rtp',
 '-f','sdp',
 ])
 .format(this._extension)
 .output(this._path)
 .size('720x?')
 .on('start', ()=>{
 this._peer.socket.emit('recording');
 })
 .on('end', ()=>{
 let path = this._path.replace('storage/recordings/', '');
 this._peer.socket.emit('recording-closed', {
 url: `${process.env.APP_URL}/recording/file/${path}`
 });
 });

 proc.run();
 this._process = proc;
 }




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