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26 septembre 2011, par
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Autres articles (28)
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MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 is the first MediaSPIP stable release.
Its official release date is June 21, 2013 and is announced here.
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...) -
De l’upload à la vidéo finale [version standalone]
31 janvier 2010, parLe chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
Upload et récupération d’informations de la vidéo source
Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)
Sur d’autres sites (4072)
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How to Stream RTP (IP camera) Into React App setup
10 novembre 2024, par sharon2469I am trying to transfer a live broadcast from an IP camera or any other broadcast coming from an RTP/RTSP source to my REACT application. BUT MUST BE LIVE


My setup at the moment is :


IP Camera -> (RTP) -> FFmpeg -> (udp) -> Server(nodeJs) -> (WebRTC) -> React app


In the current situation, There is almost no delay, but there are some things here that I can't avoid and I can't understand why, and here is my question :


1) First, is the SETUP even correct and this is the only way to Stream RTP video in Web app ?


2) Is it possible to avoid re-encode the stream , RTP transmission necessarily comes in H.264, hence I don't really need to execute the following command :


return spawn('ffmpeg', [
 '-re', // Read input at its native frame rate Important for live-streaming
 '-probesize', '32', // Set probing size to 32 bytes (32 is minimum)
 '-analyzeduration', '1000000', // An input duration of 1 second
 '-c:v', 'h264', // Video codec of input video
 '-i', 'rtp://238.0.0.2:48888', // Input stream URL
 '-map', '0:v?', // Select video from input stream
 '-c:v', 'libx264', // Video codec of output stream
 '-preset', 'ultrafast', // Faster encoding for lower latency
 '-tune', 'zerolatency', // Optimize for zero latency
 // '-s', '768x480', // Adjust the resolution (experiment with values)
 '-f', 'rtp', `rtp://127.0.0.1:${udpPort}` // Output stream URL
]);



As you can se in this command I re-encode to libx264, But if I set FFMPEG a parameter '-c:v' :'copy' instead of '-c:v', 'libx264' then FFMPEG throw an error says : that it doesn't know how to encode h264 and only knows what is libx264-> Basically, I want to stop the re-encode because there is really no need for it, because the stream is already encoded to H264. Are there certain recommendations that can be made ?


3) I thought about giving up the FFMPEG completely, but the RTP packets arrive at a size of 1200+ BYTES when WEBRTC is limited to up to 1280 BYTE. Is there a way to manage these sabotages without damaging the video and is it to enter this world ? I guess there is the whole story with the JITTER BUFFER here


This is my server side code (THIS IS JUST A TEST CODE)


import {
 MediaStreamTrack,
 randomPort,
 RTCPeerConnection,
 RTCRtpCodecParameters,
 RtpPacket,
} from 'werift'
import {Server} from "ws";
import {createSocket} from "dgram";
import {spawn} from "child_process";
import LoggerFactory from "./logger/loggerFactory";

//

const log = LoggerFactory.getLogger('ServerMedia')

// Websocket server -> WebRTC
const serverPort = 8888
const server = new Server({port: serverPort});
log.info(`Server Media start om port: ${serverPort}`);

// UDP server -> ffmpeg
const udpPort = 48888
const udp = createSocket("udp4");
// udp.bind(udpPort, () => {
// udp.addMembership("238.0.0.2");
// })
udp.bind(udpPort)
log.info(`UDP port: ${udpPort}`)


const createFFmpegProcess = () => {
 log.info(`Start ffmpeg process`)
 return spawn('ffmpeg', [
 '-re', // Read input at its native frame rate Important for live-streaming
 '-probesize', '32', // Set probing size to 32 bytes (32 is minimum)
 '-analyzeduration', '1000000', // An input duration of 1 second
 '-c:v', 'h264', // Video codec of input video
 '-i', 'rtp://238.0.0.2:48888', // Input stream URL
 '-map', '0:v?', // Select video from input stream
 '-c:v', 'libx264', // Video codec of output stream
 '-preset', 'ultrafast', // Faster encoding for lower latency
 '-tune', 'zerolatency', // Optimize for zero latency
 // '-s', '768x480', // Adjust the resolution (experiment with values)
 '-f', 'rtp', `rtp://127.0.0.1:${udpPort}` // Output stream URL
 ]);

}

let ffmpegProcess = createFFmpegProcess();


const attachFFmpegListeners = () => {
 // Capture standard output and print it
 ffmpegProcess.stdout.on('data', (data) => {
 log.info(`FFMPEG process stdout: ${data}`);
 });

 // Capture standard error and print it
 ffmpegProcess.stderr.on('data', (data) => {
 console.error(`ffmpeg stderr: ${data}`);
 });

 // Listen for the exit event
 ffmpegProcess.on('exit', (code, signal) => {
 if (code !== null) {
 log.info(`ffmpeg process exited with code ${code}`);
 } else if (signal !== null) {
 log.info(`ffmpeg process killed with signal ${signal}`);
 }
 });
};


attachFFmpegListeners();


server.on("connection", async (socket) => {
 const payloadType = 96; // It is a numerical value that is assigned to each codec in the SDP offer/answer exchange -> for H264
 // Create a peer connection with the codec parameters set in advance.
 const pc = new RTCPeerConnection({
 codecs: {
 audio: [],
 video: [
 new RTCRtpCodecParameters({
 mimeType: "video/H264",
 clockRate: 90000, // 90000 is the default value for H264
 payloadType: payloadType,
 }),
 ],
 },
 });

 const track = new MediaStreamTrack({kind: "video"});


 udp.on("message", (data) => {
 console.log(data)
 const rtp = RtpPacket.deSerialize(data);
 rtp.header.payloadType = payloadType;
 track.writeRtp(rtp);
 });

 udp.on("error", (err) => {
 console.log(err)

 });

 udp.on("close", () => {
 console.log("close")
 });

 pc.addTransceiver(track, {direction: "sendonly"});

 await pc.setLocalDescription(await pc.createOffer());
 const sdp = JSON.stringify(pc.localDescription);
 socket.send(sdp);

 socket.on("message", (data: any) => {
 if (data.toString() === 'resetFFMPEG') {
 ffmpegProcess.kill('SIGINT');
 log.info(`FFMPEG process killed`)
 setTimeout(() => {
 ffmpegProcess = createFFmpegProcess();
 attachFFmpegListeners();
 }, 5000)
 } else {
 pc.setRemoteDescription(JSON.parse(data));
 }
 });
});



And this fronted :





 
 
 <code class="echappe-js"><script&#xA; crossorigin&#xA; src="https://unpkg.com/react@16/umd/react.development.js"&#xA; ></script>

<script&#xA; crossorigin&#xA; src="https://unpkg.com/react-dom@16/umd/react-dom.development.js"&#xA; ></script>

<script&#xA; crossorigin&#xA; src="https://cdnjs.cloudflare.com/ajax/libs/babel-core/5.8.34/browser.min.js"&#xA; ></script>

<script src="https://cdn.jsdelivr.net/npm/babel-regenerator-runtime@6.5.0/runtime.min.js"></script>








<script type="text/babel">&#xA; let rtc;&#xA;&#xA; const App = () => {&#xA; const [log, setLog] = React.useState([]);&#xA; const videoRef = React.useRef();&#xA; const socket = new WebSocket("ws://localhost:8888");&#xA; const [peer, setPeer] = React.useState(null); // Add state to keep track of the peer connection&#xA;&#xA; React.useEffect(() => {&#xA; (async () => {&#xA; await new Promise((r) => (socket.onopen = r));&#xA; console.log("open websocket");&#xA;&#xA; const handleOffer = async (offer) => {&#xA; console.log("new offer", offer.sdp);&#xA;&#xA; const updatedPeer = new RTCPeerConnection({&#xA; iceServers: [],&#xA; sdpSemantics: "unified-plan",&#xA; });&#xA;&#xA; updatedPeer.onicecandidate = ({ candidate }) => {&#xA; if (!candidate) {&#xA; const sdp = JSON.stringify(updatedPeer.localDescription);&#xA; console.log(sdp);&#xA; socket.send(sdp);&#xA; }&#xA; };&#xA;&#xA; updatedPeer.oniceconnectionstatechange = () => {&#xA; console.log(&#xA; "oniceconnectionstatechange",&#xA; updatedPeer.iceConnectionState&#xA; );&#xA; };&#xA;&#xA; updatedPeer.ontrack = (e) => {&#xA; console.log("ontrack", e);&#xA; videoRef.current.srcObject = e.streams[0];&#xA; };&#xA;&#xA; await updatedPeer.setRemoteDescription(offer);&#xA; const answer = await updatedPeer.createAnswer();&#xA; await updatedPeer.setLocalDescription(answer);&#xA;&#xA; setPeer(updatedPeer);&#xA; };&#xA;&#xA; socket.onmessage = (ev) => {&#xA; const data = JSON.parse(ev.data);&#xA; if (data.type === "offer") {&#xA; handleOffer(data);&#xA; } else if (data.type === "resetFFMPEG") {&#xA; // Handle the resetFFMPEG message&#xA; console.log("FFmpeg reset requested");&#xA; }&#xA; };&#xA; })();&#xA; }, []); // Added socket as a dependency to the useEffect hook&#xA;&#xA; const sendRequestToResetFFmpeg = () => {&#xA; socket.send("resetFFMPEG");&#xA; };&#xA;&#xA; return (&#xA; <div>&#xA; Video: &#xA; <video ref={videoRef} autoPlay muted />&#xA; <button onClick={() => sendRequestToResetFFmpeg()}>Reset FFMPEG</button>&#xA; </div>&#xA; );&#xA; };&#xA;&#xA; ReactDOM.render(<App />, document.getElementById("app1"));&#xA;</script>





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