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999,999
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
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The Slip - Artworks
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Demon seed (wav version)
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Sur d’autres sites (10873)
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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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Using ffserver to stream older IP cam MJPEG to RTSP
26 mai 2016, par tmar89I have an older Sony IP camera that has an MJPEG stream. I need to connect this to an NVR that only takes ONVIP or RTSP and I’m trying to use ffserver and ffmpeg to convert the MJPEG stream to RTSP but it’s not working. Any have some idea of what I may be doing wrong ? Saw an error in the attempted playback about an unsupported Protocol.
Here is my ffserver config :Port 8090
RTSPPort 5544
BindAddress 0.0.0.0
RTSPBindAddress 0.0.0.0
MaxClients 100
MaxBandwidth 10000
<feed>
File /tmp/feed27.ffm
FileMaxSize 5M
ACL allow 127.0.0.1
</feed>
<stream>
Format rtp
Feed feed27.ffm
NoAudio
VideoCodec mjpeg
VideoFrameRate 30
VideoSize 736x480
</stream>And here is the ffmpeg command I am using :
[tm@tele ffserver-rtsp]# ffmpeg -f mjpeg -r 30 -s 736x480 -i http://[CAMIP]/image http://localhost:8090/feed27.ffm
FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
libavutil 50.15. 1 / 50.15. 1
libavcodec 52.72. 2 / 52.72. 2
libavformat 52.64. 2 / 52.64. 2
libavdevice 52. 2. 0 / 52. 2. 0
libavfilter 1.19. 0 / 1.19. 0
libswscale 0.11. 0 / 0.11. 0
libpostproc 51. 2. 0 / 51. 2. 0
[mjpeg @ 0x1ece670]Estimating duration from bitrate, this may be inaccurate
Input #0, mjpeg, from 'http://[CAMIP]/image':
Duration: N/A, bitrate: N/A
Stream #0.0: Video: mjpeg, yuvj422p, 736x480, 30 fps, 30 tbr, 1200k tbn, 30 tbc
Output #0, ffm, to 'http://localhost:8090/feed27.ffm':
Metadata:
encoder : Lavf52.64.2
Stream #0.0: Video: mjpeg, yuvj420p, 736x480, q=2-31, 200 kb/s, 1000k tbn, 30 tbc
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop encoding
[mjpeg @ 0x222d110]rc buffer underflow
frame= 640 fps= 17 q=31.4 size= 12884kB time=21.33 bitrate=4947.5kbits/sWhen I use VLC to open the stream, it cannot be found :
Your input can't be opened:
VLC is unable to open the MRL 'rtsp://localhost:5544/stream27.mpg'. Check the log for details.Finally, using ffplay on the same machine :
[tm@tele tmp]# ffplay rtsp://localhost:5544/stream27.sdp
FFplay version 0.6.5, Copyright (c) 2003-2010 the FFmpeg developers
built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
libavutil 50.15. 1 / 50.15. 1
libavcodec 52.72. 2 / 52.72. 2
libavformat 52.64. 2 / 52.64. 2
libavdevice 52. 2. 0 / 52. 2. 0
libavfilter 1.19. 0 / 1.19. 0
libswscale 0.11. 0 / 0.11. 0
libpostproc 51. 2. 0 / 51. 2. 0
ALSA lib pulse.c:229:(pulse_connect) PulseAudio: Unable to connect: Connection refused
rtsp://localhost:5544/stream27.sdp: Protocol not supportedAnd here was the log from ffserver :
127.0.0.1:5000 - - "PLAY stream27.mpg/streamid=0 RTP/UDP"
[rtp @ 0x721dc0]Unsupported codec 8
127.0.0.1:0 - - "PLAY stream27.mpg/streamid=0 RTP/TCP"
[rtp @ 0x728cb0]Unsupported codec 8
127.0.0.1 - - [SETUP] "rtsp://localhost:5544/stream27.mpg/streamid=0 RTSP/1.0" 200 641 -
How to stop ffmpeg when there's no incoming rtmp stream
5 juillet 2016, par M. IrichI use ffmpeg together with nginx-rtmp.
The thing is ffmpeg doesn’t finish the process when the stream’s finishedI use the following command :
ffmpeg -i 'rtmp://localhost:443/live/test' -loglevel debug -c:a libfdk_aac -b:a 192k -c:v libx264 -profile baseline -preset superfast -tune zerolatency -b:v 2500k -maxrate 4500k -minrate 1500k -bufsize 9000k -keyint_min 15 -g 15 -f dash -use_timeline 1 -use_template 1 -min_seg_duration 5000 -y /tmp/dash/test/test.mpdbut even the stream’s not running ffmpeg still can’t finish the process and is waiting for the rtmp stream
Successfully parsed a group of options.
Opening an input file: rtmp://localhost:443/live/test.
[rtmp @ 0x2ba2160] No default whitelist set
[tcp @ 0x2ba2720] No default whitelist set
[rtmp @ 0x2ba2160] Handshaking...
[rtmp @ 0x2ba2160] Type answer 3
[rtmp @ 0x2ba2160] Server version 13.14.10.13
[rtmp @ 0x2ba2160] Proto = rtmp, path = /live/test, app = live, fname = test
[rtmp @ 0x2ba2160] Server bandwidth = 5000000
[rtmp @ 0x2ba2160] Client bandwidth = 5000000
[rtmp @ 0x2ba2160] New incoming chunk size = 4096
[rtmp @ 0x2ba2160] Creating stream...
[rtmp @ 0x2ba2160] Sending play command for 'test'Is it possible to limit the latency time to several seconds ?
Sorry for any possible mistakes - English’s not my native language.
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