The current code work absolutely great for streaming single video camera. For adding new camera, I have to run a new server i.e. I have to copy the script, update the RTSP link and port in it and then start the new server.

Is there any way where a single script will allow for streaming multiple cameras? That is I can use the same port number for streaming the video/camera of user's choice.

I have already made the changes to this script to make it stream only when a client is connected and stop the stream when no one is connected. Have also made it SSL enabled.

Have also made the changes to be able to pass the RTSP link when connecting to the socket but now I am just not able to make it stream. It always stream the first RTSP link.

Any suggestion on how to do this?

Sometimes I received a lot of missing packets. My client browser will just turn grey. This happens a lot. I have set the -max_delay to 100000 but it is still happening. Any reason why this happened?

0|app      | [rtsp @ 0x56372214a9a0] RTP: missed 1 packets
0|app      | [hevc @ 0x563722180900] Could not find ref with POC 20
0|app      | [rtsp @ 0x56372214a9a0] max delay reached. need to consume packet
0|app      | [rtsp @ 0x56372214a9a0] RTP: missed 1 packets
0|app      | [hevc @ 0x563722180900] Could not find ref with POC 25

There are three ffmpegs in the task manager

I notice that only client connections coming in via Localhost are recognized and served. The FFMPEG configurations seem to be using the standard defaults, but I was hoping to be able to recognize those client connections from either other devices on my local network or devices off network accessing via IP. These clients will end up being jsmpeg react components, Localhost jsmpeg components work swimmingly but nothing is making it outside of localhost.

This particular version (0.0.8) gives blurry videos. The problem gets solved if I use version 0.0.5.

Hi again

Now, I'm monitoring DVR camera by your project through web. It's Perfect
I have a question. Can I capture from stream video and save it as jpg, for example every 1 min?

Thanks a lot

Hello,

Firs tof all thanks for this code, its exactly what I was looking for.

One remark, you code is no longer aligned with jsmpeg library. I had to find older versions of jsmpeg lib for the code to work .... The minified version does not suppor sending a websocket as an argument.

I am trying to use your code to display a FOSCAM IP webcam I have at home though a rapsberry pi.
I have a nodejs express that is getting the url I am calling and starting the stream.

I call node-rtsp-stream on port 3333
my videostream is http://login:password@ip:554/videoMain
my port is 333

Then I create using localtunnnel a tunnel to one of my hosted servers
this maps the local port 3333 to the url given by localtunnel

This all goes fine and generates a html page with the relative infos ....
Result is this :

I get a blank page with this in the console :

86[.WebGL-000002121E2B6A00]RENDER WARNING: texture bound to texture unit 0 is not renderable. It maybe non-power-of-2 and have incompatible texture filtering.
85[.WebGL-000002121E2B6A00]RENDER WARNING: texture bound to texture unit 1 is not renderable. It maybe non-power-of-2 and have incompatible texture filtering.
85[.WebGL-000002121E2B6A00]RENDER WARNING: texture bound to texture unit 2 is not renderable. It maybe non-power-of-2 and have incompatible texture filtering.
1:1 WebGL: too many errors, no more errors will be reported to the console for this context.
[Violation] 'setTimeout' handler took 114ms
[Violation] 'message' handler took 155ms
[Violation] 'setTimeout' handler took 58ms
[Violation] 'message' handler took 727ms
[Violation] 'message' handler took 156ms

hi guys,
I'm trying to use node-rtsp-stream and got following error on my web page.

TypeError: this.startMpeg1Stream is not a function

at VideoStream (D:\Repos\floss\ffmpegToWeb\node_modules\node-rtsp-stream\videoStream.js:22:8)

at D:\Repos\floss\ffmpegToWeb\server.js:20:14

at Layer.handle [as handle_request] (D:\Repos\floss\ffmpegToWeb\node_modules\express\lib\router\layer.js:95:5)

at next (D:\Repos\floss\ffmpegToWeb\node_modules\express\lib\router\route.js:137:13)

at Route.dispatch (D:\Repos\floss\ffmpegToWeb\node_modules\express\lib\router\route.js:112:3)

at Layer.handle [as handle_request] (D:\Repos\floss\ffmpegToWeb\node_modules\express\lib\router\layer.js:95:5)

at D:\Repos\floss\ffmpegToWeb\node_modules\express\lib\router\index.js:281:22

at Function.process_params (D:\Repos\floss\ffmpegToWeb\node_modules\express\lib\router\index.js:335:12)

at next (D:\Repos\floss\ffmpegToWeb\node_modules\express\lib\router\index.js:275:10)

at expressInit (D:\Repos\floss\ffmpegToWeb\node_modules\express\lib\middleware\init.js:40:5)

my server is just simply a express server which respond to requests and send the html back, I init the stream in the new connection callback of server.
my html is just like the example provided in the README.

Any ideas?
Thanks in advance.

I currently have several cameras with RSTP, so I tried to run the Stream object instance on the server more than once.

After consulting the database, I fill in the data to create the corresponding RTSP URL.

However, it only displays the first and the others do not.

I will want to view 9 cameras on page web simultaneously.  Is that possible with node-rtsp-stream?

Then, in the stream object even creating for 9 cameras, it only displays the first one.

Hello, I cannot deploy my stream server on Heroku, because it have issue with the ws Port. Heroku just allow one Port but I need 2 to (ws Port and Server Port) to play stream. Anyone can help me?
Thank you so much!

-rtsp_transport tcp need to put before '-i'.but options all put behind ‘-i’

buffer.js:784
throw TypeError('value is out of bounds');
^
TypeError: value is out of bounds
at TypeError ()
at checkInt (buffer.js:784:11)
at Buffer.writeUInt16BE (buffer.js:819:5)
at VideoStream.onSocketConnect (/home/ubuntu/nrs/node_modules/node-rtsp-stream/lib/videoStream.js:110:18)
at WebSocketServer. (/home/ubuntu/nrs/node_modules/node-rtsp-stream/lib/videoStream.js:86:19)
at WebSocketServer.EventEmitter.emit (events.js:95:17)
at /home/ubuntu/nrs/node_modules/node-rtsp-stream/node_modules/ws/lib/WebSocketServer.js:74:14
at completeHybiUpgrade2 (/home/ubuntu/nrs/node_modules/node-rtsp-stream/node_modules/ws/lib/WebSocketServer.js:237:5)
at completeHybiUpgrade1 (/home/ubuntu/nrs/node_modules/node-rtsp-stream/node_modules/ws/lib/WebSocketServer.js:262:13)
at WebSocketServer.handleHybiUpgrade (/home/ubuntu/nrs/node_modules/node-rtsp-stream/node_modules/ws/lib/WebSocketServer.js:290:3)

Is there a simple way to restart the stream if the ffmpeg process terminates unexpectedly?

Hello

This project is very interesting and useful for me
But can I stream multi channel by this way?

Thanks a lot

Noticed that the example code was complaining about "Websocket" not being defined on the client.js script. Just thought of pointing out that "Websocket" needs to change to "WebSocket". :)

Hi,
Thank you for this lib.

For production, chrome does not allow ws:// request if web site is https://. All connections should be secure.

Is there any workaround for this?

I'm streaming an ip camera and i need to resize the video of the streaming. I tried resizing the canvas and it doesn't work. I also tried create a new stream with different ffmpeg flags and a new JSMpeg.Player() on the same port, and it doesn't work.

Any ideas?

There are a ton of Javascript developers in the NodeJS and frontend world that would support a project like this. Coffeescript can be confusing to use when compared to regular JS. Not only is it an entirely different syntax, it doesn't even use the same variables (@ instead of this). It's also a rarely used language in professional environments, as most people are migrating toward Typescript for strongly typed languages.

Suggestion:

• Convert coffee files to ts
• Remove /lib from github, as that is a compiled folder that should only be published to npm
• Convert tests to js or ts files.
• Update package.json to compile ts to js for packaging and testing

I'm creating this issue now, with the hopes that I can offer a PR later (a sort of TODO). I'd be happy if someone can take it.

Reference comment in PR #16 (comment)

my video is 4096x2160 , cannot play it:

=====================================
Input #0, rtsp, from 'rtsp://xxxxxxx/.trunk.ts':
Metadata:
title : MPEG Transport Stream, streamed by the LIVE555 Media Server
comment : long.trunk.ts
Duration: N/A, start: 1.400000, bitrate: N/A
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0: Video: mpeg4 (Simple Profile) ([16][0][0][0] / 0x0010), yuv420p, 4096x2160 [SAR 1:1 DAR 256:135], 15 fps, 15 tbr, 90k tbn, 15 tbc
Stream mapping:
Stream #0:0 -> #0:0 (mpeg4 (native) -> mpeg1video (native))
Press [q] to stop, [?] for help
[mpeg1video @ 0x1129580] MPEG-1 does not support resolutions above 4095x4095
Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
Conversion failed!

Hi,

I am able to run my nodejs application locally on a windows pc.

making use of node-rtsp-stream, backend retrieve a RTSP stream online and my front end display the stream.

however I am having trouble after deploying it into microsoft azure, seeing Error: spawn ffmpeg ENOENT in my azure's log. (I have also tried to include ffmpeg.exe into my app)

How should i go about 'installing' FFMPEG for deployment to microsoft azure?

Hi, I am able to run the project successfully. I am now trying to adapt it into an angular project. adding the client code in component.html doesn't show the jsmpeg player canvas. I got no errors in console.
my ws server stream is working as I am able to stream using the same client code on another non angular .html.

Thanks

Is there an option to play audio in browser? The audio codec in AAC.

How to stream multiple rtsp in the same port ?

I have to stream multiple RTSP link from multiple RTSP and I have many client to view diferrent and multiple rtsp sametime. Is this lib support this case ?

Some cameras' RTSPs require authorization to access them. Now this module seems to be not authorized to access RTSP.

When passing additional loglevel parameter, the streaming stop working and could not get rendered within a browser canvas. Only with one parameter it is working fine.

ffmpegOptions: {
"-vb": "20000k",
"-loglevel": "quiet"
},

FFMPEG dumps lots and lots of stuff to stderr. Adding -loglevel panic into the mix quietens that down quite a bit, but also causes node-rtsp-stream to stop functioning as it should when run as a service via systemd.

Application

Stream = require('node-rtsp-stream');
var stream1 = new Stream({
    name: 'ipcam1',
    streamUrl: 'rtsp://192.168.0.131:554/0',
    wsPort: 9991
});

systemd config file

[Unit]
Description=IP Camera viewer
Author=genio
Wants=nginx.service
After=nginx.service

[Service]
ExecStart=/bin/node /home/genio/streamer/app.js
Restart=always
StandardOutput=null
StandardError=syslog
SyslogIdentifier=ipcams
User=genio
Group=genio
Environment=PATH=/home/genio/bin:/usr/bin:/usr/local/bin:/bin
Environment=NODE_ENV=production

[Install]
WantedBy=multi-user.target

If I'm being dumb, please don't hesitate to point that out as I'm dumb quite often. I just don't like filling up /var/log/messages with tons of

frame=   30 fps= 29 q=31.0 size=     486kB time=00:00:00.93 bitrate=4261.3kbits/s

entries.

Thanks,
Chase

I am able to rtsp stream (from IP Camera), but it stopped after approx. 2-3 minutes after displaying
this errorlines.

frame= 4164 fps= 29 q=11.8 size= 13929kB time=00:02:18.73 bitrate= 822.5kbits/
[h264 @ 00000000021f00a0] error while decoding MB 78 26, bytestream -5
[h264 @ 00000000021f00a0] concealing 1491 DC, 1491 AC, 1491 MV errors in I frame

frame= 4184 fps= 28 q=13.0 size= 14001kB time=00:02:19.40 bitrate= 822.8kbits/
frame= 4190 fps= 28 q=12.2 Lsize= 14036kB time=00:02:19.63 bitrate= 823.4kbits/s dup=2094 drop=5 speed=0.941x
video:14036kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%

Hi, I want to create a Dockerfile for the server side.

I have the following in the rtsp-server.js file

`Stream = require('node-rtsp-stream')
stream = new Stream({
"name": 'name',
"streamUrl": 'rtsp://user:[email protected]:xxx',
"wsPort": 8082,

"ffmpegOptions": { // options ffmpeg flags

  '-stats': '', // an option with no neccessary value uses a blank string
  '-r': 30, // options with required values specify the value after the key

}
})`

And in the Dockerfile, I have the following

FROM node:latest RUN mkdir -p /usr/src/app WORKDIR /usr/src/app COPY . /usr/src/app/ RUN apt-get update && apt-get -y install ffmpeg RUN npm install CMD node rtsp-server.js

This is the output I get in the console when I build and run the docker image.

libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 [rtsp @ 0x55b11cf73ec0] UDP timeout, retrying with TCP [rtsp @ 0x55b11cf73ec0] method PAUSE failed: 551 Option not supported [rtsp @ 0x55b11cf73ec0] Could not find codec parameters for stream 0 (Video: h264, none): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, rtsp, from 'rtsp://user:pass@xxxxxxxxxxx': Metadata: title : Media Presentation Duration: N/A, bitrate: N/A Stream #0:0: Video: h264, none, 90k tbr, 90k tbn, 180k tbc Output #0, mpegts, to 'pipe:': Output file #0 does not contain any stream

Is there a way for a client (I'm currently a JsmpegPlayer in react) to dynamically start and stop the server? Essentially I only want the node.js server to run when a client tries to play it, if nobody is trying to play it I want it to turn off. It looks like out of the box there's some handling to kill the stream and it can handle a loss of connection the actual RTSP server, but I was wondering if the websocket on the other side can kill/start the node server depending on some action.

when you want to use several sockets on the same server you must use this option in order to segment each Websocket to diffrent Url.
Correct me if i'm wrong.

Hi, i want some help. I have ip cam and need translate video in browser but i have artifacts.
I try many attempts but nothing help.
My current ffmpeg '-i rtsp://link -f mpegts - '

I've followed the information provided in I believe #37, to bury the ws server inside of a https server with certificates. This works for me but only under one condition. I have to access the server directly at https://IP:9999 to get the certificates in the browser. Then I can use my actual website to open up that wss:// connection. The server doesn't just listen for and accept an attempt to open a wss:// connection at the node-rtsp stream without me doing that. I can't seem to get around it.

VideoStream.prototype.pipeStreamToSocketServer = function() {
  const server = https.createServer({
    cert: fs.readFileSync('./cert/cert.pem'),
    key: fs.readFileSync('./cert/key.pem'),
  }).listen(9999, '0.0.0.0');
  this.wsServer = new ws.Server({
    server
  })
  this.wsServer.on("connection", (socket, request) => {
    return this.onSocketConnect(socket, request)
  })
  this.wsServer.broadcast = function(data, opts) {
    var results
    results = []
    for (let client of this.clients) {
      if (client.readyState === 1) {
        results.push(client.send(data, opts))
      } else {
        results.push(console.log("Error: Client from remoteAddress " + client.remoteAddress + " not connected."))
      }
    }
    return results
  }
  return this.on('camdata', (data) => {
    return this.wsServer.broadcast(data)
  })
}

This piece of code works perfectly only if the browser is already certified. To do that I have to manually visit the IP the server is running on over https://IP:9999, accept the warnings, then leave. Now my hosted website over https:// can connect to that wss:// socket no problem and play video, but first having to certify with an https: visit is a major problem

Hi,
Thanks for this module.

I got this client error when implemented your example

This is my Server side

This is my Client side

Anyone can help me ?

Is there a way to also embed audio to the page with this?

stream.wsServer.close() - as suggested in one of the closed issues, doesn't stop the FFmpeg. The FFmpeg(s) keeps running in the background. How can I close the socket and stop FFmpeg, too?

I have a requirment where I need to shutdown the socket and start again on every request as the stream urls might change. Or let me know if there is an option to stop creating new instance again and again which is giving me port already in use error.

Hello when I execute the following stream of the example:
stream = new Stream({
name: 'name',
streamUrl: 'rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov',
wsPort: 9999,
ffmpegOptions: { // options ffmpeg flags
'-stats': '', // an option with no neccessary value uses a blank string
'-force_fps': 30, // options with required values specify the value after the key
},

I have the follwing output
: No such file or directory

I have been searching in ffmpeg about this problem and didn't find any solution. Any idea on it?
Thanks
Gorka

Hi,
after deploying it to my server (heroku) , it doesn't work because it requires a secure connection wss.
someone managed to solved it?

Dear @kyriesent , I have a issue when setup rtsp stream server.
My code stream-server.js:
Stream = require('node-rtsp-stream');
stream = new Stream({
name: 'name',
streamUrl: 'rtsp://192.168.83.47:8554/faded',
wsPort: 9999
});
That rtsp url is working properly on VLC Player.
When I start server with node stream-server.js, It's failed and show error ouput as below:
ffmpeg version 2.8.6-1ubuntu2 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 60.100 / 56. 60.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 40.101 / 5. 40.101
libavresample 2. 1. 0 / 2. 1. 0
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 2.101 / 1. 2.101
libpostproc 53. 3.100 / 53. 3.100
[rtsp @ 0x2101360] method SETUP failed: 461 Client error
rtsp://192.168.83.47:8554/faded: Protocol not supported

Please help me correct it.
Thanks so much.

Hey,

Could you please push a new version to NPM?

Latest version is v0.0.8 on NPM but it has errors that lead to crash on server.
It's already fixed on master branch but package is not available on NPM.

Thank you!

4c82465#diff-9ff6f769dd6b0da8294d5320b34c5987

Hi,
i am getting a "websocket connection to 'ws://localhost:9999/' failed" when I try to browse it on a browser. what ip/url should i configure for both ffmpeg config and jsmpeg?

Thanks

Hello, is there a live demo of this working? Someone I know was going to use this but they said the video quality was not good. I wonder if it might have been something they were doing so I would like to see someone else's implementation of this.

Hello,

First, thank you for your effort. When I try to run my serverside api.js I always get 'Error: spawn ffmpeg ENOENT' error. I couldn't find a solution. I use Windows pc. Does it matter ? If you can respond I wil be really glad.

Hi I'm a beginner . I want to make reconnection if RTSP back online without restarting node server, i've tried with PM2 and forever but it restart the node server and ffmpeg terminal keeps popping out. Help pls

How to detect programmatically if video is already streaming?
Is there any kind if flag or method which can help me know if video is streaming or there is no video signal from camera?

Commit dd5cffa introduced a new bug where variable i is not defined in videoStream.js file line #103.

this.spawnOptions = [ ...this.frontFlags, "-i", this.url, ...this.midFlags, '-f', 'mpegts', '-codec:v', 'mpeg1video', // additional ffmpeg options go here ...this.additionalFlags, '-' ]