I've been looking for days a solution to convert a video stream (video/webcam) from a web browser to a backend RTSP stream.
All I could find was backwards, from RTSP to WebSockets (to display in a web page).
I want the user to choose from a web browser client, a video or webcam locally and then send it to a nodejs server.
Opening a webcam and send the chunks via websockets, seems easy, but how to "convert" these chunks of video to a RTSP server, and then connect via VLC to see the stream ?
Thank you in advance.
Related
I am implementing secure HLS streaming for my educational website like udemy and other platform , the content will be streamed from Amazon s3. I have already implemented a demo where I am streaming from my local server with encrypted HLS file while getting the chunks from server the enc.key file is also available in the network tab, So is this safe to have it or some other strategies like DRM need to apply.
everyone .I need some help for some stream problems, here is my to do list.
I want stream my camera stream，microphone stream via broswer to a media server, and a python server
need to pull this stream to do some asr, tts things. After that ,It generate a reply stream and push back to media server ,and the broswer pull this stream. My plan is like this.
broswer stream to rtmp server use rtmp protocol
python server pull this stream use rtmp
python server push its reply to rtmp server
broswer pull it
but, my question is how can broswer stream rtmp to rtmp server? cause As far as I know I can just use webrtc to do this in broswer?
so the process maybe like this?
broswer stream to janus(or other webrtc server) use webrtc
python use webrtc to pull this stream
python server push its reply to janus
broswer pull this stream use webrtc
but, I'm not sure whether step 2 or step 3 can be done, cause I don't know how to use webrtc in a python environment and without a broswer and use my own stream(not a camera stream)
or can janus convert and push webrtc stream to rtmp server ?
Any helps will be appraciated, thanks.
I know that with mediasoup you can send the camera to the server with webrtc. Then you can use the server to retransmit it to your algos with rtp and lastly send the response back to the client using webrtc again.
I am trying to build a live-streaming web application. I am using a Java FFmpeg wrapper and using it to stream my webcam feed live to AWS MediaLive, which channels to MediaPackage to transcode and send the feed back to the website. However, while this would work locally, I will eventually have to host this application on something like AWS EC2, and detect a user's webcam and audio through the server (with FFmpeg being installed on the EC2 instance).
How can I do this? Services like Discord ask permission for the audio device on the browser and access it. How do websites like Discord achieve this?
I'm working on a project and I need to send an audio stream from a webpage (through javascript) to a server written in C++. Is this possible? How can I do this? I was thinking on use WebRTC and a WebRTC library for C++ but I don't really know hoy to achieve this.
In general I need some king of webserver in C++, that allows me to send/recieve audio stream and json and works with multiple web clients.
I have worked with Socket.io and once I coded a webserver in Java EE 7, with those I was able to send/recieve json from the webpage but I don't really know if I can send audio stream via websocket or json.
The question (or implementation in answer to the question) really consists of two parts, which are:
How to send audio stream from browser in Javascript
How to receive audio stream on server in C/C++
This is because sending data over the network only loosely couples the client and the server when they use the same protocol. You could write a server in C++, then write two different clients that communicate with it, one in Javascript, then also a desktop app written in Java.
Javascript on Client Side
For the client side, sending audio from the browser in Javascript should follow the normal libraries available for WebRTC; the WebRTC site has some useful information on this, including a video streaming example here (https://webrtc.github.io/samples/)
Some of the links which might be of interest on that page:
Audio-only getUserMedia() output to local audio element
Stream from a video element to a video element
There are some StackOverflow answers already about WebRTC and audio in javascript, here are a couple, these (and libraries) will be more plentiful than C++ questions on the topic:
Sending video and audio stream to server
Sending a MediaStream to host Server with WebRTC after it is captured by getUserMedia
For the C++ Server:
The WebRTC site has a link to the Native API for the libraries here (https://webrtc.org/native-code/native-apis/) and an excellent simple example of a peer connection WebRTC server using them is here (https://webrtc.googlesource.com/src/+/master/examples/peerconnection). It also has an implementation of a C++ client there, which may help in testing the server to get it working first, or see the general principles.
I am capturing audio data using getUserMedia() and I want to send it to my server so I can save it as a Blob in a MySQL field.
This is all I am trying to do. I have made several attempts to do this using WebRTC, but I don't even know at this point if this is right or even the best way to do this.
Can anybody help me?
Here is the code I am using to capture audio from the microphone:
navigator.getUserMedia({
video:false,
audio:true,
},function(mediaStream){
// output mediaStream to speakers:
var mediaStreamSource=audioContext.createMediaStreamSource(mediaStream);
mediaStreamSource.connect(audioContext.destintion);
// send mediaStream to server:
// WebRTC code? not sure about this...
var RTCconfig={};
var conn=new RTCPeerConnection(RTCconfig);
// ???
},function(error){
console.log('getUserMedia() fail.');
console.log(error);
});
How can I send this mediaStream up to the server?
After Googling around I've been looking into WebRTC, but this seems to be for just peer to peer communication - actually, now I'm looking into this more, I think this is the way to go. It seems to be the way to communicate from the client's browser up to the host webserver, but nothing I try even comes close to working.
I've been going through the W3C documentation (which I am finding way too abstract), and I've been going thru this article on HTML5 Rocks (which is bringing up more questions than answers). Apparently I need a signalling method, can anyone advise which signalling method is best for sending mediaStreams, XHR, XMPP, SIP, Socket.io or something else?
What will I need on the server to support the receiving of WebRTC? My web server is running a basic LAMP stack.
Also, is it best to wait until the mediaStream is finished recording before I send it up to the server, or is it better to send the mediaStream as its being recorded? I want to know if I am going about doing this the right way. I have written file uploaders in javascript and HTML5, but uploading one of these mediaStreams seems hellishly more complicated and I'm not sure if I am approaching it right.
Any help on this would be greatly appreciated.
You cannot upload the live stream itself while it is running. This is because it is a LIVE stream.
So, this leaves you with a handful options.
Record the audio stream using one of the many recorders out there RecordRTC works fairly well. Wait until the stream is completed and then upload the file.
Send smaller chuncks of recorded audio with a timer and merge them again server side. This is an example of this
Send the audio packets as they occur over websockets to your server so that you can manipulate and merge them there. My version of RecordRTC does this.
Make an actual peer connection with your server so it can grab the raw rtp stream and you can record the stream using some lower level code. This can easily be done with the Janus-Gateway.
As for waiting to send the stream vs sending it in chunks, it all depends on how long you are recording. If it is for a longer period of time, I would say sending the recording in chunks or actively sending audio packets over websockets is a better solution as uploading and storing larger audio files from the client side can be arduous for the client.
Firefox actually has a its own solution for recording but it is not supported in chrome so it may not work in your situation.
As an aside, the signalling method mentioned is for session build/destroy and really has nothing to do with the media itself. You would only really worry about this if you were using possibly solution number 4 shown above.
A good API for you would be MediaRecorder API but it is less supported than the Web Audio API, so you can do it using a ScriptNode or use Recorder.js (or base on it to build your own scriptnode).
WebRTC is design as peer-to-peer, but the peer could be a browser and a server. So it's definitely possible to push the stream by WebRTC to a server, then record the stream as a file.
The stream flow is:
Chrome ----WebRTC---> Server ---record---> FLV/MP4
There are lots of servers, like SRS, janus or mediasoup to accept WebRTC stream. Please note that you might need to covert the WebRTC(H.264+Opus) to MP4(H.264+AAC), or just choose SRS which supports this feature.
yes it is possible to send MediaStream to your server, but the only way you can achieve is by going through WebSocket which enable client browser to send data to your server in real time connection. so i recommend you to use websocket