I have a piece of software running on a node in my network that generates RTP streams carried over UDP/IP. Those streams contain streaming data, but not in any standard audio/video format (like H.264, etc.). I would like to have a simple Web app that can hook into these streams, decode the payloads appropriately, and display their contents. I understand that it isn't possible to have direct access to a UDP socket from a browser.
Is there a way to, using JavaScript/HTML5, to read an arbitrary RTP stream (i.e. given a UDP port number to receive the data from)? The software that sends the stream does not implement any of the signaling protocols specified by WebRTC, and I'm unable to change it. I would like to just be able to get at the RTP packet contents; I can handle the decoding and display without much issue.
As far as I know, there is nothing in the set of WebRTC APIs that will allow you to do this. As you have pointed out, there also isn't a direct programmatic way to handle UDP packets in-browser.
You can use Canvas and the Web Audio API to effectively playback arbitrary video, but this takes a ton of CPU. The MediaSource extensions can be used to run data through the browser's codec, but you still have to get the data somehow.
I think the best solution in your case is to make these connections server-side and use something like FFmpeg to output a stream in a codec and container that your browser can handle, and simply play back in a video element. Then, you can connect to whatever you want. I have done similar projects with Node.js which make it very easy to pipe streams through, and on out to the browser.
Another alternative is to use WASM and create your own player for your stream. It's pretty incredible technology of these recent years > 2014. Also as stated by #Brad, WebRTC doesn't support what you need even as of this year 2020.
Related
My goal is to create a web-radio station hosted on localhost using JavaScript with node.js server for the backend part and HTML/CSS/JavaScript for the frontend part. My radio is going to be simple, and the goal is to create a server which is going to constantly broadcast a single song (or many songs using the .mp3 format) so they can be consumable from each client connected.
The streaming part is not so hard. The part I am struggling so far, is to achieve the "broadcast" transmission simultaneously to all consumers. The image below explains better my thoughts:
Does anyone have a code example so I can easily understand it and at the same time implement it for my project's use?
Has anyone faced a similar situation like this?
This is exactly what SHOUTcast/Icecast and compatible servers do. You basically just need to copy the audio data to each client.
You can use normal HTTP for the streaming protocol. The clients don't need to know or care that they're streaming live. A simple audio element works:
<audio src="https://stream.example.com/some-stream" preload="none" controls></audio>
On the server-side, you do need to ensure that you're sending a stream aligned in such a way that the client can just pick right up and play it. The good news is that for raw MP3 and ADTS (which normally wraps AAC) streams, all the data they need for playback is in each frame, so you can just "needle drop", stream from an arbitrary position, and the client will figure it out.
Inside your Node.js app, it will look a bit like this:
Input Stream -> Buffer(s) -> Clients HTTP Responses
And in fact, you can even drop the buffer part if you want. As data comes in from the codec, you can just write it to all the clients. The buffer is useful to ensure a fast starting playback. Most clients are going to need some buffered data to sniff stream type and compatibility, as well as to fill their own playback buffer for smooth streaming.
That's really all there is to it!
Bringing this over from softwareengineering. Was told this question may be better for stackoverflow.
I am sending a video stream of data to another peer and want to reassemble that data and make that stream the source of a video element. I record the data using the npm package RecordRTC and am getting a Blob of data every 1s.
I send it over an WebRTC Data Channel and initially tried to reassemble the data using the MediaSource API but turns out that MediaSource doesn't support data with a mimetype of video/webm;codecs=vp8,pcm. Are there any thoughts on how to reassemble this stream? Is it possible to modify the MediaSource API?
My only requirement of this stream of data is that the audio be encoded with pcm but if you have any thoughts or questions please let me know!
P.S. I thought opinion based questioned weren't for stackoverflow so that's why I posted there first.
The easiest way to handle this is to proxy the stream through a server where you can return the stream as an HTTP response. Then, you can do something as simple as:
<video src="https://example.com/your-stream"></video>
The downside of course is that now you have to cover the bandwidth cost, since the connection is no longer peer-to-peer.
What would be nice is if you could use a Service Worker and have it return a faked HTTP response from the data you're receiving from the peer. Unfortunately, the browser developers have crippled the Service Worker standards by disabling it if the user reloads the page, or uses privacy modes. (It seems that they assumed Service Workers were only useful for caching.)
Also, a note on WebRTC... what you're doing is fine. You don't want to use the normal WebRTC media streams, as not only are they lossy compressed, but they will drop segments to prioritize staying realtime over quality. This doesn't sound like what you want.
I've been wondering this - is the raw mediastream returned from something like getusermedia what format is that in?
The MediaStream is the raw data, but it isn't accessible directly. If you attach the MediaStream to a Web Audio API graph, whatever format the sound card captured in is converted to 32-bit floating point PCM. At this point, you can use a script processor node to capture the raw PCM data.
I'm looking at a way to implement video encoder using web browser. Youtube and Facebook already allow you to go live directly from the web browser. I'm wondering how do they do that?
There are a couple of solutions I've researched:
Using web socket: using web browser to encode the video (using mediarecorder api) and push the encoded video to the server to be broadcast.
Using WebRTC: web browser as a WebRTC peer and another server as the other end to receive the stream and re-broadcast (transcode) using other means (rtmp, hls).
Is there any other tech to implement this that those guys (YouTube, Facebook) are using? Or they also use one of these things?
Thanks
WebRTCHacks has a "how does youtube use webrtc" post here which examines some of the technical details of their implementation.
In addition one of their engineers gave a Talk at WebRTC Boston describing the system which is available on Youtube
Correct, you've hit on two ways to do this. (Note that for the MediaRecorder method, you can use any other method to get the data to the server. Web Sockets is one way... so is a regular HTTP PUT of segments. Or, you could even use a data channel of a WebRTC connection to the server.)
Pretty much everyone uses the WebRTC method, as there are some nice built-in benefits:
Low latency (at the cost of some quality)
Dynamic bitrate
Well-optimized on the client
Able to automatically scale output if there are not enough system resources to continue encoding at a higher frame size
The downsides of the WebRTC method:
Ridiculously complicated stack to maintain server-side.
Lower quality (due to emphasis on low latency, BUT you can tweak this by fiddling with the SDP yourself)
If you go the WebRTC route, consider gstreamer. If you want to go the Web Socket route, I've written a proxy to receive the data and send it off to FFmpeg to be copied over to RTMP. You can find it here: https://github.com/fbsamples/Canvas-Streaming-Example
What are the steps and means of capturing microphone audio stream through HTML5 / Javascript (no flash) and then sending it to an already set up icecast server?
The solution must be solely browser/web-based, no additional software.
The server is on Rails 5.0.0.1.
Where should I start?
I'm struggling to find any relevant info online as everything talks about uploading/recording audio files as complete files, not streams.
The solution must be solely browser/web-based, no additional software.
That's not possible at the moment, because there is no way to make a streaming HTTP PUT request. That is, to make an HTTP request either via XHR or Fetch, the request body has to be immutable.
There is a new standard of ReadableStream which will be available any day now, allowing you to make a ReadableStream (your encoded audio) and PUT it to a server. Once that is available, this is possible.
The server is on Rails 5.0.0.1.
Not sure why you mention that since you said it has to all live in the browser. In any case, let me tell you how I implemented this, with a server-side proxy.
Client-side, you need to capture the audio with getUserMedia(). (Be sure to use adapter.js, as the spec has recently changed and you don't want to have to deal with the constraints object changes manually.) Once you have that, you can either send PCM samples (I'd recommend reducing from 32-bit float to 16-bit signed first), or you can use a codec like AAC or MP3 client-side. If you do the codec in the client, you're only going to be able to send one or two streams. If you do the codec server-side, you can derive as many as you want, due to CPU requirements, but you will take more bandwidth depending on the streams you're deriving.
For the codec on the client, you can either use MediaRecorder, or a codec compiled to JavaScript via emscripten (or similar). Aurora.js has some codecs for decoding, but I believe at least one of the codecs in there also had encoding.
To get the data to the server, you need a binary web socket.
On the server side, if you're keeping the codecs there, FFmpeg makes this easy.
Once you have the encoded audio, you need to make your HTTP PUT request to Icecast or similar, as well as update the metadata either out-of-band, or muxed into the container that you're PUT-ing to the server.
Self Promotion: If you don't want to do all of that yourself, I have some code you can license from me called the AudioPump Web Encoder, which does exactly what you ask for. You can modify it for your needs, take components and embed them into your project, etc. E-mail me at brad#audiopump.co if you're interested.
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