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.
Related
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.
I'm developing a web application that captures video from a webcam and saves the stream to Amazon Kinesis.
The first approach I came up with is getUserMedia / mediaRecorder / XMLHttpRequest which posts chunked MKV to my unix server (not AWS), where simple PHP backend proxies that traffic to Kinesis with putMedia.
This should work, but all media streams from user will go through my server which could become a bottleneck. As far as I know, it's not possible to post chunked mkv to Amazon directly from browser due to cross-origin problems. Correct me if I'm wrong or there's a solution for this.
Another thing that I feel I'm missing - is WebRTC. XHR feels a little bit like a legacy in 2019 for streaming media. But if I want this to work, I will need a stack of three servers: webrtc server to establish connection, webrtc->rtsp proxy, and Kinesis gstreamer plugin, which grabs rtsp stream and pushes it to Kinesis. It looks a bit overcomplicated, and media traffic still runs through my server. Or maybe there is a better approach?
I need a suggestion on how to make better architecture for my app. I feel the best solution would be direct webrtc connection with some amazon service, which proxies stream to kinesis. Is it possible?
Thanks!
I was looking into this also for general education/research purpose. The closest example is featured on AWS blog.
And this is github repo. From the README.md
If the source is a sequence of buffered webcam frames, the browser client posts frame data to an API Gateway - Lambda Proxy endpoint, triggering the lambda/WebApi/frame-converter function. This function uses FFmpeg to construct a short MKV fragment out of the image frame sequence. For details on how this API request is executed, see the function-specific documentation.
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 searching for a way to stream video with webRTC. I am very new to webRTC. I have seen a lot of applications on the web that have p2p video chat. The tutorials I follow explain how WebRTC working for the client, but they do not show what use a backend script. And that's exactly what I'm looking for, a backend script (preferably node.js) that ensures that I can stream live video (getUsersMedia) to the client.
Marnix Bouhuis
Its really simple to get started, checkout a simple demo here
1.You need a WebRTC supported browser. Chrome and firefox are best at now
A signalling server to exchange a media options. SocketIO with Nodejs
TURN and STUN server to solve NAT and Symmetric NAT (Only if you public)
MCU, if you want to limit the bandwidth usage. It give flexibility to a star network rather than mesh network in normal p2p