Summary: How do I stream high quality video using WebRTC native?
I have an h264 stream that's 1920x1080 at about 30fps. I can currently stream this from a server on localhost to a native client on localhost just fine.
I wrote a WebRTC server using Google's WebRTC native library. I've written a VideoEncoder and VideoEncoderFactory that takes frames consisting of already encoded data and and broadcasts it over a video track. Using this I can send my h264 stream to the WebRTC server over a pipe and I can see the video stream in a browser.
However, any time something moves the video gets corrupted. It continues to play but is full of artifacts. Eventually I discovered that WebRTC is dropping some of my frames. When I attach a sequentially increasing ID to each frame before I pass it to rtc::AdaptedVideoTrackSource::OnFrame and I log this same ID in webrtc::VideoEncoder::Encode I can see that some of my frames simply disappear.
This kind of makes sense, I'm trying to stream high quality video over something meant for video chat and lowing my framerate fixes the corruption. However, I'm not asking the WebRTC library to do a lot, it's just forwarding already encoded data to a client on localhost. I have a native app that does this fine and I've seen one browser WebRTC client that can do this. Is there a field in the SDP or some configuration change that will allow me to stream my video?
This was the solution How to control bandwidth in WebRTC video call? .
I had heard about changing the offer sdp but dismissed it because I was told that the browser will accept unlimited bandwidth by default and that you'd only need to to this if you want to limit bandwidth. However, adding "b=AS:high number" has fixed all of my problems.
Related
I am trying to loop audio from my Icecast server 24/7.
I have seen examples where people talk about storing their audio files on the EC2 instance or in an S3 bucket.
Do I also need a source client running on my EC2 Instance to be able to stream audio to the server? Or is there a way to play static files from Icecast?
Icecast and SHOUTcast servers work by passing a live audio stream from a source on to the users. You need something to produce a single audio stream in realtime from those source files.
The flow looks something like this:
Basically, you'll need to do everything you would in a normal radio studio, but automated. You'll stream the files from your bucket, play them to a raw audio stream, send that stream to your encoder to be compressed with the codec, and then sent to your streaming servers for distribution.
You can't simply push your audio files as-is to the Icecast server, for a few reasons:
Stream must be realtimeThe server doesn't really know or care about the timing of the stream. It takes the data its given and sends that off to the client. Therefore, if you push data faster than realtime, the server will attempt to deliver it to the client at this faster rate. Some clients will attempt to buffer this fast stream, but most will put backpressure on the stream, causing the TCP window to close, causing the client to eventually get far enough behind that the server drops the connection.
Consistent format is requiredChances are, your source files have varying sample rate, channel count, and even codec. Most clients are unable to take a change in sample rate or channel count mid-stream. I don't know of any client that supports a codec change mid-stream. (Theoretically possible with Ogg and Matroska/WebM, but yeah... not worth messing with.)
Stream should be free of ID3 tags and other file format cruftIf you simply PUT your files directly to your Icecast server, the output stream will contain more than just the audio data. At a minimum, you'd want to remove all that. Depending on your container format, you'll need to deal with timestamps as well.
Solutions
There are a handful of ways to solve this:
Radio automation softwareMany folks simply run something like RadioDJ on cloud-based servers. If you already have a radio station that uses automation, this might be a good solution. It can be expensive though, and not as flexible. You could even go as low as VLC or something for playout, but then you wouldn't have music transitions and what not.
Custom playout script (recommended)I use a browser engine, such as Chromium, and script my channels with normal JavaScript. From there, I take the output stream and pass it off to FFmpeg to encode and send to the streaming servers. This works really well, as I can do all my work in a language everybody knows, and I have easy access to data on cloud-hosted services. I can use the Web Audio API to mix and blend audio based on what's happening in realtime. As an alternative, there is Liquidsoap, but I do not recommend it these days as its language is difficult to deal with and it is not as flexible as a browser engine.
I want to build a live streaming app.
My thought process:
Get the Video/Audio data from the
navigator.mediaDevices.getUserMedia(constraints); [client-streamer]
create rooms using sockets(Socket.IO or WebSockets from flask) [backend]
Send the data in 1 to the room members using sockets.
display the media on the client-side.
Is that correct? How should I do it?
how do I broadcast data to specific room members and not to everyone? (flask)
How to consistently send data from the streamer -> server -> room members. the stream is given from 1 is an object, where is the data?
any other better ideas will be great! thanks.
I need to implement the server-side by myself without help from libraries that will do the work for me.
Implementing a streaming platform is not trivial. Unfortunately, it is not as simple as emitting chunks received from the MediaRecorder with onndatavailable and forwarding them to users using a WebSocket server - this is not scalable nor efficient nor reliable.
Below are some strategies you can try for different types of scenarios:
P2P: If you want to have simple peer-to-peer streaming, you can use WebRTC to achieve that with a simple socket.io server for signaling purposes.
Conference: Here things start to get more complicated. You will need a media server if you want to be somewhat scalable. One approach is to route your stream to the users using an SFU or MCU. This will take care of forwarding/processing media to different peers efficiently.
Broadcast: Here things are also non-trivial. Common WebRTC-based architectures include ingesting the WebRTC stream and forward that to an HLS server which will let your stream chunks available for clients through a CDN, or perform RTP forwarding of the WebRTC stream, convert it to RTMP using something like FFmpeg and deliver it through Youtube Live or Twitch to leverage from their infrastructure.
Be aware that the last 2 items are resource-intensive and will certainly not be cheap to maintain.
Below are some open source projects that could help you along the way:
Janus
MediaSoup
AntMedia
Jitsi
Good luck!
Explaining all this is far beyond the scope of a Stack Overflow answer.
Here are a few hints:
You need to use the MediaRecorder API to capture compressed data from your gUM (getUserMedia) stream. MediaRecorder support is inconsistent between makes and models of browser. though.
It kicks a Blob into its onndatavailable handler every so often.
They're compressed as a webm data stream.
You can push those Blobs to a server with socket.io, and the server can turn around and push them to whatever clients you want to.
Playing the webm on the clients is tricky. You may, on some makes and models of browsers, be able to feed the webm stream to the Media Source API using appendBuffer(). But some browsers cannot consume the webm streams.
These webm streams are useless to a player without all their Blob data in order. You can't just start sending a new client the Blobs of the stream when they sign in; you have to restart the MediaRecorder.
(You may be able to make it work without a MediaRecorder restart if you send the first few k bytes of the stream to each new client before sending the current Blob. Extracting those bytes is an intricate programming job involving the ebml package to parse the webm stream and extract the prologue. I have not proven this concept.)
Because getting all this to work -- originator -- server -- viewer is such a pain in the xxx neck, you may want to investigate using something like mediasoup instead. It uses WebRTC transport rather than socket.io, and works cross-platform.
I manged to run WebRTC peerconnection example, but it is not running on the browser.
I'm trying to find a way to stream both video and audio from browser to my native program.
Is there any way?
It can be done. WebRTC is designed to work in a peer-to-peer manner between two WebRTC agents (typically a Web Browser). Your native program needs to become the second peer.
If you need to rely on open source components a good starting point is:
OpenSSL for the DTLS key exchange.
libsrtp to encrypt the RTP packets.
ffmpeg to decode the PCM audio from the browser (libvpx if you need to do video).
You'll also need to handle the ICE negotiation which requires processing STUN messages. Also extract the media payloads from the RTP packets. All these steps are also after you've determined a signalling method to exchange the SDP offer and answer between you app and the browser.
As you've probably realised starting from scratch it's a major task. There are probably some commercial libraries that will do the job and save you a lot of pain.
If that doesn't scare you and you do still want to make an attempt using open source components this example "may" help. The sample is doing the reverse of what you've asked and is sending a video stream to Chrome rather than receiving an audio stream. The useful aspect is the connection negotiation. The sample program is able to get RTP packets flowing which is often the main problem.
The example is also using Windows Media Foundation which is Windows specific. It also has lots of shortcuts particularly with the RTP and STUN packet processing.
I am planning (using native API), to establish webRTC session between two clients.
My requirement is to establish webRTC session without using PC's audio/video devices (as I plan to have multiple simultaneous webRTC sessions in same PC).
I am currently following this tutorial and want to know where in these files the following things happen:
open the audio / video device
read from audio / video device (capture)
write to audio / video device (play)
close the audio device
Kindly guide me if somebody knows the file-name / function-name where I need to look for above 4 points.
I want to create server and client applications for controlling electronic device through network.
Server application should stream video from web camera (RGB, 320 * 240) with some information about current state of device and control microcontroller device via RS-232.
Client application should allow to adjust controlling process and show video from web camera and some information about device.
What i have done: I use Qt framework in oder to create GUI of both applications and TCP sockets, also I use OpenCV to get images from camera. Server application compresses images in JPEG, add some information about device (~ 60 bytes) and send to client.
Problem: In Local network everything works fine, but working through Internet I can get only about 15 fps because JPEG images are too large. With stronger JPEG compression I can get suitable fps, but with bad image quality. So I wonder is there any better ways to stream video with some extra information about current state of device? Maybe with FFMPEG or something else.
Thanks for your replies and sorry for my english!