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.
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I don’t know if I can say “I’m sorry for ask” but I spent more than a week looking for a solution without success. I have a Jetson Nano and with OpenCV I get and process an image at 4fps, I need to send this video to a web server to allow the client connected to the server get the video. Everything need to be written in C++.
Because a need a low latency I did test with GStreamer and WebRTC without success. I don’t have any web server ready, so I can use any implementation.
Anyone know where I can find some example implementation with this schema?
You can use mediasoup to send data to the server to then send the stream with rtp to another endpoint like gstreamer or ffmpeg.
Here is a recording project where data is sent from the browser -> server -> gstreamer -> file.
Mediasoup is written in c++ and has a wrapper for js.
I had similar problem and used such example from GStreamer WebRTC official repo. It's written in Python for Janus Gateway video rooms but I think it can be easily rewritten in C++ as you need.
In the code for OpenCV, I used V4L2Loopback as a virtual output device to be used as input for GStreamer WebRTC example.
I hope such approach may help you.
I think no need to send it to a Web Server. In Gstreamer examples [https://github.com/GStreamer/gst-examples]. The SendOnly example sends a video to a Web Client Using WebRTC. You can modify it to send an OpenCV mat.
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.
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.
I have an audio processing server, and I'd like to be able to connect to it via WebRTC.
The native library from Google seems suitable for that (from looking at the peerconnection example): https://webrtc.org/native-code/native-apis/
But the library relies too much on the audio devices: it opens them behind the scenes. I've managed to grab the incoming audio by appending my own AudioTrackSinkInterface to the stream, but haven't yet found how to inject the audio into the outbound stream. And these hacks don't avoid opening the devices anyway.
How to do it cleanly?
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!