webrtc | A pure Rust implementation of WebRTC

I'm building a C++ GStreamer project with CMake which depends on GStreamer, GLIB, Libsoup and json-glib. I'm new to CMake and having trouble setting up my project. I've managed to include many of the dependencies but some seem to remain unresolved even though they are part of GStreamer. All GStreamer methods and types are resolved with the exception of SDP and WebRTC. They are, to my understanding, part of GStreamer and are also located inside of the directory which GMake correctly "finds".

These are the errors that are occurring when trying to build the project.

I am trying to display received WebRTC frames using OpenCV imshow(). WebRTC delivers frames as objects of webrtc::VideoFrame and in my case, I can access webrtc::I420Buffer from it. Now my question is how do I convert the data in webrtc::I420Buffer to cv::Mat, so that I can give it to imshow()?

Thsi is what the definition of webrtc::I420Buffer looks like

After upgrading to android 12, the application is not compiling. It shows

"Manifest merger failed with multiple errors, see logs"

Error showing in Merged manifest:

Merging Errors: Error: android:exported needs to be explicitly specified for . Apps targeting Android 12 and higher are required to specify an explicit value for android:exported when the corresponding component has an intent filter defined. See https://developer.android.com/guide/topics/manifest/activity-element#exported for details. main manifest (this file)

I have set all the activity with android:exported="false". But it is still showing this issue.

My manifest file:

I am trying to build a webrtc flutter app on my m1 macbook air. But I got different issues both on android and ios. Latest one ^0.8.2 has error on both then ^0.7.0+hotfix.1 demo demo only works for android.

On iOS part 'Libyuv''s deployment target is set to 8.0 but min deployment target is 9.0 occurs. I set the deployment target above 10 then it still happens.

I have been reading a bit about WebRTC, and I'm not getting why we need a Turn Server if only 1 peer is using Symmetric NAT, and the other is using neither Symmetric nor Port Restricted NAT, so let’s say A is using Full Cone NAT, B is using Symmetric NAT:

1. STUN SERVER will send the correct IP address of B to A, and the correct IP + Port address of A to B.

2. A tries to connect to B (now A will be able to accept messages from B since it’s in the Dest Address Column).

3. B tries to connect to A, which will allow requests from A going to B (ofc A needs to update the port to the one received from B instead of the Sdp).

am I missing something, or is this correct (and implemented), or is this too complicated to be implemented?

And if this is correct, then theoretically, if I’m peer A and I'm using Full Cone NAT, any peer B can connect to me (as long as I send the connection request first), without needing a TURN server.

Thanks

I have a task, but I can't seem to get it done. I've created a very simple WebRTC stream on a Raspberry Pi which will function as a videochat-camera. With ionic I made a simple mobile application which can display my WebRTC stream when the phone is connected to the same network. This all works.

So right now I have my own local stream which shows on my app. I now want to be able to broadcast this stream from my phone to a live server, so other people can spectate it.

I know how to create a NodeJS server which deploys my webcam with the 'getUserMedia' function. But I want to 'push' my WebRTC stream to a live server so I can retrieve a public URL for it.

Is there a way to push my local Websocket to a live environment? I'm using a local RTCPeerConnection to create a MediaStream object

First, I want to mention that I am very new to WebRTC, so any advice would be very helpful.

Currently I am using aiortc library to build my own WebRTC app. Here is what I am trying to do.

I have 2 peers, one is web browser, which is written in javascript, and another one is python script, which is working as signaling server and peer at the same time. So If you access to my web page, you will send video frame to server and then the server will make modification of that then send it back.

So I finished testing my app on LAN environment and everything worked as I expected. But once I deployed my app to remote server (Google cloud run) , I encountered Ice connection state failing issue. And gets this log on remote server.

enter image description here

(I think it is due to disconnection between peers, not low memory problem. I tried with 16GB RAM and 4 cpus and still didn't work)

Then, I dig into more information, and found that TURN/STUN server is necessary to build WebRTC app over Internet. So I added google STUN server to my RTCPeerConnection like this. [{'urls': 'stun:stun.l.google.com:19302'}, {'urls': 'stun:stun1.l.google.com:19302'}, {'urls': 'stun:stun2.l.google.com:19302'}] (I added both side on javascript and python because both side is working as peer) Unfortunately, it still didn't work.

Now, I am planning to build my own TURN server, but I am afraid if TURN server wouldn't solve this problem. So I would like to have any advice from you since I am quite stuck within my situation.

p.s I have done SSL encryption.(So GetUserMedia is working fine)

Sdp details(Offer/Answer):

SDP

Offer

I followed this post on Stackoverflow to disable Firefox WebDriver detection.

Launch Geckodriver:

It has been discussed many times on Stackoverflow that by default WebRTC technology leaks your real IP even if your using a proxy to browse the web. What I haven't seen discussed is whether this requires the end user to click a button to enable this kind of leak or whether the leak occurs regardless of any action taken by the user.

For example, when you go to Express VPN they require you press a button to test for WebRTC leak. My question is - is this done for privacy reasons or somehow the button activates WebRTC tech so it can leak your IP?

In other words, assuming you never need to use WebRTC tech (just browser a blog or eCommerce shop) and all you do is click a few links - can a website still detect your real IP through WebRTC?

Thanks