rtcp | 利用 Python 的 Socket 端口转发，用于远程维护 | Socket library

I have a c++ application that gets the video in RTSP and H264 format from a camera using gstreamer an re-sends the videos using webrtcbin. I have followed the example from this link and I can see the video trough firefox (with the tips suggested in this post), when use VP9 encoding.

The pipeline I have used is:

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

As the title says, I have a problem of getting dtmf digits from b-leg. When I call a queue in callcenter, the two legs can be bridged. But It doesn't have any reaction when I press '*7' on my b-leg telephone.

Here is the dialplan:

Hi there, I am attempting to build a WebRTC client in Android that subscribes to a video feed that is being broadcast using NodeJS and JavaScript.

The broadcaster code can be viewed in its entirety in this lovely article by Gabriel Tanner.

It works beautifully when running it in localhost under the http://localhost:4000/broadcaster.html in Chrome and then visiting my IP Address from another device on the network. I can see the video and it is near real time.

I have tried this using two different webcam devices, both a built-in and a USB webcam but the Android client does not work even though the JavaScript Broadcaster and Client works fine.

After following the tutorial and getting the example to work I decided to try and implement my own Android application for which the entire source code can be viewed right here on my GitHub.

I have followed various tutorials around the place and the issue always stems from attempting to set the remote description which is done with the following bit of code:

As the title indicates using gStreamer or VLC I can get the video to show without issues with the command below

gst-launch-1.0 rtspsrc protocols=tcp location="rtsp://{USER}:{PASSWD}@{URL}:63554/axis-media/media.amp?videocodec=h264" short-header=TRUE ! rtph264depay ! decodebin ! fpsdisplaysink

I can set it as a Network Stream in VLC as below with similar results

rtsp://{USER}:{PASSWD}@{URL}:63554/axis-media/media.amp?stream=0&videocodec=h264

but janus (latest/stable: v0.11.3) configuration as below is unable to show the stream on the webpage

I am trying to send an MP4 video through Pion WebRTC to the browser.

Using FFmpeg, I split it into an Opus OGG stream and an Annex-B H.264 video stream. While the video works fine, the audio keeps cutting in and out. It plays fine for a few seconds, then stops for a second, and continues.

This is the FFmpeg command I use for audio:

I have a mediasoup server setup where I now need recording feature. To accomplish this I am doing the following -
On getting the event to start recording I call this function :

Am new to webRTC and am trying to create a react native app with video calling functionality using this tutorial here as an example to follow https://dipanshkhandelwal.medium.com/video-calling-using-firebase-and-webrtc-14cc2d4afceb

However i keep getting this error on iOS and on android the app just closes once i try to join a call. The error i get on iOS says:

I'm working on WebRTC streaming which streams video from a device to a browser. This streaming works in Chromium browsers just fine but fails in Firefox. There is a failure with the SDP exchange which then halts the rest of the connection (no ICE candidates sent after SDP exchange).

There are some issues with Firefox's answer SDP I've found but I haven't discovered a reason for the issues: SDP mentions VP8 but we use H264 only; m=video 0 has port 0 but typically that's non zero; I typically get an a=inactive line; a=sendrecv should probably be a=recvonly; many other lines are missing (for example, ICE-specific lines)

SDP examples below: