rtp | fork github.com/wernerd/GoRTP

I'm using ffmpeg to generate an RTP stream for testing purposes, but I need to change the RTP payload type to 107. Currently, it's 96.

This is the command I use to generate the stream:

The media team is running some ad campaigns that I need to report on. The reports need to be based on either cities or city groups. A small example of the campaign names are:

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 am trying to record RTP stream with video and audio to HLS. Using GStreamer and nodejs. I get WARNING: erroneous pipeline: no element "fdkaacenc". I am using macOS with M1 CPU and all plugins base, good, bad, and ugly are installed.

The whole command I am trying to execute is this

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:

Here is my code. Basically I am trying to stream android screen to vlc media player using rtsp somehow I have managed to stream video perfectly but audio is not playing. I am using AMRNB encoding method to send stream on Local Socket. I can see that there is audio track by using log also it is being send but don't know why it is not getting played.

I am receiving h264 frames over a serial link, trying to play them with gstreamer. I set the caps to gst_caps_from_string("video/x-h264"), and it seems to accept them (if I use other caps, e.g. application/x-rtp, then the gstreamer log output complains about incompatible caps).

More specifically, I use the following elements: appsrc ! h264parse ! rtph264pay, and it seems like h264parse is the one not being happy.

When I pass (through appsrc) a frame that I get as a byte[] of length 8018, I get the following log output:

I am trying to start a capture of the webcam, transcode it and output it to RTSP but I cannot view the stream when I view: rtsp://127.0.0.1:53211

I get a log in my VLC player saying:

I have the following function being called in a ViewController that passes in data to build a list of transactions. Each time the page refreshes, this is called and new data is shown. All is good, except for one remaining frustrating bug: when the new data is loaded the previous balancedOwedAmountLabel.text still shows the previous amount.

So on each refresh, the text is written on top of the previous text each time. This results in a jumbled mess of text each time the page is refreshed. It would be expected to have the previous text removed, and replaced with the new text that I determine.

I've tried manually setting the balancedOwedAmountLabel.text = nil to simply remove it and then reset it each time the data is retrieved and painted, but it won't work. I've also tried removing the subview entirely via balancedOwedAmountLabel.removeFromSuperview() and that doesn't seem to do the trick either.

Since I'm not using a Storyboard I can't connect IBOutlets so I'm having to resort to figuring out how to get around this problem programmatically. How do I go about solving this with the following code?