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Hi everyone, I am trying to set up conference app and wondering about the following scenario: Let's say we have a room with 1 publisher publishing via WebRTC and 100 watchers watching the publisher stream via WebRTC. Publisher have a good network connection, but some of watchers are not. As far as I know, WebRTC have built-in adaptive bitrate algorithm which forces publisher to decrease quality of the video if the peer have poor network connection. Will the fact that some of watchers have poor network connection (high ping, low bandwidth) lead to a decrease in the quality of the stream which publisher sends to the server? So all watchers will see low-quality video even if most of them have pretty good network connection?

Hi Guys We've just prepared first version of WebRTC Samples so that users/developers can experience/play WebRTC Samples with Ant Media Server easily. There are 3 samples for now and we're going to add a new sample in each week. The full draft list is below. If you have any idea for samples or feedback for the samples or want to take part in preparing an easy to use webrtc sample, let me know. We can also discuss in the community hours tomorrow. Here is the webrtc samples link -> https://antmedia.io/webrtc-samples/ Cheers Oguz Full WebRTC Samples List(Draft) - PARTIALLY DONE: Publish and Play in the Same Page(Needs Refactor)- DONE: Publish and Play in different pages - DONE: Audio Only Publish and Play - DONE: Virtual Background - DONE: Data Only Publish and Play in different pages - RTMP Publishing and Playback - ScreenSharing Publish- Canvas Publishing- Show the Latency in Publish and Play - Change the video tracks or audio tracks on the fly- Conferencing- Multitrack Streaming- SRT Publishing and Playback- RTSP(IP Camera Streaming) and Playback- PTZ Functions for IP Camera- White Board Publish & Play- Embedded Player- Publish with User and Screen with high performance(Insertable Streams)- Transmit File with Data Publishing- Change bandwidth on the fly- Change Camera Resolution- Change Camera and Microphone- Constraints and Statistics - Session Restored Sample- requestVideoFrameCallback- Web Audio output as input to peer connection- Peer connection as input to Web Audio - Use different TURN/STUN Servers- Support SVC before the call - Check bandwidth and connectivity (test tool)- Insertable Streams: - Video Processing - Audio Processing

Hello, everyone. I wonder if it is possible to play the stream which I am currently publishing from the server via WebRTC (let's say for diagnostic sake) instead of just playing the stream from webcam directly? I'm trying to call webRTCAdaptor.play() passing my own streamId in publish_started callback, but getting "notSetRemoteDescription" error which states: "webrtc_adaptor.js:908 set remote description is failed with error: InvalidAccessError: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote offer sdp: The order of m-lines in subsequent offer doesn't match order from previous offer/answer." Am I doing something wrong or there is no such possibility at all?

I build a server (Google Cloud Compute VM) with Ant Media Server Community Edition and when I publish a stream with /WebRTCApp/, and playlist /WebRTCApp/play.html?id=<myStreamID>, the video is so blurry even if I increased video bit rate up to 320,000 kbps. I need to demo for my company that Ant Media Server is feasible for the company's demand.

Hi Guys, I'm wondering what you are thinking about adding WebRTC Support to FFmpeg. Please join the survey below https://forms.gle/QSgAMGLumPC6YKmK7 Thank you Regards, A. Oguz