Michel Promonet

Hi, [MPEG2TransportStreamFromESSource::maxInputESFrameSize](https://github.com/rgaufman/live555/blob/master/liveMedia/MPEG2TransportStreamFromESSource.cpp#L76) is set to 100000. If you have bigger image, you can try to increase it. Best Regards, Michel.

Hi luo199393, The backend do not need to know backend url to use, it is just a catalog of url. Building you own javascript, it is easy to manage the...

Hi, You may submit a PR with your needs. However as I told you can develop your own server that manage url list and call webrtc-streamer to connect to it....

Hi ekavet, Did you look to CPU, memory consumpsion ? Did you test to connect using another RTSP client, like VLC, ffplay or whatever. Best Regards, Michel.

Hi, WebRTC is 2-way connection, you can look to https://github.com/mpromonet/webrtc-streamer-html/blob/master/sendstream.html that was a try to send a stream from web browser to webrtc-streamer. The stream is notified in [PeerConnectionObserver ::OnAddStream](https://github.com/mpromonet/webrtc-streamer/blob/master/inc/PeerConnectionManager.h#L213)....

Hi Erwan, It needs to update the server side code to get the AudioSink and implement it. Best Regards, Michel.

Hi Erwan, Don't be afraid, this is a really small project, the tricky part is probably to manage multiple session, and choose how to output the audio (using WebRTC SDK...

Hi daltonjoi, Still now, I did not look at this subject. Feel free to implement it and submit pull request. Best Regards, Michel.

Hi daltonjoi, Did you tried http://recordrtc.org/RecordRTC.html ? Regards, Michel.

Hi @ricardoboss Please read the [CONTRIBUTING.md](https://github.com/mpromonet/v4l2rtspserver/blob/master/.github/CONTRIBUTING.md), it is really short. Best Regards, Michel.