In fact WebRTC is SRTP(secure RTP protocol). Thus we can say that video tag supports RTP(SRTP) indirectly via WebRTC. Therefore to get RTP stream on your Chrome, Firefox or another HTML5 browser, you need a WebRTC server which will deliver the SRTP stream to browser. 2. Websocket. It is TCP based, but with lower latency than HLS.

For live streams, the RTSP stream you request is simply a name of a stream. It doesn't need to refer to a file on the server, the server's RTSP implementation can parse that stream, put together a live graph, and then provide the SDP (description) for that stream name. But, this is of course specific to the way the RTSP server has been implemented.

It's much more complicated than the other answers. But this way, with just one connection to the camera, you could "fork" the same stream simultaneously to several multiprocesses, to the screen, recast it into multicast, write it to disk, etc. Of course, just in the case you would need something like that (otherwise you'd prefer the earlier ...

2019年10月25日 · I want to create a multicast RTP stream using ideally Windows. Linux is also acceptable. I also want to output the SDP for this stream (so that the receiver can use it). The stream source can be a fake signal or a video file. Either is fine. Note that I do not want to read a stream, rather I want to produce one!

2013年4月9日 · Have the robot generate an RTP stream encoded in MP3 format (other formats available) using FFmpeg (the robot is running on a Raspberry Pi running Arch Linux) Captured stream on the client computer using VLC ActiveX control; Found that the SpeechRecognitionEngine has the available methods: recognizer.SetInputToWaveStream()

2012年4月16日 · It doesn't do any RTSP negotiation; eventually this will be pointed at Feng or some other external application to handle RTSP streaming to clients. However, that doesn't explain why nothing can make heads or tails of the RTP stream that libavformat is generating. –

2020年2月4日 · I use OpenCV with ffmpeg (api-preference CAP_FFMPEG) to receive a RTP-Stream and show the video. When I send a short video, the video is played as expected. But when I try to send a second video, the second one is not shown at all. My code:

2017年12月10日 · You also need to guarantee delivery of your frames (because your delivering incremental deltas) so RTP (which is traditionally over UDP) would be challenging. Some thoughts. At the end of the day RTP is just a standardized 12 byte header and packetization rules for compressed media. It's not going to help with latency.

2016年8月9日 · RTP. Do a Google search for c++ programming RTP Packets; Video Compression. Do a Goggle search for both H.26 & MJPEG compression and structure information on stream objects. Audio Compression. Do a Google search for each of AAC(encoding only), G.711 A-Law, G.711 U-Law if you are interested in the audio as well.

2016年11月27日 · @AndreyTurkin I've changed it to use avcodec_parameters_from_context .But without avformat_write_header called repeatedly, VLC does not play the stream (stuck at buffering 0%) and gstreamer lags (more specifically, if avformat_write_header is called on an interval of x amount of time, then gstreamer will repeatedly get stuck for x amount of time (up to ~1s) and then jump ahead).