I'm implementing an RTSP server which reads a video H264 file stream and makes it available under RTSP protocol.
The idea is this:
The server creates a pipeline using video1.mp4 as videosrc and then, when video ends and the EOS event is generated, it has to switch filesrc to video2.mp4 and restart streaming automatically. When video2.mp4 ends it switches back to video1.mp4 again. The client is supposed to never stop receiving a video stream in this way.
Is it possible to obtain a server behavior like this?
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I can't figure out how to call rtsp methods with headers, for understanding:
I have an rtsp player on qt, I want to add the rtsp playback speed functionality, many vendors do this by sending a PLAY request with a Speed header,
but I don't understand how I can send this request from gstreamer.
You set playback rate for the pipeline via a GstSegemnt. E.g. when issueing a seek command. See https://gstreamer.freedesktop.org/documentation/gstreamer/gstsegment.html#GstSegment.
rtspsrc should then take care of sending the required header info to the RTSP server when initiating the session.
I have a client in gstreamer which is sending his audio to another client. I want the sender client to be able to change between different audio sources (Microphones in the device) dynamically.
I am using wasapisrc for getting the audio stream from the microphone
I tried changing the device property of the wasapisrc (when the connection is established) but it didn't work.
you can use input-selector element to switch between input sources
https://gstreamer.freedesktop.org/documentation/coreelements/input-selector.html
Summary: How do I stream high quality video using WebRTC native?
I have an h264 stream that's 1920x1080 at about 30fps. I can currently stream this from a server on localhost to a native client on localhost just fine.
I wrote a WebRTC server using Google's WebRTC native library. I've written a VideoEncoder and VideoEncoderFactory that takes frames consisting of already encoded data and and broadcasts it over a video track. Using this I can send my h264 stream to the WebRTC server over a pipe and I can see the video stream in a browser.
However, any time something moves the video gets corrupted. It continues to play but is full of artifacts. Eventually I discovered that WebRTC is dropping some of my frames. When I attach a sequentially increasing ID to each frame before I pass it to rtc::AdaptedVideoTrackSource::OnFrame and I log this same ID in webrtc::VideoEncoder::Encode I can see that some of my frames simply disappear.
This kind of makes sense, I'm trying to stream high quality video over something meant for video chat and lowing my framerate fixes the corruption. However, I'm not asking the WebRTC library to do a lot, it's just forwarding already encoded data to a client on localhost. I have a native app that does this fine and I've seen one browser WebRTC client that can do this. Is there a field in the SDP or some configuration change that will allow me to stream my video?
This was the solution How to control bandwidth in WebRTC video call? .
I had heard about changing the offer sdp but dismissed it because I was told that the browser will accept unlimited bandwidth by default and that you'd only need to to this if you want to limit bandwidth. However, adding "b=AS:high number" has fixed all of my problems.
I want to receive rtsp stream using gstreamer I knw rtspsrc can be used for this purpose but the problem is that it only receives it as a client but in my case i have a ffmpeg application which streams the video as a client and waits for a server to connect with it before streaming. So i want gstreamer to act as server and receive the stream from ffmpeg
I haven't used it myself, but I believe there is a separate package for RTSP server functionality. In Debian based systems it should be under something like:
libgstrtspserver-0.10-0
I am developing a client program which will display the media captured from IP camera. So I want to whether the RTP packets using RTSP protocol contain both audio and video if contains both how should I extract it?
RTSP stream does not carry video/audio itself, it provides a method to control independent RTP video and audio streams (they are in turn independent one from another).
One of the options though is when RTP streams are tunnelled through RTSP connection, in which case all communication might be taking place through single TCP connection.
You can read the SDP returned in the SETUP request to the RtspServer.
There should be a MediaInformation for each stream available.
That will tell you if there is audio or video etc...
http://en.wikipedia.org/wiki/Session_Description_Protocol