as the title shows, is there any methods I can use to play multiple videos continuously using simple rtmp client(my rtmp server is wowza)? Here is the way I think:
when the first video is about to be finished，open a new thread to send a new createStream command and a new play command and get the video rtmp packet and put them into a buffer list, when the first video is finished, then play the video rtmp in the buffer list..
Can this way be available or are there any other recommended methods to achieve it? Any suggestion will be appreciated, thanks!
While the functionality is not built-in, Wowza does have a module called StreamPublisher to allow you to implement a server-side type of playlist. The source code for the module is available on GitHub. A scheduled playlist of multiple VOD files is streamed as a live stream, similar to a TV broadcast.
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I'm trying to build an web app where there's a broadcaster of a camera stream, and viewers who can watch and control the stream. Is it possible for a viewer to control the constraints of the camera (exposure, brightness, etc.) and possibly pause, rewind, and record footage, being used to broadcast the stream using webrtc? Wanted to know before I decide to use webrtc as the way to accomplish this task. Based on my reading of the webrtc guide and example pages, I think recording is possible. But I wasn't sure about a remote peerconnection changing the local peerconnection's settings or vice versa.
I am able to stream video with Kurento using WebRTC, I need to implement multi party audio conference using MCU feature of Kurento Media server. So audio coming from all clients are merged and send back that combined audio to all clients in efficient manner using WebRTC.
if it will works then we need only two connection(one for send and one for receive) other wise we need peer connection to all clients using WebRTC. It is not feasible to establish peer connection to all all clients.
Please suggest me any sample code which have implemented MCU for audio using Kurento Media Server or guide me to implement same using Kurento Media Server.
I'm afraid there's no code that allows that un Kurento. There is the Composite media element, but that is usually for audio AND video. It combines streams into a single stream matrix of the required size, usually more than 9 streams may have performance problems. If you only want to process audio, surely it could handle much more than 9 streams. To use only audio just connect AUDIO stream to the HubPort.
EDIT 1
The code to generate the media elements needed, and the correct way establish an audio-only connection is as follows.
WebRtcEndpoint webrtc = new WebRtcEndpoint.Builder(pipeline).build();
Composite composite = new Composite.Builder(pipeline).build();
HubPort hubport = new HubPort.Builder(composite).build();
webrtc.connect(hubport, MediaType.AUDIO);
Please note that the connection is from the WebRtcEndpoint to the HubPort. If you need it to be bidirectional, you'll need to connect that way also.
hubport.connect(webrtc, MediaType.AUDIO);
Is it possible to stream video and audio to a rtmp://-server with GPUImage?
I'm using the GPUImageVideoCamera and would love to stream (video + audio) directly to a rtmp-server.
I tried VideoCore which streams perfectly to e.g. YouTube, but whenever I try to overlay the video with different images I do get performance problems.
It seems as GPUImage is doing a really great job there, but I don't know how to stream with that. I found issues on VideoCore talking about feeding VideoCore with GPUImage, but I don't have a starting point on how that's implemented...
how to interrupt rtmp flv broadcast and publish live rtmp broadcast on red5?
I am using osmf strobe player. I have my flv playlist working but when I broadcast live from my webcam what is the formula to stop the flv streams, then play flv countdown video then connect live broadcast from web cam?
Here is how I would do it, from a high-level since I don't have the code for what you're asking and "easy" transitions between streams isn't built-in to the server.
First, create a signaling or event system within your app to accept actions triggered by your broadcaster. Using the signaling system, transition your subscribers / viewers by sending triggered events telling their players to stop playing a current video and start a new one. I suggest using Shared Objects for this when passing signals around. Use server-side methods called by your broadcaster to send the signals on the Share Object. The "play" functionality is the easy part since you simply provide the stream name in your signal / event.
I'm streaming both RTMP and HLS(for IOS and android), with RTMP video.js display correct currentTime. According to me currentTime should be when the stream started, not when the client started to view the stream. But when I go with the HLS-stream currentTime returns when the client started the stream and not when the stream started(same result using any player from android or ios or VLC).
Using ffprobe on my HLS-stream I get the correct values, i.e when the stream started, which makes me believe that I should start looking at the client to find a solution, but I'm far from sure.
So please help me get in the right direction to solve this problem.
I.e is it HLS in nature that doesn't give me correct currentTime, but weird that ffprobe gives me correct answer?
Can't find anything in the video.js code on how to get any other time code.
Is it my server that generates wrong SMTPE timecode for HLS and ffprobe are using other ways to get correct currentTime?
Anyway I'm just curious, I have a workaround for it, by initially counting used fragments I will atleast get in the 5 seconds ballpark, i.e good enough for my case.
Thanks for any help or input.
BR David
RTMP and HLS work in different ways.
RTMP is always streaming, and when you subscribe to the stream, you subscribe to the running stream, so the begin time will be when the stream started.
HLS works differently. When you subscribe to a HLS stream, it creates a stream for you. So the current time will be when the HLS stream was started, which means when you subscribed and the HLS stream was created.