I am involved in building a Real Time Messaging Protocol Parser.I am collecting the video/audio data from the RTMP packets.Now to play a video in any player I need to know the container format as well as the codec used.In the video data I am getting from the RTMP packets I know the codec used (for eg. On2 VP6).But I don't know how to know the container of the audio/video stream that I am receiving . So should I assume that RTMP support only FLV container ??? Or is it possible for me to get audio/video packets from any other container formats ?? If Yes then how to know the type of container used from the RTMP data from the information present in RTMP packet ?Adobe specification for RTMP does not provide any information regarding the container of the audio/video data. Any help on this ??? I am stuck here for quite some time.
It is a bit wrong question.
RTMP is a transport protocol that includes containers inside.
Technically it is not correct to say that RTMP carries FLV, because FLV has two layers of incapsulation and RTMP carries only bottom level.
So, it is right to say that RTMP can transfer only those codecs that FLV can and it is not 100% right to say that RTMP transfers FLV.
Adobe's specification of RTMP was created not for developers but for a legal issue against Wowza, so it is not written for you to understand what is happening. Read sources of red5, crtmp or some other rtmp server, they are rather easy to understand.
Related
I don't know how to get started with this.
What I am trying to do is get a video + audio stream from front-end and host the live stream as mp4 thats accessible on browser.
I was able to find information on WebRTC, socket.io, rtmp, but I'm not really sure what tool to use / whats best suited for something like this?
also follow up question, my front-end is iOS app. So what format would I send the live stream to the server?
It depends on which live streaming protocol you want to play on the player, as #Brad said, HLS is the most common protocol for player.
Note: Besides of HLS, iOS native app is able to use fijkplayer or FFmpeg to play any format of live streaming, like HLS, RTMP or HTTP-FLV, even MKV. However, the most straight forward solution is HLS, only need a tag to play MP4 or HLS, and MSE is also a optional solution to use flv.js/hls.js to play live streaming on iOS/Android/PC, this post is about these protocols.
The stream flow is like this:
FFmpeg/OBS ---RTMP--->--+
+--> Media Server---> HLS/HTTP-FLV---> Player
Browser ----WebRTC--->--+
The protocol to push to media server, or receive in node server, depends on your encoder, by RTMP or H5(WebRTC):
For RTMP, you could use FFmpeg or OBS to push stream to your media server.
If want to push stream by H5, the only way is use WebRTC.
The media server coverts the protocol from publisher to player, which use different protocols in live streaming right now(at 2022.01), please read more from this post.
I write program to receive data from webcam, afterward I forward it to video frames type and add it to application client - Jframe in Java.
But now I want to send this frames to rtsp server - with purpose live streaming to other client (same VLC).
I know that if input is video source so simple, but this way is video frames type.
So could you help me how to send this data to server RTSP or introduce to me what's server can resolve this issue.
Thanks & Best Regard.
I'd like the .mp3 files being streamed inaccessible to the listeners, but without having to sacrifice mobile compatibility. Which protocol would be best for that?
There is no such thing as inaccessible streaming. How are you going to stream if it is inaccessible? :) If a user can listen to any song via streaming, it is accessible to user.
If you are trying to prevent users from recording or downloading .mp3 files for your stream, you are falsely thinking security through obscurity. If a device can get a data over a network and play it, there is surely a way record the data. It's either by capturing the network traffic, or reverse-engineering your application to understand the protocol you are using to play songs. Whatever you do to obscure your protocol, it will surely be reverse engineered.
i have an RTSP server which is re-streaming the A/V stream from the camera to clients.
Client side we are using MF to play the stream.
I can successfully play the video but not able to play the audio from the server. However when i use vlc to play, it can play both A/V.
Currently i am implementing IMFMediaStream and have created my customize media stream. I have also created a separate IMFStreamDescriptor for audio and added all the required attributes. When i run , everything goes fine but my RequestSample method never gets called.
Please let me know if i am doing it wrong or if there is any other way to play the audio in MF.
Thanks,
Prateek
Media Foundation support for RTSP is limited to a small number of payload formats. VLC supports more (AFAIR through Live555 library). Most likely, your payload is not supported in Media Foundation.
So I am trying to create a RTSP server that streams music.
I do not understand how the server plays a music and different requests get what ever is playing at that time.
so, to organize my questions:
1) how does the server play a music file?
2) how does the request to the server look like to get whats currently playing?
3) what does the response look like to get the music playing in the client that requested the music?
First: READ THIS (RTSP), and THEN READ THIS (SDP), and then READ THIS (RTP). Then you can ask more sensible questions.
It doesn't, server streams little parts of the audio data to the client, telling it when each part is to be played.
There is no such request. If you want, you can have URL for live streaming, and in RTSP DESCRIBE request, tell the client what is currently on.
Read the first (RTSP) document, all is there! Answer to your question is this:
RTSP/1.0 200 OK
CSeq: 3
Session: 123456
Range: npt=now-
RTP-Info: url=trackID=1;seq=987654
But to get the music playing you will have to do a lot more to initiate a streaming session.
You should first be clear about what is RTSP and RTP. The Real Time Streaming Protocol (RTSP) is a network control protocol designed for use in communications systems to control streaming media servers. where as Most RTSP servers use the Real-time Transport Protocol (RTP) for media stream delivery. RTP uses UDP to deliver the Packet Stream. try to Understanding these concepts.
then Have a look at this project.
http://sourceforge.net/projects/unvedu/
This a open source project developed by our university, which is used to stream video(MKV) and audio file over UDP.
You can also find a .Net Implementation of RTP and RTSP here # https://net7mma.codeplex.com/ which includes a RTSP Client and Server implementation and many other useful utilities e.g. implementations of many popular Digital Media Container Formats.
The solution has a modular design and better performance than ffmpeg or libav at the current time.