I have an IP camera which sends out a live stream in RTSP over UDP and I want to display this stream in the browser, and I want it to work on the major browsers and on mobile (both iOs and Android). To achieve this I want to convert the stream to HTTP Live Streaming (HLS) on the server before sending it to the client. Now I've read that not very long ago Apple added support for fragmented MP4 (fMP4) as format for the stream, whereas normally the stream would be sent in MPEG-TS format. And fMP4 is also the format that MPEG-DASH supports, and MPEG-DASH might be the industry standard in a few years.
Now my question is, what the advantages and disadvantages of both fMP4 and MPEG-TS are?
EDIT: According to the technical notes for HLS from Apple, live streams must be encoded as MPEG-TS streams (https://developer.apple.com/library/content/technotes/tn2224/_index.html#//apple_ref/doc/uid/DTS40009745-CH1-ENCODEYOURVARIANTS). Is there a reason for this or is this information outdated?
fMP4 is likely to replace TS as a standard. It has less overhead and is required for HEVC, but the main advantage is compatibility with DASH - i.e. you can generate both HLS and DASH using the same files, which helps with compute and storage costs. For your particular use case, HLS TS probably has the more coverage (due to old devices and players) than HLS fMP4, but HLS+DASH fMP4 is what I would choose.
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I've implemented an HLS service with ffmpeg (which pulls a live stream from nginx-rtmp). That all works fine, but now I'm wondering what kind of programming pattern I should be using to get live captioning to work.
I'm planning on using ffmpeg to output the incoming mp4 stream to multiple WAV chunks (i.e., the same way HLS fMP4 parts are created), and then sending those chunks over to Azure Cognitive Services for speech-to-text recognition. My question is, what do I do when I receive the speech results? Do I dump that vtt file into the same directory as my HLS chunks, and then serve that up using a single m3u8 file (with audio/video tracks along with the text track)?
Currently ffmpeg is updating the m3u8 playlist for HLS clients; would it be possible for me to create the m3u8 playlist just for the vtt files, and serve that concurrently with the "regular" HLS playlist? Also, time synchronization would seem to be difficult, because I'll be sending discrete WAV files over to Azure, so the vtt timestamps are going to be relative to the chunk I'm sending.
Help! I've done searches online, and I grasp the various issues, but I'm not sure how to plumb them all together.
I'm trying to stream the video of my C++ 3D application (similar to streaming a game).
I have encoded an H.264 video stream with the ffmpeg library (i.e. internally to my application) and can push it to a local address, e.g. rtp://127.0.0.1:6666, which can be played by VLC or other player (locally).
I'm not particularly wedded to h.264 at this point, or rtp. I could send as srtp if that would help.
I'd like to use WebRTC to set up a connection across different machines, but can't see in the examples how to make use of this pre-existing stream - the video and audio examples are understandably focused on getting data from devices like connected web cams, or the display.
Is what I'm thinking feasible? I.e. ideally I'd just point webRTC at my rtp://127.0.0.1:6666 address and that would be the video stream source.
I am writing out an sdp file as well which can be read by VLC, could I use this in a similar way?
As noted in the comment below there is an example out there using go to weave some magic that enables an rtp stream to be shown in a browser via webRTC.
I am trying to find a more "standard" way to be able to set the source of a video track in webRTC to be the URL of an encoded stream. If there isn't one, that is valuable information to me too, as I can change tack and use a webrtc library to send frames directly.
Unfortunately FFMPEG doesn't support WebRTC output. It lacks support for ICE and DTLS-SRTP.
You will need to use a RTP -> WebRTC bridge. I wrote rtp-to-webrtc that can do this. You can do this with lots of different WebRTC clients/servers!
If you have a particular language/paradigm that you prefer happy to provide examples for those.
On a project, we have a camera with a RTSP stream (video & audio, encoded in H264). We need to make the stream available on all browsers (desktop & mobile).
I've seen some solutions:
Convert the stream on HLS (iOS) and MPEG DASH (other browsers) with FFMPEG on a server
Video only with jsmpeg
The problem is since we need a really live streaming (e.g. the user can record some pictures/video on live), a low the latency solution is a specification of the project.
Any ideas?
My client has a requirement where he needs me to transcode a source file into a proxy with a unique burn in on it per playback.
For the proxy I will be using ffmpeg, nothing fancy, but ideally the users can play back the file as it is being transcoded since it may take up to several minutes to complete the transcoding.
Another restriction is that the player does not support HLS and other live streaming options and can only accept MP4s as a source.
Any ideas/suggestions would be great.
It seems you have conflicting requirements. mp4 is VERY poorly suited for live streaming. It is 'possible' to create a fake moov and have the player perform byte ranges. But it is very inefficient. You really need a player or platform that supports streaming formats such as fmp4 (fragmented mp4/dash) hls, ts, flv, rtmp, rtc, etc.
I have an Open GL application that renders a simulation animation and outputs several PNG image files per second and saves these files in a disk. I want to stream these image files as a video streaming over HTTP protocol so I can view the animation video from a web browser. I already have a robust socket server that handles connection from websocket and I can handle all the handshake and message encoding/decoding part. My server program and OpenGL application program are written in C++.
A couple of questions in mind:
what is the best way to stream this OpenGL animation output and view it from my web browser? The video image frames are dynamically (continuously) generated by the OpenGL application as PNG image files. The web browser should display the video corresponding to the Open GL display output (with minimum latency time).
How can I encode these PNG image files as a continuous (live) video programmatically using C / C++ (without me manually pushing the image files to a streaming server software, like Flash Media Live Encoder)? What video format should I produce?
Should I send/receive the animation data using a web-socket, or is there any other better ways? (like JQuery Ajax call for instead, I am just making this up, but please guide me through the correct way of implementing this). It is gonna be great if this live video streaming works across different browsers.
Does HTML5 video tag support live video streaming, or does it only work for a complete video file which exists at a particular URL/directory (not a live streaming)?
Is there any existing code samples (tutorial) for doing this live video streaming, where you have a C/C++/Java application producing some image frames, and have a web-browser consuming this output as a video streaming? I could barely find tutorials about this topic after spending few hours searching on Google.
You definitely want to stop outputting PNG files to disk and instead input the frames of image data into a video encoder. A good bet is to use libav/ffmpeg. Next, you will have to encapsulate the encoded video to a network friendly format. I would recommend x264 as an encoder and MPEG4 or MPEG2TS stream format.
To view the video in the web browser, you'll have to choose the streaming format. HLS in HTML5 is supported by Safari, but unfortunately not much else. For wide client support you will need to use a plugin such as flash or a media player.
The easiest way I can think of to do this is to use Wowza for doing a server-side restream. The GL program would stream MPEG2 TS to Wowza, and it would then prepare streams for HLS, RTMP(flash), RTSP, and Microsoft Smooth Streaming (Silverlight). Wowza costs about $1000. You could setup an RTMP stream using Red5, which is free. Or you could do RTSP serving with VLC, but RTSP clients are universally terrible.
Unfortunately, at this time, the level of standardization for web video is very poor, and the video tooling is rather cumbersome. It's a large undertaking, but you can get hacking with ffmpeg/libav. A proof of concept could be writing image frames in YUV420p format to a pipe that ffmpeg is listening to and choosing an output stream that you can read with an RTSP client such as VLC, Quicktime, or Windows Media Player.
Most live video is MPEG2 internally, wrapped up as RTMP (Flash) or HLS (Apple). There is probably a way to render off your OpenGL to frames and have them converted into MPEG2 as a live stream, but I don't know exactly how (maybe FFMPEG?). Once that is done you can push the stream through Flash Media Live Encoder (it's free) and stream it out to Flash clients directly using RTMP or push publish it into Wowza Media Server to package it for Flash, Http Live Streaming (Cupertino), Smooth Streaming for Silverlight.
Basically you can string together some COTS solutions into a pipeline and play on a standard player without handling the sockets and low level stuff yourself.