After many hours of research and nothing relevant coming up I decided to ask.
I am pretty new to the concept of video streaming, so please forgive me if my questions may seem elementary.
I am building a project that needs to include media streaming functionality. It should has the following options:
VOD - user uploads a file to the server, that needs to be transcoded to few MP4 files of different resolutions. For transcoding I am trying the approach using CloudTranscode (https://github.com/bfansports/CloudTranscode) deployed as a Docker image. The server should supply stream to the player with certain buffer size, so when the playback is paused we buffer for instance +5 seconds and that's it. Adaptive bitrate would be nice, however I'm not sure how this works with different players (I was thinking about using Video.JS due to high customization option, plus it's free).
Live video capturing - user visits a certain page that captures video from the webcam and sends the stream to the server for further stream distribution to clients. For most browsers WebRTC could be a good option, but iOS devices probably won't work with it, so any suggestions here would be much appreciated
Live video streaming - users visit a certain page where they can watch the stream captured from the user mentioned in point 2. Here the stream may be watched by one or many users (may be as well 1 or 10,000 users)
Cutting to the chase my questions follow:
What would be the best media server software that I can use for that purpose, having on mind high scalability (deployed as Docker container on AWS EC2), and possible huge load of both streaming and watching users, as well as multi-device/platform/browser support?
What would be the best media player for webpage that (again) would be cross-browser/platform/device, keeping in mind good integration with media server itself for purpose of adaptive resolution streaming? Also it would be nice if the player has broad customization options in matter of appearence (for instance thumbnail display when hovering the timeline).
Do you know any better solution for video transcoding than mentioned CloudTranscode, having on mind Docker setup, and some easy to use API (here some on-the-fly transcoding would be nice, so the worker wouldn't need to wait for the whole file to be uploaded)?
What happens if I use autoscalling functionality on EC2 instance, and more instances of the media server are being automatically started? Let's say we have instance 1 (I1) and instance 2 (I2). Some user started broadcasting on I1, and 1000 users are watching the stream which is the server instance's limit because it's running out of resources. Next, another couple of users are trying to view the stream, so they are being connected to I2 by AWS load balancer - how does that work with live stream? Sorry, but I am total newbie to the concept, so again - forgive me for elementary questions.
So far a was able to find a few media servers that may be relevant to my needs including:
Wowza Media Server (paid)
Red5 media server (free)
Kurento Media Server (free)
My application is written in Laravel, ergo I need some PHP integration with the media server.
Obviously free solutions are the most welcome, however I do not mind to pay as long as paid solution covers my needs.
Any input here will be much appreaciated - even partial solutions / suggestions. I'm kinda stuck here, so any suggestions that can bring me closer to the solution are very welcome!
Best regards
If anyone needs such information I ended up using Nginx Plus media server functionalities. It's capable of serving both live and VOD streams, it has out-of-the-box load balancer to switch traffic over multiple container instances and many more great features. Plus they have images to deploy directly from AWS Marketplace, and the license is paid hourly when the EC2 instance is running. Ofcourse there is free version as well, but I am really satisfied with Nginx Plus support.
Capturing live stream from user I've done using getUserMedia() in JS. Still having minor glitches, but I will get it to work (problems are related with WebM chunks that MediaRecorder API spits out, but I'm almost done here using some Python piece of code modifying each chunk on server side).
If anyone needs help I will be happy to help.
Related
As a beginner at working with these kinds of real-time streaming services, I've spent hours trying to work out how this is possible, but can't seem to work out I'd precisely go about it.
I'm prototyping a personal basic web app that does the following:
In a web browser, the web application has a button that says 'Stream Microphone' - when pressed it streams the audio from the user's microphone (the user obviously has to consent to give permission to send their microphone audio) through to the server which I was presuming would be running node.js (no specific reason at this point, just thought this is how I'd go about doing it).
The server receives the audio close enough to real-time somehow (not sure how I'd do this).
I can then run ffmpeg on the command line and take the real-time audio coming in real-time and add it as the sound to a video file (let's just say I'm going to play testmovie.mp4) that I want to play.
I've looked at various solutions - such as maybe using WebRTC, RTP/RTSP, Piping audio into ffmpeg, Gstreamer, Kurento, Flashphoner and/or Wowza - but somehow they look overly complicated and usually seem to focus on video along with audio. I just need to work with audio.
As you've found there are numerous different options to receive the audio from a WebRTC enabled browser. The options from easiest to more difficult are probably:
Use a WebRTC enabled server such as Janus, Kurento, Jitsi (not sure about wowzer) etc. These servers tend to have plugin systems and one of them may already have the audio mixing capability you need.
If you're comfortable with node you could use the werift library to receive the WebRTC audio stream and then forward it to FFmpeg.
If you want to take full control over the WebRTC pipeline and potentially do the audio mixing as well you could use gstreamer. From what you've described it should be capable of doing the complete task without having to involve a separate FFmpeg process.
The way we did this is by creating a Wowza module in Java that would take the audio from the incoming stream, take the video from wherever you want it, and mix them together.
There's no reason to introduce a thrid party like ffmpeg in the mix.
There's even a sample from Wowza for this: https://github.com/WowzaMediaSystems/wse-plugin-avmix
I'm used to using ffmpeg and stuff to broadcast/do testing.. but I don't understand how iptv servers succeed at having 50+ input streams, making 50+ output streams and sharing them, as I can't even run 3 ffmpeg commands with encoding without having the CPU crying for help...
I've tried to get infos, but except Wowza that seems to do what I'm trying to understand, I don't find any info...
I hope that you can enlight me on understanding how this whole thing works. Also, I'd like to test it out so if you got any recommendations on how to do this, I'll be thankful to you !
Most large streaming services actually will have multiple servers - this is partly due to different function being performed by different servers and also due to performance as you have noted.
There are many different ways you can stitch a service together but it will generally (for live streams) have the following elements:
some sort of live encoder which receives the external stream and converts it to a format the rest of the system understand
transcoders - these take the inout video and create multiple bit rate versions of it to support Adaptive Bit Rate Streaming (see: https://stackoverflow.com/a/42365034/334402)
Packagers - these package the resulting video streams into the required video streaming protocol, usually HLS or MPEG DASH these days. This is typically done 'Just in Time' so only streams and bit rates required are actually packaged. If encryption is required it is typically applied at this point also.
Origin server and CDN - the video streams, which actually consist of packets of data making up the ABR video segments, are delivered to an Origin server which is the source for the CDN. The CDN, Content Delivery Network, is alike a large dispersed video cache and it copies the video to the edge of the network to reduce latency when a user request the video.
You can also build this using cloud services rather than installing or spinning up the servers yourself - it might be useful to look at some of the documentation from providers like AWS Media Services or BitMovin.
Whichever way it is done, your initial thoughts are correct - it takes quite a bit of horsepower to server large numbers of video streams.
Preface
I have read this two part tutorial (Part-1 and Part-2) by Steamroot on MPEG-DASH, and below is my understanding (please correct me if I am wrong):
The video needs to be encoded into multiple bit-rates using FFmpeg.
The encoded videos need to be transcoded (dashified) using MP4Box.
The dashified videos can be served using a web server.
Problem
I intend to live-stream an event and I need help to understand the following:
Can I club the FFmpeg and MP4Box commands into a single step? Maybe through a wrapper program so that I do not have to run them separately? Is there any other or better solution?
How do I send the dashified content to the web server? FTP? Would any vanilla web server do?
Lastly, a friend had hinted that I could also use GStreamer to achieve my objective. But, I could not find any good resource on the internet for the same. So, where (and how) does GStreamer fit in the above process?
What is the format you will be getting out of your camera for your live-event? There are a lot of solutions a lot more adapted for live streaming (the tutorial I wrote is for VOD streams only). You can check out simple solutions like Wowza Streaming Server, Nible streamer (free), etc, that take a RTMP stream and transform it into other formats (HLS, DASH, etc...).
Most of the livestreaming platforms can even do that for you (livestream.com, youtube, twitch, or even facebook now)
The dashified content will be requested as HTTP ressources by the browser or other players. In the case of a VoD stream, indeed you just need to make the dash segments available through a web-server. For live content, you need something smarter, that will encode, package the segments and make them available on the fly.
Gstreamer can transcode and transmux the original content, and can do it on the fly. You will be able to get different formats as outputs, like RTMP, HLS, and probably even mpeg-dash. Then you still need to make your content available via a webserver.
In conclusion, if you just want to transmit an occasional live event, it's probably a lot easier a platform that will ingest your RTMP stream and do all the complicated steps for you.
Can anyone comment on the performance implications of storing streaming media in a SharePoint 2007 document library? I’ve heard this can be detrimental to the performance of the farm due to the media being streamed from storage in a SQL DB.
Has anyone had any firsthand experience with this and if so, what alternatives have you used to provide users with the ability to publish and mange their own video content? Assume a secure internal environment so external services like YouTube are not viable in this scenario.
I have tried this on a test deploy and it had very poor performance. Not only did the SharePoint server struggle, but the video the client was trying to stream was very laggy. Granted, we did not have a state of the art server set up, but I was the only one accessing the server and it couldn't even handle that. Given my experience, I would advise against it.
I can stream FLV's for flash movies from a document library with reasonable performance. I still opted for deploying them to a separate non-sharepoint website because the video's where fairly static and do take up a lot of SQL space.
You might consider activating blobcaching to get around the streaming from the database, see: http://msdn.microsoft.com/en-us/library/aa604896.aspx
I want to have a stress/performance testing for my content management site, especially for hosted streamed video part. I am using IIS to host the videos. More specifically, I am using the new Windows Server 2008 x64 and IIS 7.0.
The confusion is,
I plan to write code to start a lot of threads, and in each thread I will send web request to video URL, and read response stream from server, but I am not sure whether in this way, it behaves the same as a real user using player to render the video (in my code, I just read the stream, without really play it or write to anywhere). I want to test similar to the real scenario as much as possible;
I also plan to use real Media Player to render video (or what-so-ever media player), but my concern is if I start multiple Media Players on my test machine, since Media Player will utilize some H/W or some other resources (video card specific memory?) to decode/render the video (not sure, needs guru help to check and confirm), if I start multiple players, are there any potential H/W or resource contention between the players? If there is contention, it is also not actual ens user scenario, i.e. few user will start 100 players on his/her machine. :-)
Does anyone have any advice to me?
BTW: I prefer to use any .Net based solution, but not a must.
thanks in advance,
George
You should use mplayer. It has a lot of command line options. I don't know how all theses options are available under Windows, but under linux something like this is possible :
mplayer some_url -dump-video -dump-file=some_file
It will behave the same as a "normal" player I think, and your test machine won't need to handle hundreds of decompression thread, sot it fits your need 1 and 2
If you know the bit rate of your video stream, you can pace your downloading request to simulate video player clients. The bit rate can be calculated from the information carried in the stream, but it's a little more complicated. There is software for stressing testing video server too, such as this IP Video Monitor.