I decided to text you, because I need help with this.
In the image below, you can see that OpenTok shows the user the internet connectivity latency or quality icon. (the space closed in White circle)
I was looking at the documentation and different forums, but I could not find how to do it or show it in Javascript.
Could you please help me and share where I can find the implementation steps or how I accomplish this?
Thank you so much.
Regards!
TokBox Developer Evangelist here.
You could use the OpenTok Network Test JS module to check network connectivity before people join the video chat or you can use subscriber.getStats() and publisher.getStats() for when people are in the video call. These methods return the following:
Stream quality
Total audio and video packets lost
Total audio and video packets received
Total audio and video bytes received
Current video frame rate
Please keep in mind that you would have to build the widget based on the data returned from the methods mentioned above.
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 have a Dahua IP camera with a microphone. I would like to get the audio stream playing on a website like a live radio.
I have a raspberry pi which I was planning to use with ffmpeg, but I haven't had much success in bridging the gap between that and my website to form an audio stream.
Can this be done via sftp/ftp with mp3 files and some fancy php/javascript to play like a live radio?
Do I need to use another service? (would like to minimise costs as much as possible!)
Thanks!
Peter
Browsers can't play RTSP source directly so you need a streaming server to get the RTSP stream and convert it to something suitable for HTML5 playback (HLS/DASH).
This can be done with Wowza SE (the free developer license could be used for 10 subscribers).
You can see how this works with the VideoNow.Live - IP Camera Restreaming feature and see how this is done by taking a peek at the open source code of the Live Streaming HTML5 RTSP WP plugin that powers that solution.
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.
idea is to create a video from images provided by a user and at the same time stream the generated video to other user demanding it.
kindly tell any efficient way to do this and which language out of PHP and C# .net will be suitable.
have looked into ffmpeg to take images and convert to video and save to server and then stream .. kindly tell if this the possibility or any other method for live streaming.
regards
UPATE
consider the following scenario as I understand:
get images from server and start combining them to form a video. at the same time, stream the video to the users requesting it.. for new coming clients, stream the previously generated video from the begining and keep on sending the new video which is being generated from images to the previous clients.
kindly tell if this is possible, if so then what can be the approach. Have read something about pipes but am completely new to ffmpeg and streaming in general.
Yes, this is possible with ffmpeg. Any language that is turing complete is suitable. They are many methods of live streaming including HLS, RTP, RTMP, etc.
If you need more detailed answers. Please ask more detailed questions.
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.