I'm trying to put in place a basic streaming system from the browser.
The idea is to let the user stream audio live from his mic through the browser and then allow others to listen to this stream with their browser (desktop, mobile, etc ...) and iOS/Android apps.
I started doing some tests with the Red5 Server (which is a great free alternative to the Flash Media Server).
With this technologie, I can publish a stream with the RTMP (ex: rtmp://myserver/myApp).
But the problem is that I can't find a way to read the published stream on other plateforms (using the video tag with HTML5, in iOS, etc ...).
As i failed to that, my question is:
How can I let a user to stream his voice over the net (using flash or not) and then allow the others to listen to that stream by using lightweight technologies (HTML5) and mobile apps?
Thanks,
Regards
Looks like RED5 should be able to do what you want...
0.9.0 RC2 has the ability to:
Streaming Audio (MP3, F4A, M4A)
Recording Client Streams (FLV only)
some links that may help:
http://osflash.org/pipermail/red5_osflash.org/2007-April/010400.html
http://www.red5chat.com/
Though not exactly what you're after, you could take a look at BigBlueButton which is a web conferencing suite based on open source components (RED5 is one of them). It's has a rather complex architecture but they do have a flash based client you can take a loot at.
Related
I want to make an Web app which streams user's webcam which broadcast to viewers
like one to many!
I know getuserMedia() will help me to get user's webcam. now how to stream this data with audio.
I google about this i get few result like using WebRTC and peerjs can do this but I need some kick-start guide like some code or documentation!
Well, you can try easyrtc, is a node.js library for giving WebRTC support to your apps. Is open source and packs some cool demos for you to start making awesome stuff.
my current setup involves streaming from a GoPro to a linux box, and I managed to get bareSIP running on the box to stream the video locally with the 'v' command. However, there's no documentation or commands to configure an RTP broadcasting stream. Would anyone have any insight into publishing an RTP/RTSP output stream for other users to view on their devices?
I've used Unreal Streaming Media components and found them to be very good. They are lightweight and fast yet very powerful.
Using Unreal components you could install the stream forwarder on your laptop, point it at the RTSP stream and tell it to forward to the Distribution server application.
This app can host thousands of connections (supposedly) and last I looked you didn't need a license if you have 3 or fewer sources. The stream can be viewd via their own small player app, via a web player such as jPlayer or via VLC etc.
I've been pretty happy with this before - it saved me from having to use the Live555 streaming mess.
Good Luck!
I am looking to build an app that needs to process incoming audio on a phone call in real time.
WebRTC allows for this but i think this works only in their browser based P2P audio communications functionality but not for phone calls/ VOIP.
Twilio and Plivo allow you record the audio for batch/later processing.
Is there a library that will give me access to the audio streams in real time? If not, what would I need to build such a service from scratch?
Thanks
If you are open to using a media server (so that the call is not longe P2P but it's mediated by the media server using a B2B model), then perhaps the Kurento Media Server may solve your problem. Kurento Media Server makes possible to create processing capabilities which are applyied in real time onto the media streams. There are many examples in the documentation of computer vision and augmented reality algorithms applied in real time over the video streams. I've never seen an only-audio processing module, but it should be simple to implement just by creating an additional module, which is not too complex if you have some knowledge about C/C++ and media processing concepts.
Disclaimer: I'm part of the Kurento development team.
I'm working on a web app in node.js to allow clients to view a live streaming video via a unique url that another client will broadcast from their webcam, i.e., http://myapp.com/thevideo
I understand that webRTC is still not supported in enough browsers to be useful.
I would also like to save this the video stream to be viewed later within the app.
Things get somewhat confusing as I try to narrow down a solution to make this work.
I would like to get some recommendations on proven solutions out there to make this work on desktop and mobile? Any hints would be great.
I'll make a quick suggestion based on the limited details. I would use ffmpeg to encode to HLS. This format will playback natively on iOS and safari on Mac. For all other platforms, either provide an rtmp stream with a flash front end, or use jw player 6 commercial version that can play HLS. Or use a wowza server to handle this all for you.
I am working on a project which will involve http live media streaming from a variety of devices like android phones/tablets, iphone, ipad, browser,etc. It will be a 2 way communication for all the devices with multiple devices connected to a conversation. I have implemented it partially i.e. one way by capturing audio from android phone(native app) and streaming to a web browser(HTML5 app) with a PHP server using ffmpeg and cvlc. I wanted to know of the best way to go ahead about it. Like, if there are any standards to be followed. Also what kind of a server should I be using? I don't want to use any streaming servers like Red5. I would like to implement the streaming logic similar to Http LiveStreaming by apple. I have come across MPEG-DASH that seems to be a standard for http streaming. I still have to look deeper into it. I was also thinking of using NodeJS for its popularity with streaming. Another worry was how do I go about capturing of media from devices? As in, should I use the native capability of the devices to convert media into an mp4 or any container that it supports and then stream it to the server or capture audio and images for a particular period of time and then send it to server and create a common output(I am not really sure of this idea). The separate capture is basically for simplifying the process of video streaming from the server end to any device. I was also thinking if I could completely bypass the server in any cases like a phone to phone or phone to tablet connection.
I just wanted to be sure of the things I will be using/implementing so that I wouldn't have to make drastic changes later on. Any help is deeply appreciated. Thank you.