I have been using Ustream for a small adult education startup for about a year now. Great service and all, but the delay to end user is become unacceptable. It's about 25-30 seconds.
My plan is to have a embedded player and "live talk" window developed on a secure part of our website where distance-learning participants (about 50% of our students) can talk live to the lecturer being recorded.
Our setup is pretty conventional: one consumer video camera fed to my PC via Firewire 800 using Ustream Producer (really a white-labeled version of Telestream Wirecast) and out to the Ustream server.
Where can I find or create a livestream/video stream mechanism with little delay (3-5 seconds max)? Will a dedicated server be needed for this? Adobe Media Server?
UStream is a live streaming service. With most if not all live streaming providers the emphasis is on using a longer latency to increase the video quality.
What you are looking for is a video conferencing or video chat application. There are plenty of commercial services offering those or you can set up you own server.
Adobe FMS, Wowza or Red5 are Flash servers that you can use for that.
You then need a Flash application to run in the user's browser, some example of how to do that are:
http://www.adobe.com/devnet/adobe-media-server/articles/first_im_app.html
http://www.wowza.com/resources/3.6.0/examples/VideoChat/FlashRTMPPlayer/player.html
http://code.google.com/p/flash-videoio/
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My primary intention is to setup a VoIP session between 2 users A & B; Here the raw audio / video media bytes are fetched from A's browser are played in B's browser and vice versa.
The reason is that, when the user C & D are added into this call, we need not have to create a P2P mesh network which limits the performance.
Tried recording media with getUserMedia() and playback, but it is not real time. It also gives a bad user experience. (However, haven't experimented yet with videos of small chunks as 200 ms)
Is there any approach where I can get the raw bytes of the media and play it on other browser? Currently I have a server in between which can connect to both peers if required.
Any online examples or libraries are welcome.
Have already asked 2 questions in this regard with 100-100 bounties, but not much of use:
How to use libsrtp or similar library to decrypt/encrypt the WebRTC data stream?
How to integrate part of WebRTC as a static / dynamic library with the existing C++ code?
Related: How to stream, live video playing on my browser to browser of another user?
If i understand you well is you're looking on how to have more than two users on the session right? without using mesh topology
thats possible and configurable as well by means that some maybe active speaker or everyone is active speaker not only receiver whatever configuration you choose but to me it seems that you're asking for video conferencing
there are couple of tools for this the best one i might recommend is mediasoup its a SFU as selective fowarding unit mediasoup
I don't know if I understand correctly, but it is not likely that you will get raw video data and play it on the browser, it will just kill your bandwith and performance because the raw data is huge.
You need to use the compressed data ( media codec ex.H264 ) and you need a protocol to send and receive it. If you are looking for sub-second latency than webrtc is your best choice in here already. If you have a server in between, distribute your media through that server instead of Mesh. Check this out for webrtc network topologies:
https://antmedia.io/webrtc-servers/
I have an UWP app that capture a live video stream (webcam), encodes it in h264, and sends it through a TCP socket (in a local network, I need high performance) to a Linux device.
Is there a way to do this? I need the video not for playing it but for extract single frames. I could do that with opencv but it requires a local video file, instead I'm using a live stream.
I would send photos instead of a video stream if the time needed for capture one was acceptable, but it requires about 250 ms.
Is RTP required? Does UWP (windows) provides a way to achive this?
Thank you
P.S.: The UWP app runs in Hololens.
You can use WebRTC to transmit live video from the HoloLens easily to any target. That's probably the easiest way to do it without going really low level.
For an introduction just grab this repo and try the sample app which runs perfectly on the HoloLens https://github.com/webrtc-uwp/PeerCC/tree/e95f231e1dc9c248ca2ffa040276b8a1265da145/Client
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 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.