I am trying to implement a live streaming application using flutter. For that, I need to implement a webRTC server with NodeJS. Is there a way to implement a webrtc server for one to many server with node JS?
Check out Mediasoup. It's a WebRTC server implemented in NodeJS. You can install it just like any npm package and use along with rest of your implementation.
Related
I'm developing a cross-platform application (Windows/Linux/macOS), which:
Should be able to run locally as a desktop application
Should be able to run on a remote machine, preferably with the same UI as on local.
So far Electron looks good enough for the first bullet.
The question is about the second one. I'd like to reuse both the logic and the UI from the local scenario for the remote scenario. In other words, I'd like to use Electron based application as a web server and connect to it via browser.
Is there any way to do that?
You can create a remote web server without Electron, just with Node.js, and deliver a web page like a PHP server for example.
But if you want to deliver a complete remote UI and manage the window remotely, that's VERY complicated to do... That needs to configure a lot of things and manage connections between client and server, using asynchronous keyboard & mouse and encryption to secure the communications.
You want a hybrid app I think, like a few of them that out out there (i.e. Slack). Generally there's a web app plus an Electron client version with some shared UI code but it's not Electron all around. The main point of electron is to be a local web server acting as a desktop app. You could certainly keep it all node though.
Are there currently solutions where your server can act as the peer of a WebRTC connection?
The reason I am interested in WebRTC is not the peer-to-peer part of it, but because it enables you to use UDP. You could let players participate in a fast-paced game like Quake without needing any plugins.
It seems that essentially this same question was asked before, but surely things must now be quite different as 2 years have passed.
Yes, it is possible to deploy your WebRTC peer code on server. But since you need to run it on server, it's essentially different from how you run the WebRTC code within the browser - i.e. through a Java Script.
For server based WebRTC peer, you would need to use the WebRTC native code available on platforms - Windows, Mac OS X, Linux, Android and iOS. You can get the WebRTC native code from - https://webrtc.org/native-code/development/
Follow the instructions here to download and build the environment. Sample applications are also present in the repository at the locations - src/webrtc/examples and src/talk/examples
In summary you have use the WebRTC source code that is embedded in the browser in your application code and call the relevant methods / API for the WebRTC functionality.
I have answered similar question at: WebRTC Data Channel server to clients UDP communication. Is it currently possible?
We have implemented the exact same thing: a server/client way of using WebRTC. Besides we also implemented data port multiplexing, so that server would only need to expose one data port for all rtcdata channels.
Here is 2018 update for you: your off-the-shelf solutions are:
Red 5 Pro, Wowza, Kurento, Unreal Media Server, Flashphoner
Also note that in modern public networks TCP is not much slower than UDP;
but UDP may have a considerable packet loss, so try WebRTC-TCP for your Quake idea.
I tried to build a server for integrating the webrtc native APIs in an native app, but I am not sure about how the server should be configured, like the ICE/STUN/TURN, signaling, media server etc..
So far as I know is the open source project: https://github.com/priologic/easyrtc
Can anybody give some recommendations?
Thanks
In a WebRTC infrastructure, there are several things involved. The client part is written in JavaScript and runs on the browser.
But as you said it is a server side part. First there is a ICE/STUN/TURN server that it's used for a client to discover its public IP address if it is located behind a NAT. Depending on your requirements could not be necessary to build/deploy your own server, but use an already public (and free) existing one - here's a list. You can also deploy an open source one like Stuntman.
Then it comes the signaling part, used by two clients to negotiate and start a webrtc session. There is no standard here and you have a few options.
You can use an XMPP server with a Jingle extension. You can deploy an existing XMPP server, like OpenFire or Tigase
You can also use SIP, a protocol much more encountered for VoIP. You can use JAIN-SIP or SIP Servlets.
Or you can develop your own signaling protocol using something like websockets.
The server side options that I was giving you were Java based ones, but you can find similar for other infrastructures too.
STUN/TURN is required. Use public ones (not absolutely stable) or get a Ubuntu machine ans install from the source: https://code.google.com/p/rfc5766-turn-server/
Signaling is trivial. You just forward messages between peers. Just build a simple chat server.
Media server is whole different story and require sophisticated client-server configuration.
In WebRTC, I always see the implementation about peer-to-peer and how to get video streaming from one client to another client. How about server-to-client?
Is it possible for WebRTC to streaming video file from server-to-client?
(I am thinking about using WebRTC Native C++ API to create my own server application to connect to the current implementation on chrome or firefox browser client application.)
OK, if it is possible, will it be faster than many current video streaming services?
Yes it is possible as the server can be one of the peers in that peer-to-peer session.
If you respect the protocols and send the video in SRTP packets using VP8, the browser will play it. To help you build these components on other applications or servers, you can check this page and this project as a guide.
Now, comparing WebRTC with other streaming services... It will depend on several variables like the Codec or the protocol. But, for instance, comparing WebRTC (SRTP over UDP with VP8 Codec) against Flash (RTMP over TCP with H264 Codec), I would say that WebRTC wins.
The player will be Flash Player against the native <video> tag.
The transport would be TCP against UDP.
But of course, everything depends on what you are sending to the client.
I have written some apps and plugins using the native WebRTC API, and there isn't a lot of information out there yet, but here are a few useful resources to get you started:
QT Example: http://research.edm.uhasselt.be/jori/qtwebrtc
Native to Browser example: http://sourcey.com/webrtc-native-to-browser-video-streaming-example/
I started with the WebRTC Native C++ to Browser Video Streaming Example but it doesnot build anymore with the actual WebRTC Native Code.
Then I made modifications merging into a standalone process :
management of the peerConnection (the peerconnection_server)
access to Video4Linux capture (the peerconnection_client).
Removing the stream from browser to the WebRTC Native C++ client give a simple solution to access throught a WebRTC browser to a Video4Linux device that is available from GitHub webrtc-streamer.
Live Demo
We are attempting to replace MJPEGs with Webrtc for our server software and have a prototype module for doing this using a smattering of components tied to the Openwebrtc project. It has been an absolute bear to do, and we have frequent ICE negotiation errors (even over a simple LAN), but it mostly works.
We also built a prototype with the Google Webrtc module, but it had many dependencies. I find it easier to work with the Openwebrtc modules because Google's stuff is so tightly tied to general peer-to-peer scenarios on the browser.
I compiled the following from scratch:
libnice 0.1.14
gstreamer-sctp-1.0
usrsctp
Then I have to interact with libnice a bit directly to gather candidates. Also have to write out the SDP files by hand. But the amount of control--being able to control the source of the pipeline--makes it worthwhile. The resulting pipeline (with two clients off one server source) is below:
Of course. I'm writting a program using native WebRTC api which can join the conference as a peer and record both video and audio.
see: How to stream audio from browser to WebRTC native C++ application
and you can definitely streaming media from native app.
I'm sure you can use dummy_audio_file to streaming audio from local file, and you can find a way to access the video streaming progress by your own.
Yes it is. We have developed an load test tool to publish and play for Ant Media Server. This tool can broadcast media file. We used the same native WebRTC library used in Ant Media Server.
Sure it's possible, it allows covert live streaming to WebRTC, for example:
OBS/FFmpeg ---RTMP---> Server ---WebRTC--> Chrome/Client
For this scenario, it allows the ultra low latency live streaming, about 600~800ms, to play the live streaming by WebRTC. Please take a look at this demo.
I would like to use Adobe Flex to make a client and node.js to make a server for long polling. I am trying to develop a chat application in Flex.
Can any one suggest me how I can accomplish this?
Try to look at (source code of) Socket.io client and server modules which are using Adobe Flash Socket as one of it's transport options.
Broadly there are two ways to approach this:
Direct TCP using a combination of flash.net.Socket (Flash) and net.Server (node.js).
HTTP-based polling using a combination of flash.net.URLLoader (Flash) and http.Server (node.js).
It depends on your requirements.