I'm developing a web application that want to share the posts that one user posted in one topic argument via webRTC.
The problem is that webRTC allow only one-to-one peer connection.
What can i do for solve this problem using webRTC?
I need that my information is transfered broadcast to the other peer that are connected in the same room.
For this you need some signaling server like easyrtc, which packages some really awesome functionalities. One of those is multy-party videocalls.
If you want to read more about, and see some cool demos check https://easyrtc.com/ and https://demo.easyrtc.com/demos/
After you establish the connection via signalling server, you can open the DataChannel and make the transfer that you said..
Related
Let's say I have access to a server via ssh and my own laptop. However, I wish to use webrtc to transmit data via a peer connection for it's ultra low latency over UDP. It's also crucial that I have NO communication between a third server.
What I would like to do, is use ssh to get the signalling information for the two machines (offers and answers) to then create a webRTC channel. This way I can say that no information ever went to a third server (including the signalling part).
I understand the requirement is a bit silly, but it's due to how we network things currently, and the important part is the super low latency webRTC provides.
Or is this the wrong tool for the job perhaps? Open to other suggestions :)
Well I am studying about Web-RTC from its official documentation. I need to integrate live streaming in my website but after seeing this and studying about all its documentation. I only learn about streaming on locally like on same browser and same page. But this is not what i want. I want to start stream from my admin panel(This part has been completed) and broadcast that stream whoever has access to my website whatever the website is and whatever device. Whoever open my website if i m streaming then he should see that and if some of u have worked on live streaming then u should tell me. It will be a great help for me. All i did until now is to make a connection between two peers on same page. Now i want to make global peer to peer connection
I have done this implementation using simple-peer. Basically a wrapper for webRTC.
As soon as a new user connects a new webrtc connection should be made between the receiver and the caller. The receiver is firstly initialized and then sends a message to the sender to start the connection. This first connection is all done by your own server you should write.
Here is a working example. And here is the demo. Any connected devices will be automatically connected to the call. Multiple users supported. You'll find all the webrtc code in /public/js/main.js
You have to do signaling which mean that you have to exchange the PEER CONNECTIONS over the server and which required you to build a server page and client page so both of you can exchange the peer connection.
here is the complete procedure of exchanging the peer connection over the server.
find the heading
RTCPeerConnection plus servers
https://www.html5rocks.com/en/tutorials/webrtc/basics/
I want to implement webrtc server in my project. I want to make my own webrtc server and deploy it in amazon server. How can I achieve this?
WebRTC is a peer-to-peer protocol so you don't need a server for this.
You will need a signaling server for session negotiation. How you'll implement this depends on the technology that you'll use - client side: polling, ajax, websockets, stomp etc and server side.
For STUN/TURN you can deploy an existing server or use RFC and develop your own from scratch.
#Adrian Ber is correct, you need a signalling server such as this one:
https://github.com/peers/peerjs-server
You can set one of these up on AWS
You'll also need some code on the client side. There is a matching javascript client library (which does most of the work) here: http://peerjs.com/
There are some examples on the peerjs web site - they either need to be run on your local machine or on https servers (browsers will no longer allow camera access over http)
Ignore the people saying that WebRTC is peer to peer only. There is no reason why you can't implement an application, run it on a server, and treat it as a 'peer' for the sake of webRTC when it is actually a server.
That said, we've looked into pulling the WebRTC implementation out of Chrome, but it is a huge task. Depending on what you want to do, you will likely only need to support a subset of WebRTC functionality (Data channel / unreliable for example if you're doing a multiplayer web game).
There might be a few implementations out there that have cropped up now, but last I checked there wasn't anything of note.
I have a WebRTC app, who work like this:
A person create a room with id
Peers connect to the room
Currently, each peer (also the session's host) share his stream to others.
But I'd like that the host share his stream to server, without sharing directly to peers.
Then what you need is not actually WebRTC, which is a peer-to-peer solution, but a streaming server.
since few days I'm trying to build a basic webRTC Videochat. I've got some Demos running localy, even via LAN. But now I want to build one by my one at the really basics without so much overload some Demos come with.
But I still don't get a complete peer connection.
Eg. this example seems to be broken, because I can't "createSignalingChannel();" w3.org/TR/webrtc/#simple-example
Some other examples (https://webrtc-experiment.appspot.com/) want me to link their scripts, but I wont do this, because I want to understand the magic of the peer connection and how to get a handshake between 2 browsers.
I also explored examples with the Google App Engine but thats not what I want.
I want to run it in really easy JS and HTML just on the minimum of what is neccessary.
Here is my code:
https://github.com/mexx91/basicVideoRTC EDIT: Should work now
So what will I have to add to get an handshake and peer connection, so that I can send eg. the mediaStream to eachother.
Thanks a lot!
createSignalingChannel() is only pseudo-code to illustrate the existence of a separate channel. You need for the initial connection handling a separate message channel.
You can achieve that with hosted services like Pusher, Brightcontext or PubNub, or you can host your own backend with open-source projects like socket.io or SignalR.
Then you just need to send the offers, answers and iceCandidates through your separate channel.
List of Realtime Services: http://www.leggetter.co.uk/real-time-web-technologies-guide
Imagine a video conferencing web-app, which users A and B originally access from some webserver. Suppose that web app supports presence, so the web server knows who's currently on-line. Imahine the UI allows A to try and place a video call to B. Via say XMLHttpRequest(), A's browser informs the server this is wanted, and B's javascript pops up something saying that A wants to call B. No WebRTC has happened at all yet. But at this stage, A can indirecttly communicated with B by sending messages using e.g. XMLHttpeRequest. In WebRTC parlance, this is the "signalling channel". So, A and B can both interact with their ICE agents to discover candidate addresses, and SDP descriptions, and send these to each ot6her, via the server, over this signallinh channel. E.g. the web app on A calls a WebRTC API to get its ICE candidates, and packages these up as it sees fit, to send to B. B's reader receives this message from the server (e.g over a WebSocket or long poll) and hyence it can unpack this, and format as needed to send to the ICE agent on B, using the RTCPeerConnection object. Similalrly, SDP offer/answer can be sent betweent he two apps, and passe through into the ICE agnet in the browsers, to get agreed media formats etc. At that stage, media connections can get set uo by the browser (meida streams are added to the RTCPeerConnection initially (which aren't communicating, but whihc have attributes that can be queried to describe the codec etc, and when the API is asked to create an SDP description, it does that using these attributes, but adjust the IP address and port based on how the ICE agent on each local browser has figured out what addresses can reach that local browser / port (NAT traversal).