I am using CO-TURN as my VoIP platform.
What I am looking for is that how can I get information about which users are in an ongoing voice call?
That depends on how you are creating sessions on turn server. You can check it from WebUi.
But your application server should manage all this not turn server as this is not role of turn server.
Related
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.
Is it possible to have Asterisk as the signaling server for WebRTC enabled mobile app. I have found that I need to create the signaling server in node.js. I was wondering whether Asterisk can do the job for me.
Also, does the WebRTC media pass through the signaling server (or) is it direct device to device media transfer.
Any help is appreciated. Thank you. :)
If you want WebRTC signal switching alone then you can use some websocket server like socket.io but it won't handle STUN or TURN.
If you want WebRTC signalling along with STUN,TURN and media recording then you can go with Kurento, Freeswitch, asterisk etc.
If you carry media over signalling server then you might consume more CPU and memory processing on server side. Try to keep peer-peer direct media to get better audio and video.
But in all these case you need to register your endpoints to the signalling server. So if you call B from A. the signalling server knows where B is!
Is it possible to have Asterisk as the signalling server for WebRTC enabled mobile app?
Of course. Just use SIP over websocket (RFC 7118). See this Asterisk WebRTC setup guide for the details. If you already have an Asterisk server, then there is no need to implement a separate signaling server in node.js.
Also, does the WebRTC media pass through the signaling server (or) is it direct device to device media transfer.
This is handled automatically by ICE. Ideally your goal should be to have as many peer to peer media as possible to minimize the server load. However in some situations a relay is required which can be the Asterisk server itself or a separate TURN server. Also make sure to set the STUN correctly (that is required for STUN to find the direct path)
I'm trying to achieve peer to peer video conference using google stun server.
I can connect anyone by stun server randomly.Because stun gives multiple and random addresses and connect with it.
But is there any way to connect specific peer by stun server for a login based system or room based system?
I want to achive something like - https://apprtc.appspot.com/
You need to design your signalling method (this is up to the application developer), which is independent of STUN.
WebRTC does not specify the mechanism for signalling. Signalling is the method whereby users discover each other and establish that a call (media streams between two peers) is going to take place.
The 'discovery' process could involve a registration-based system (eg using SIP proxy) or room based where two users have access to a 'room' (by knowing the credentials or some means of authentication). Once two peers have found each other, their browsers then need to share and negotiate network topology and media capabilities to ensure that the streams can reach the intended destination and can be encoded/decoded properly.
I want to make an app that displays new data whenever they arrive inside a folder via xml. I want to use html5 web sockets but I am confused on how it should be done. I am using xaamp on my machine for development. Do I have to install another server to use websockets? Is apache as it is compatible and if yes how do I make the connection with the client. Thank you in advance..
Your options are:
Use something like mod_websocket, as pointed out by Phillip Kovalev. Or pywebsocket. You could also try PHP WebSocket.
Use a dedicated self-hosted realtime web technology for realtime communication between server and client. If you do this you'll also need to define a way of application to realtime web server communications - normally achieved through message queues.
Use a hosted realtime web solution and offload the realtime push aspect of your application.
There are concerns about using Apache with this type of technology since this technology maintains long-running persistent connections between the server and client and Apache isn't know to be too great at this. So, the best solution may be to:
Go with a 2nd dedicated realtime web server in conjunction with using Apache as your application server
Use a self-hosted realtime web server that has the ability to handle many concurrent connections
Use a hosted service along with your Apache application server.
If you don't expect many concurrent connections or if you are just trying out the technology then it's possible that Apache alone will be all you need.
Look at mod_websocket. It supports latest and commonly implemented by browsers vendors protocol version.