I am beginner level.
I try to broadcast data to the browsers in local area ( Same router by sending . . . 255 ).
I should implement the real time streaming service to the local level browsers.
But it will occur high traffic when the client browsers is increased.
To broadcast data, it seems to need UDP protocol.
But web browser base on TCP.
So i investigated the webRTC that is based on UDP.
But i don't really know how to use this.
Is it possible to broadcast the data to the web browser like chrome in local area ?
If not, why it is impossible to implement ? just for hazard of DDOS ? How can i solve this high traffic problem ?
( It really occur high traffic when each clients respond to every data from server (TCP) or the server send same data to the every client amount to number of clients (not broadcasting).
I just want to implement that the server just send one broadcasting datagram packet to the local area and each clients in local level receive same one data from the server but not respond to that. )
From a web app (not a modified web browser itself), you cannot create nor manipulate raw (UDP/TCP) sockets.The sandboxing and other mechanisms won't let you.
with webRTC, you will need to make an handshake, and use ICE.
=> You cannot push to a peer knowing only his IP/port
=> You have to have the receiver accept and acknowledge the transfer
you might have more change with WebSockets, but that requires additional mechanisms as well and not all parties will be able to support web socket (or accept the upgrade from http to WS).
For illustration purpose, you can see the work of Jess on a web-based bit torrent. He has exactly the same problems. https://github.com/feross/webtorrent
Related
As far as I search, All WebRTC handshakes are done through any signaling server [ HTTP, WebSocket, etc..] even through Mail or Whatsapp.
But I expect to connect without using any of them. Is there any way to archive this?
If yes, please give me a brief solution. ThankYou!
WebRTC (as implemented in browsers) requires signalling. You can choose a different medium for signalling such as a piece of paper or calling someone on the phone, but it needs signalling.
If you want to be able to initiate unsolicited connections you need to use TCP or UDP directly which (if we ignore NAT and firewalls) don't have that restriction.
Every time I set up WebRTC video call clients, it never works unless I specify a TURN server. No matter how many STUN servers I supply, it always falls back onto TURN. It could be the case that the people I have tested on all coincidentally happened to be behind symmetric NAT. The only time it doesn't fall back to TURN is when I test locally on my own network. Are STUN servers just very infrequently or rarely used? Or are they used more often and my experience just happens to be anomalous.
STUN servers get used very sparingly, during session setup, to help WebRTC endpoints behind NATs discover their public IP addresses. STUN services put a very small load on their server machines. They're similar to the "what's my ip?" websites on the internet.
TURN servers, when needed, relay the media data from endpoint to endpoint. All the video, audio, and media streams go up to a TURN server and then back down to a recipient. The TURN server load is higher. TURN service is only needed when endpoints cannot reach each other via direct peer-to-peer connections.
STUN isn't a substitute for TURN.
I am building this video teaching site and did some research and got a good understanding but except for this thing. So when a user want's to connect to another user, P2P, I need signaling server to get their public IP to get them connected. Now STUN is doing that job and TURN will relay the media if the peers cannot connect. Now if I write signaling server with WebSocket to communicate the SDP messages and have ICE working, do I need coTURN installed? What will be the job of the job of them particularly?
Where exactly I am confused is the work of my simply written WebSocket Signaling server (from what I saw in different tutorials) and the work of the coTURN server I'll install. And how to connect them with the media server I'll install.
A second question, is there a way to use P2P when there is only two/three participants and get the media servers involved is there is more than that so that I don't use up the participant's bandwidth too much?
The signaling server is required to exchange messages between peers (SDP packets) until they have established a P2P connection.
A STUN server is there to help a peer discover information about its public IP and to open up firewall ports. The main problem this is solving is that a lot of devices are behind NAT routers within small private networks; NAT basically allows outgoing requests and their response, but blocks any other "unsolicited" incoming requests. You therefore have a Catch-22 scenario when both peers are behind a NAT router and could make an outgoing request, but have nowhere to send it to since the opposite peer doesn't expose anything to make a request to. STUN servers act as a temporary middleman to make requests to, which opens a port on the NAT device to allow the response to come back, which means there's now a known open port the other peer can use. It's a form of hole-punching.
A TURN server is a relay in a publicly accessible location, in case a P2P connection is impossible. There are still cases where hole-punching is unsuccessful, e.g. due to more restrictive firewalls. In those cases the two peers simply cannot talk 1-on-1 directly, and all their traffic is relayed through a TURN server. That's a 3rd party server that both peers can connect to unrestrictedly and that simply forwards data from one peer to the other. One popular implementation of a TURN server is coturn.
Yes, basically all those functions could be fulfilled by a single server, but they’re deliberately separated. The WebRTC specification has absolutely nothing to say about signaling servers, since the signaling mechanism is very unique to each application and could take many different forms. TURN is very bandwidth intensive and must usually be delegated to a larger server farm if you’re hoping to scale at all, so is impractical to mix in with any of the other two functions. So you end up with three separate components.
Regarding multi-peer connections: yes, you can set up a P2P group chat just fine. However, each peer will need to be connected to every other peer, so the number of connections and bandwidth per peer increases with each new peer. That’s probably going to work okay for 3 or 4 peers, but beyond that you may start to run into bandwidth and CPU limits of individual peers, especially if you’re doing decent quality video streaming.
What I want is, basically, to create a connection between two different computers on same local network. But i want to do this by computers' local IP's. (like 192.168.2.23 etc)
This must be a totally local connection. no TURN or STUN Servers. I am not sure if this is possible. Because there are not much documentation/example/information about WebRTC.
So, how can I create a connection from my computer to another one just passing its local IP as parameter?
Update: To be more clear; imagine there is an html page contains some code that activates my camera and audio services. and another -almost same- page is open in other computer. Waiting a connection request... And there is a textbox in my page to type an IP belongs to other computer on my local network. type 192.168.2.xx and bingo! i have connection between me and other computer.
I want this process as IP based, because there may be more than 2 devices on the network. And all of them are possible devices to create connection. So i need to reach them by their IP's.
Any example code or explanation would be great! even if it tells that this is not possible.
Thanks
Peer discovery is a vital part in any WebRTC application. It's an expensive term for saying: "Hi, I'm computer 4 and I want to talk to you!".
See it as calling a friend over the phone. You need to dial his number first.
This part is not defined in the WebRTC standards. You need to implement this logic in your application. Once you know who you want to call, you need a way of exchanging vital information. This is called signaling, like flo850 put in his answer.
Signaling is needed before any peer-to-peer connection can be set up.
To come up with an idea for your use case of 7 devices in a LAN.
If you have these devices connected to for example a WebSockets server and are in the same channel.
The WebSockets server can be written to route messages to specific receivers.
Devices connected to the channel often are identified with some kind of ID, imagine you use the device's IP.
When you want to talk to computer 4 with IP 192.168.0.4 you send the exchange messages (signaling) on the channel to the receiver with ID, the IP of the device you want to connect with.
How to send the signaling (offer, answer) is described here with example code.
Hope this helps
Users usually sit behind NATs; that's why ICE concept implemented in WebRTC.
If both users are sitting behind same NAT; you can skip ICE servers by passing "NULL" parameter value over "RTCPeerConnection" constructor:
var peer = new [webkit|moz]RTCPeerConnection ( null );
Now, browser will use "host" candidates, also known as "local" candidates.
you still need a signaling server. During the ICE candidate search, your clients will exchange their local ip through this signaling server
Is it possible to use WCF discovery to access services that reside outside your local network ?
The short answer is no.
Discovery uses a UDP broadcast packet. You can discover anything that your UDP broadcast packet is allowed to reach. There is the catch, most routers, firewalls, and commercial switches block udp broadcast packets. You may be able to change the settings on your router where you connect to the next larger network (or internet), and you 'might' extend your discovery slightly. Again though, the very next switch or router you hit will most likely be set to block udp.
In this situation, most people design a "report in" server. This is one static place to which all other hosts and clients and pre-programmed to "report in" on startup. This one server keeps a table of where all hosts and clients are, and if one client wants to find a certain host, it asks this main server for the uri of the host its looking for.
EDIT:
Robin mentioned increasing the TTL (Time To Live) from the default of 1 to a higher number. Maybe this will help someone.
https://serverfault.com/a/619825/146341