I am trying to test sip capabilities of firewalls using webrtc. However I noticed using the servers needed for webrtc (stun turn websockets etc.) will give me a false positive in that it won’t catch nuanced issues with the ALGs. For reference this is being done from a chrome app so I can’t just run a native sip stack in the browser.
My Question: can I leverage webrtc to just send sip(invite, options, register) and not use any other methods that would get around the firewall?
Your question doesn't make sense because WebRTC doesn't use SIP - SIP is a signaling protocol, and WebRTC doesn't do signaling. What that means is that SIP can be used to establish a WebRTC connection, but they are mutually exclusive.
SIP is sent over a data connection, like a hard line from a phone to a PBX or a websocket from a browser to a server.
It is possible to set up a WebRTC connection using out of band mechanisms, but then that wouldn't be SIP.
Actually there might be a way around that.
Use the signalling server to to do any sort of preconfigurations you might want to do before setting up the peer connection. This would allow you to specify codecs and resolution of the feed as a SessionDescription before hand or even check if the other peer is capable of WebRTC or not.
I'd recommend Socket.io =D
Related
I am trying to use WebRTC to implement browser to browser communication.
I want to allow communication between two browsers running on two different computers in the same LAN.
Can somebody confirm is this possible and if so, how.
I tried looking up the demos and seems they all have examples for application running in the same page. But I would like to connect to a peer computer using some IP address.
Regards
I have just solved your problem using https://github.com/feross/simple-peer
Yes, it is possible to create a Peer to pear connection using PEERJS. PeerJS simplifies WebRTC peer-to-peer data, video. PeerJS wraps the browser's WebRTC implementation to provide a complete, configurable, and easy-to-use peer-to-peer connection API within a LAN Network. PEERJS
Is there a way to stream a video and audio on a website just to the clients, using a camera installed on the server - for instance, like youtube does ?
I've started reading webrtc, but if I use webrtc I should create a stun/turn server and other things, which for one way stream I think is not necessary (this is just my understanding of the things..) because I don't need anything from the clients, literally, neither their video, or audio..
So is there a way to achieve this using html5, streaming just in one direction:
server (camera) -> clients
Is there something about this out there, or should I stick with webrtc ?
I'm going to explain a possible solution for this scenario, there might be others, but I hope mine gives you a rough idea of how you could do it and a start point to explore more about the amazing possibilities of WebRTC. Please let me know if something is not understood.
So, WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. Sweet, that is: WebRTC has a quite good browser support (not in every browser though, Safari just started supporting it a month ago with Safari 11). But in this case we want to use WebRTC in the server side. At the end of the day we can still think about peer-to-peer real time communication, where one of our peers is the server.
I don't know if you are familiar with Node.js, but I recommend you to write your Server app with it (<3 Javascript!):
There are a few libraries that wrap WebRTC functionality to be used in the server side, like node-webrtc and node-rtc-peer-connection.
But I recommend you to to take a look at electron-werbrtc, since
the others might be using deprecated methods or be incomplete.
electron-webrtc runs a headless Electron client in the background to
use Chromium's built-in WebRTC implementation. So with it you should
be able to access the Camera in your server and create a stream to be
served to the other peer (the browser).
All above would be the WebRTC related tasks, in this case: streaming video peer(server)-to-peer(browser).
Now, let's talk about signaling process, stun and turn.
Signaling: imagine now a scenario peer-to-peer with 2 browsers, they want to establish a direct connection and stream video and audio between each other. But they don't know each other, like if I don't know your home address, I can't send you a letter. So they need a service that helps them know each other, so they can have the other's IP. This should be done by what is called "a signaling server". If somehow you know the other peer IP, you wouldn't need a signaling server.
STUN/TURN: the scheme above works perfectly in a local area network where each peer has its own IP address and there are no firewalls and routers between them. But otherwise, you can have peers behind a NAT or firewalls, and then your signaling server won't be able to make both peers to discover themselves. If you have peers behind a NAT, you'll need a STUN server, and if you have peers behind firewalls you'll need a TURN server. This is a bit simplified, but I just want you to have the general picture of when you might need STUN/TURN servers.
To better understand Signaling, STUN and TURN, there is a very graphic article that explains them perfectly.
Now, for your scenario:
I think you prob don't need STUN/TURN servers and also you prob don't need to implement the signaling process, because the browsers that are supposed to receive the stream from the server will know that server address, right? So they can establish a WebRTC connection with it.
EDIT: it is likely that you will need to implement some sort of handshake between the server and the clients (browsers), so this will be the signaling process. This is not part of WebRTC and this is why you need to implement it yourself. As I said, it is the way 2 peers can discover each other, but they also exchange information as their local media conditions, like codecs, resolutions they can handle, etc. For your case, your signaling server could be hosted in the same server you use to strea: you can build a small node.js app that runs there and that manages all the signaling process easily, it is not a big deal. I recommend you to read this article, and specially the section "How can I build a signaling service?". In general all WebRTC articles from that site are very helpful.
Does this make sense to you? I think with it you can start digging a little bit more and see if with this is enough or you need to implement more stuff. Hope it helps!
can HTML5 communicate with Java Serversocketchannel?
if possible can anyone tellme the details.
thank you in advance.
I'm assuming that you are talking about WebSockets and not some other protocol (Flash, Java applet and Silverlight native sockets, or XMLHttpRequest connections). WebSockets are an HTTP family spec from IETF and not related directly to HTML5 (though they are both in the extended family of next-gen web standards).
Browser WebSocket implementations can only talk to servers that deliberately support the WebSocket protocol. You can certainly write a server that supports the WebSocket protocol using a ServerSocketChannel, but WebSocket will not be able to connect to an arbitrary service that was written (using ServerSocketChannel or not) without the WebSocket protocol in mind.
This is a deliberate security measure to prevent web browsers being forced to connect to non-web-related services (eg to port 25 to send spam).
If you want to write a WebSocket protocol layer on top of ServerSocketChannel you'll need to put a non-trivial amount of work into implementing the spec. It would seem more sensible to re-use an existing library.
I am aware that an early draft of HTML5 specified peer-to-peer connections using the PeerToPeerConnection() constructor. However, this was replaced by WebSocket which, to my knowledge, does not support peer-to-peer connections.
In addition, the device element specifies a peer-to-peer connection interface, however no web browsers have implemented it yet.
Opera Unite allows Peer to peer (basically gives your browser web server, file system and nat-traversal capabilities), but this is Opera only.
It's not exactly P2P, but you can use proxy server using push mechanism
http://en.wikipedia.org/wiki/Push_technology
For example if you want to send data from Browser A to Browser B, you send that data to proxy server and that server push it to Browser B.
This question already has answers here:
Do websockets allow for p2p (browser to browser) communication?
(5 answers)
Closed 1 year ago.
Using HTML5 Websockets, can I open a socket directly between 2 clients (browsers) (assuming I know their IP addresses and there are no NAT traversal issues). I want to send some data directly from one browser to another browser, essentially creating a P2P network. Everything I've read so far on the web, websockets are all used to communicate between client and server, but nothing about client to client.
No
Already answered here for more details: Do websockets allow for p2p (browser to browser) communication?
Update:
The WebRTC protocol and API is making rapid progress and allows a Data Channel to be established between two peers (you still may need a STUN/TURN server for the initial NAT traversal and setup).
From past to future: Pick your implementation.
Most modern browsers support peer to proxy to peer using Websockets.
Peer to peer is only available using certain browsers plug-ins for example Opera Unite and FireFox Pow.
Major browsers are in the process of implementing WebRTC which will allow peer to peer.
If you just don't want to implement server your self you can use https://httprelay.io . No additional libraries required, all communication based on simple AJAX calls.