Are there limitations that would prevent from establishing a tcp/ip socket connection between two distant devices? (Not on the same WiFi LAN.)
[Edit]
I missed this one: http://msdn.microsoft.com/en-us/library/windowsphone/develop/hh202858(v=vs.105).aspx
I'll try it as it seems like the usual client/server snippet, but for WP, so that looks promising.
I don't think you can do this directly device to device due to dynamically assigned IP address visibility etc. You might have to expose a web service on the public internet for this purpose. Basically a web service both devices can see and communicate with. (ex. Heroku, Amazon AWS). One device can keep a blocking connection ("Comet") while the other device initiates connections.
Related
Is it somehow possible to tell the browser to choose a specific interface and ignore the Windows routing table?
I have the following problem:
I have a PPP dial-up, where I have to access some https websites, if I make my standard gateway point to this PPP interface, I can access the website.
But I don't want to add a specific route, as I connect to different devices, and all have different IP Addresses, so it is also possible that my local LAN Interface, where I have my internet access, has the same IP Address as the remote https address I need to connect to.
So I think the only way would be to somehow tell the browser to use a specific interface for all the traffic. Is this somehow possible, or possible with VB.net programming?
"Is it somehow possible to tell the browser to choose a specific interface and ignore the Windows routing table?"
No. As an application, the browser relies entirely on the OS's network stack to establish communication.
The point of a network is to allow many-to-many communication. So, if you do it right, you can use a single network adapter to communication with hundreds or even thousands of other nodes at the same time.
You could use a NIC to establish a PPPoE session to the Internet while at the same time communicating with your local network. However, this in turn requires you to connect the modem to the LAN as well which is not a good idea - you should either use two separate NICs or an Internet router.
WebRTC signalling is driving me crazy. My use-case is quite simple: a bidirectional audio intercom between a kiosk and to a control room webapp. Both computers are on the same network. Neither has internet access, all machines have known static IPs.
Everything I read wants me to use STUN/TURN/ICE servers. The acronyms for this is endless, contributing to my migraine but if this were a standard application, I'd just open a port, tell the other client about it (I can do this via the webapp if I need to) and have the other connect.
Can I do this with WebRTC? Without running a dozen signalling servers?
For the sake of examples, how would you connect a browser running on 192.168.0.101 to one running on 192.168.0.102?
STUN/TURN is different from signaling.
STUN/TURN in WebRTC are used to gather ICE candidates. Signaling is used to transmit between these two PCs the session description (offer and answer).
You can use free STUN server (like stun.l.google.com or stun.services.mozilla.org). There are also free TURN servers, but not too many (these are resource expensive). One is numb.vigenie.ca.
Now there's no signaling server, because these are custom and can be done in many ways. Here's an article that I wrote. I ended up using Stomp now on client side and Spring on server side.
I guess you can tamper with SDP and inject the ICE candidates statically, but you'll still need to exchange SDP (and that's dinamycally generated each session) between these two PCs somehow. Even though, taking into account that the configuration will not change, I guess you can exchange it once (through the means of copy-paste :) ), stored it somewhere and use it every time.
If your end-points have static IPs then you can ignore STUN, TURN and ICE, which are just power-tools to drill holes in firewalls. Most people aren't that lucky.
Due to how WebRTC is structured, end-points do need a way to exchange call setup information (SDP) like media ports and key information ahead of time. How you get that information from A to B and back to A, is entirely up to you ("signaling server" is just a fancy word for this), but most people use something like a web socket server, the tic-tac-toe of client-initiated communication.
I think the simplest way to make this work on a private network without an internet connection is to install a basic web socket server on one of the machines.
As an example I recommend the very simple https://github.com/emannion/webrtc-web-socket which worked on my private network without an internet connection.
Follow the instructions to install the web socket server on e.g. 192.168.1.101, then have both end-points connect to 192.168.0.101:1337 with Chrome or Firefox. Share camera on both ends in the basic demo web UI, and hit Connect and you should be good to go.
If you need to do this entirely without any server, then this answer to a related question at least highlights the information you'd need to send across (in a cut'n'paste demo).
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
My task is to write a Messenger program for both internal and external staffs, I actually made it. However, I thing this is really not a good approach by using the client software keep "check-mesg" from server. So I think I am just simulating the IM program.
I want to make the client app become a listening server, and let user p2p talking without a "mesg-centre" at the main server(unless offline mesg happen). The question is how do I tell the external user ( other client app ) my location while I am behind a router ?
Are those other IM programs running on the client machine as a server too? and how do they get through ?
Thanks in advance!
It's quite complicated to connect to systems behind a router and not always possible. A well-documented way to do this with UDP is the STUN protocol (used mainly for SIP-based VoIP). If it is not possible to get behind the router, you can only use a server in the open network as intermediator (some P2P systems also promote well-connected peers to such intermediators). SIP uses TURN for as intermediator protocol. SIP's protocol to find out the right solution for a client is ICE.
See also NAT traversal.
In my office building we have laptops on multiple floors all running a WCF Service. When WCF services communicate with each other, will a message for an out-of-range device automatically reach it by multi-hopping? Does WCF/the WLAN device driver handle this? Or do I have to detect if a device is not contactable/out-of-range and implement hopping in my own service?
As long as you have a connection from your WCF client to the service - yes, all avenues will be used. You shouldn't have to concern yourself with things like what network path your messages take - the network just has to be present and stable for the duration of a call ;-)
There's nothing in WCF to deal with this, really - this should be handled way lower in the network stack, by the driver or the OS.
Short answer
With WCF can do either or both of these:
Rely on an underlying protocol like IP to handle roaming
Use custom channel code that handles retries, roaming, etc the way you want it
No special mechanism for enhancing roaming is provided in the WCF classes Microsoft provides, but the framework itself is easily capable of supporting this seamlessly if you write or find a channel implementation to do this.
Full answer
WCF is not an on-the-wire protocol. It is a framework that allows you to communicate using a wide variety of protocols and network stacks. This allows you to use the same client and server code whether you are using HTTPS, raw TCP, named pipes, or any other protocol.
WCF ships with many channels in the box, and you can add your own. For example if you want to communicate over BlueTooth or IRDA, just create a new channel that talks these protocols and you can use your WCF services over it. These channels can also be found online or purchased from vendors.
Most networking today is done using the IP protocol, and if you are using WCF to communicate between desktop machines you will probably be using some protocol(s) on top of IP, for example TCP or HTTP. In this case, IP's normal routing rules will be used, so if the two machines can exchane IP packets you can communicate using WCF.
So if your WiFi access points allow seamless roaming you will be able to tap into that functionality using WCF.
If your WiFi routing doesn't have seamless roaming, you will have to do some extra work if you want to maintain a connection during roaming. Specifically you will need to create a channel that will respond to a closed connetion by re-resolving the server nane and retrying the request. Of course you will have to use DNS or another protocol so the server can update its registration as its IP address changes.
WCF is flexible enough to allow you to create such a channel and use it without your application code ever realizing it. But nothing like this comes in the box: You would have to build it, or download or purchase it.
it has nothing to do with WCF....
if there is a connection between the computers, on the IP, then the message will get through...