I need to stream several "channels" (by channels I'm thinking of radio channels, so playlists might be more appropriate) of queued MP3 files to around 200 clients over the internet from a Windows 2008 R2 / IIS 7 web server. Encryption of the stream is not a requirement.
I need some way to ensure each client can only stream one channel at a time. I was thinking of restricting by IP address, and would welcome any suggestions on how I might go about this, or if there may be a better way.
For my clients I assume I'd need a player on the client end that could facilitate "logging in". Ideally I would be able to stream to windows xp/7/8, mac and android clients.
I would need to be able to log in remotely and control each channel / playlist from a finite list of MP3 files hosted on the web server.
I'm wondering if there any off the shelf products that I could use to do this. If not, I'm stuck on what would be the best way to go about this.
I've briefly read up on shoutcast, auto dj, streamcast and ice cast, but I don't have any experience with these solutions, and can't find any information about how to implement the security requirements I have (limiting access to one stream per client / ip address).
IP addresses do not uniquely identify users. There are plenty of situations where NAT comes into play, and you can have hundreds of users all behind the same public IP address.
What you need to do is have another method for identifying users. Assuming you don't want to require accounts, you can use a session ID.
Basically, you assign an ID with a cookie to a browser. When the user clicks a link to launch their audio player, the session ID is passed in the URL to the stream. With this method, it doesn't matter if the browser itself or a separate audio player is used to play the stream. It is up to the streaming server to then accept or reject the request.
It is common to accept any new request and disconnect the old when a new stream starts. Icecast doesn't support this sort of thing natively, but it does provide an API of sorts with the admin interface that you can use from your own scripts to get this behavior.
Alternatively, I have written a server called AudioPump that provides similar functionality. It isn't generally available yet, but please contact me at brad#audiopump.co if you're interested.
Related
I was following THIS tutorial in order to implement video calling using WebRTC. The example allows people of the same group to communicate using video or audio and it works well. I was wondering if you could Video call people who are on different pages of the same website without having to be on the specific 'video call page' only?
What is the way to allow incoming calls even when you are not joined in a group? I believe it must be a group request (Sending a request to join from Person A to Person B). Is this possible using WebRTC?
WebRtc is not based on "who is connected to what page".
The connections only happen through your signalling server logic. This means that any webrtc peerconnection can connect to anyother peerconnection as long as your signalling server relays the connection startup logic(SDPs and Ice candidates).
The webpage is only a way to display the media that you have, the javascript you wrote will run on any page you want.
After knowing about some great features of WebRTC, I thought of using WebRTC one to one audio/video calls in my web application. The web application is for many organizations/entities of a category who can register and keep recording several records daily for their internal working and about their clients. The clients of these individual organizations/entities also have access to the web application to access their details.
The purpose of using WebRTC is for communication between clients and organizations. Also for daily inquires by new people to these organizations about products and services.
While going through articles on google etc. I found broadcasting or one to many calls requires very high bandwidth to users if we don't make use of Media Server.
So what is the case for one to one calls?
Will it affect the performance of web application or bring any critical situation if several users are making audio/video calls(one to one) to each other simultaneously as a routine?
The number of users will be very large and users will be recording daily several entries as their routine work. But still it is manageable and application will be running smoothly but I am not sure about the new concept WebRTC. Will it require a very high hosting plan? Is using WebRTC for current scenario suitable or advisable?
WebRTC by its nature is Peer-to-Peer. Meaning that the streaming data is handled CLIENT side. All decoding, encoding, ICE candidate gathering/negotiation, and media encrypting/transmitting will happen on the client side and not on server side. So, you will be providing the pages, client side JS, and some data exchange(session negotiation signalling) but all in all, it is not a huge amount of work. It should be easily handled without having to worry about your host machine being over worked.
All that said, here are the only a performance concerns that would POSSIBLY affect your hosting server.
Signalling session startup, negotiations, and tare down. This is very minimal(only some json data at the beginning of a session). This should not be too much of a burden but you should be aware that if 1000 sessions start at the same time, you will have a queue of messages to direct to the needed parties. How you determine the parties, forward the messages, and what work you do server side could all affect performance. If written smartly(how to store sessions, how to forward messages, etc.) should not be a terrible burden.This could easily done with SignalR since you are on ASP.NET or you could use a separate one running Node.js(or the same box, does not matter) if you so desired.
RTP TURN relay if needed. This will probably be through a different server(or the same one as your hosting server if you want). For SOME connections, a TURN server is needed and any production ready WebRTC solution should take this into account. Here is a good open source turn server. Bandwidth usage here could be very high as RTP packets are sent to this server and the forwarded to the peer in the connection.
If you are recording the streams, you may have increased hosting traffic depending on how you implement it. Firefox supports client side recording of the streams but Chrome does not(they say it is in the works currently). You could use existing JS libraries to record the feeds client side and then push them anywhere you want. You could also push all the data through a MediaServer that will mux, demux, and forward the data to be recorded anywhere you like. Janus-Gateway videoroom is a good lightweight example of a mediaserver.
Client side is a different story.
There are higher level concerns in the Javascript. If you use one of the recording JS libraries, this is especially evident as they do canvas captures numerous times a second which are a heavy hit and would degrade the user experience.
CPU utilization by the browser will increase as the quality of the video being streamed increases. This is rather obvious as HD video frames take more CPU power to encode/decode than SD frames.
Client side bandwidth usage can also be an issue. Chrome and Firefox try to modify the bitrate of each video/audio feed dynamically but the video Bitrate can go all the way up to 2 Mbps. You can cap this in Chrome( by adding an attribute in the SDP) but not in Firefox(last I checked) as of yet.
I want to create a program in which a user enters data and saves it on a text file. For example they enter a name and their age and it saves it and loads it. The thing is that i want this program to be available to the user at all times on any computer which means having the program online.
Do i need to make the program have networking capabilities? Or can i just have a server or host computer to hold the program and have the client access that directly?(like using the spreadsheets from google).
I don't know what you mean by "do I need to make the program have networking capabilities". If you choose to avoid the web site route then you need to have a server and client side app. If you want the program to be able to save data to a server then you need to do some socket programming.
To create a server, you need to:
create a socket
bind the socket to an address and port
listen for incoming connections
wait for clients
accept a client
send and receive data
To create a client, you need to:
create a socket
connect to a server
send and receive data
Hope this helps!
There are may ways to accomplish your goal, but the route that is probably easiest and most useful is to create a web page that implements this functionality.
Some of the many benefits include
No special software to install. Just requires a web browser.
Runs on any platform (including mobile) that has a web browser.
No software updates to push out to users. Update your website, and everyone gets the latest code.
I'm making a network game (1v1) where in-game its p2p - no need for a game server.
However, for players to be able to "find each other", without the need to coordinate in another medium and enter IP addresses (similar to the modem days of network games), I need to have a coordination/matching server.
I can't use regular web hosting because:
The clients will communicate in UDP.
Therefore I'll need to do UDP Hole Punching to be able to go through the NAT
That would require the server to talk in UDP and know the client's IP and port
afaik with regular web hosting (php/etc) I can only get the client's IP address and can only communicate in TCP (HTTP).
Options I am currently considering:
Use a hosting solution where my program can accept UDP connection. (any recommendations?)
UDPonNAT seems to do this but uses GTalk and requires each client to have a GTalk account for this (which probably makes it an unsuitable solution)
Any ideas? Thanks :)
First, let me say that this is well out of my realm of expertise, but I found myself very interested, so I've been doing some searching and reading.
It seems that the most commonly prescribed solution for UDP NAT traversal is to use a STUN server. I did some quick searches to see if there are any companies that will just straight-up provide you with a STUN hosting solution, but if there even were any, they were buried in piles of ads for simple web hosting.
Fortunately, it seems there are several STUN servers that are already up and running and free for public use. There is a list of public STUN servers at voip-info.org.
In addition, there is plenty more information to be had if you explore SO questions tagged "nat".
I don't see any other choice than to have a dedicated server running your code. The other solutions you propose are, shall we say, less than optimal.
If you start small, virtual hosting will be fine. Costs are pretty minimal.
Rather than a full-blown dedicated server, you could just get a cheap shared hosting service and have the application interface with a PHP page, which in turn interfaces with a MySQL database backend.
For example, Lunarpages has a $3/month starter package that includes 5gb of space and 50gb of bandwidth. For something this simple, that's all you should need.
Then you just have your application poll the web page for the list of games, and submit a POST request in order to add their own game to the list.
Of course, this method requires learning PHP and MySQL if you don't already know them. And if you do it right, you can have the PHP page enter a sort of infinite loop to keep the connection open and just feed updates to the client, rather than polling the page every few seconds and wasting a lot of bandwidth. That's way outside the scope of this answer though.
Oh, and if you're looking for something absolutely free, search for a free PHP host. Those exist too! Even with an ad-supported host, your app could just grab the page and ignore the ads when you parse the list of games. I know that T35 used to be one of my favorites because their free plan doesn't track space or bandwidth (it limits the per-file size, to eliminate their service being used as a media share, but it shouldn't be a problem for PHP files). But of course, I think in the long run you'll be better off going with a paid host.
Edit: T35 also says "Free hosting allows 1 domain to be hosted, while paid offers unlimited domain hosting." So you can even just pay for a domain name and link it to them! I think in the short term, that's your best (cheapest) bet. Of course, this is all assuming you either know or are willing to learn PHP in order to make this happen. :)
There's nothing that every net connection will support. STUN is probably good, UPnP can work for this.
However, it's rumored that most firewalls can be enticed to pass almost anything through UDP port 53 (DNS). You might have to argue with the OS about your access to that port though.
Also, check out SIP, it's another protocol designed for this sort of thing. With the popularity of VOIP, there may be decent built-in support for this in more firewalls.
If you're really committed to UDP, you might also consider tunneling it over HTTP.
how about you break the problem into two parts - make a game matcher client (that is distinct from the game), which can communicate via http to your cheap/shared webhost. All gamers who wants to use the game matching function use this. THe game matcher client then launches the actual game with the correct parameters (IP, etc etc) after obtaining the info from your server.
The game will then use the standard way to UDP punch thru NAT, etc etc, as per your network code. The game dont actually need to know anything about the matcher client or matcher server - in the true sense of p2p (like torrents, once you can obtain your peer's IPs, you can even disconnect from the tracker).
That way, your problems become smaller.
An intermediate solution between hosting your own dedicated server and a strictly P2P networking environment is the gnutella model. In that model, there are superpeers that act like local servers, having known IP addresses and being connected to (and thus having knowledge of) more clients than a typical peer. This still requires you to run at least one superpeer yourself, but it gives you the option to let other people run their own superpeers.
What all would be the requirements for the following scenario:
A GSM modem connected to a PC running
a web based (ASP.NET) application. In
the application the user selects a
phone number from a list of phone nos.
When he clicks on a button named the
PC should call the selected phone
number. When the person on the phone
responds he should be able to have a
conversation with the PC user.
Similarly there should be a facility
to send SMS.
Now I don't want any code listings. I just need to know what would be the requirements besides asp.net, database for storing phone numbers, and GSM modem.
Any help in terms of reference websites would be highly appreciated.
I'll pick some points of your very broad question and answer them. Note that there are other points where others may be of more help...
First, a GSM modem is probably not the way you'd want to go as they usually don't allow for concurrency. So unless you just want one user at the time to use your service, you'd probably need another solution.
Also, think about cost issues - at least where I live, providing such a service would be prohibitively expensive using a normal GSM modem and a normal contract - but this is drifting into off-topicness.
The next issue will be to get voice data from the client to the server (which will relay it to the phone system - using whatever practical means). Pure browser based functionality won't be of much help, so you would absolutely need something plugin based.
Flash may work, seeing they provide access to the microphone, but please don't ask me about the details. I've never done anything like this.
Also, privacy would be a concern. While GSM data is encrypted, the path between client and server is not per default. And even if you use SSL, you'd have to convince your users trusting you that you don't record all the conversations going on, but this too is more of a political than a coding issue.
Finally, you'd have to think of bandwidth. Voice uses a lot of it and also it requires low latency. If you use a SIP trunk, you'll need the bandwidth twice per user: Once from and to your client and once from and to the SIP trunk. Calculate with 10-64 KBit/s per user and channel.
A feasible architecture would probably be to use a SIP trunk (they optimize on using VoIP as much as possible and thus can provide much lower rates than a GSM provider generally does. Also, they allow for concurrency), an Asterisk box (http://www.asterisk.org - a free PBX), some custom made flash client and a custom made SIP client on the server.
All in all, this is quite the undertaking :-)
You'll need a GSM library. There appear to be a few of these.
e.g. http://www.wirelessdevstudio.com/eng/
Have a look at the Ekiga project at http://www.Ekiga.org.
This provides audio and or video chat between users using the standard SIP (Session Initiation Protocol) over the Internet. Like most SIP clients, it can also be used to make calls to and receive calls from the telephone network, but this requires an account with a commercial service provider (there are many, and fees are quite reasonable compared to normal phone line accounts).
Ekiga uses the open source OPAL library to implement SIP communications (OPAL has support for several VoIP and video over IP standards - see www.opalvoip.org for more info).