I have just started out with learning WebRTC for implementing audio and video application and know there are various Public stun servers available for peer connection. But i am a bit confused can I use these Public servers for a commercial application?
Also I would like to know if there is any tutorial or guide available from where i can understand how to make and deploy my own stun or turn server if i want to create a commercial app?
Whether you can use public STUN servers for commercial applications depends entirely on the licensing/Terms of Service agreement of the operator of said servers. Peruse those if available. If not indicated otherwise, I wouldn't distinguish "commercial" use from any other use.
There are many many implementations of STUN/TURN servers available you can set up yourself on any machine you happen to have (in practice that probably means an instance on AWS, Azure or the like). Search for and pick one you like. STUN servers use relatively little resources, while TURN servers typically need powerful CPUs and fast internet connections to be useful (they must relay the entire video stream as quickly as possible).
Operating such a server yourself may become expensive, depending on your usage. Using a commercial provider for TURN servers may be the better option; personally I've had good experiences with Twilio in this regard, but do shop around for other offerings.
Related
I'm managing a rather widely distributed software application in a semi industrial environment. The software at its heart is based on SOA and employs OPC-UA to make communications between important processes (on local or LAN-based machines) possible. These processes are either a server (e.g. an outer network management server, hardware managers server etc.) or a client (customer panel) or both (servers talking to each other).
OPC-UA has the following problems:
Configuring and maintaining the configurations is a hard job (just Config file settings takes lots of time)
Security measures are too much detailed for my needs (certificate management and sudden invalidation of certifications on customer systems)
Modeling and networking overheads in the library make it hard to work with in my communications (high data rates usually ends in server and client disconnecting)
Unspecified and weird errors like UA Discovery Server stopping to work or respond, etc. which I have reported to OPC GitHub forum many times.
Troubleshooting in internal parts of OPC UA is nearly impossible.
Overall, for me its performance and stability are not reliable enough. I am willing to sacrifice features for better performance and reliability. I've even considered to write sockets from bottom-up for my inter-process (IPC) needs. This way I could at least trace errors to their core. Since I do not need its most advertised feature (aka PLC support) I'm desperate to find a good alternative for it. My main requirements are:
OPC-UA like Data Modeling support that enables me to provide a clean interface to customers and other teams (something like IDL).
Publish/Subscribe, Remote Commands, Update Notifications and Node Based Behavior.
Tough Security is not my concern as my network is closed.
High performance for data rates up to 1Gbps (this could mean UDP support).
I am entirely working in .NET framework. So C# support of OPC-UA is a great help for me.
I've looked at DDS (lacks commands and Update Notifications) and WCF (lacks cross platform support) and many more.
This link also notes about MQTT: Alternative to OPC-UA
What about Google's gRPC + protobufs?
https://grpc.io/
I would like to implement a video / audio call feature from a browser. The goal is to allow two users to communicate remotely without having to install a third part (when I say third part, I'm talking about a software or an extension on a browser).
I know WebRTC, which is very popular today and free. However, it is very difficult to implement and the documentation is difficult to understand (not very easy for a beginner).
Here is the official webRTC documentation, and honestly, where to start? https://webrtc.org/start/
If you have an experience about WebRTC, is it possible to share with positive or negative points? This would be very useful for the community.
Moreover, if you have experience with another library, I think it would be interesting to hear it.
There is no other way to develop a call service in a website without the use of WebRTC today.
The alternatives are:
Use WebRTC
Use Flash (which is... dead)
Use a plugin (which is... dying as a mechanism in browsers)
Use an app you download (not exactly a service in a website)
Node.js is the way to go, but you will need to learn some new technology, especially when it comes to the backend.
The servers you will need are:
1. The traditional web application server
2. A signaling server (the one you plan on using Node.js for - you can use that for the web application server as well)
3. A STUN/TURN server (for NAT traversal)
4. Maybe a media server, depending on your use case
For some alternative open source and commercial products, you can check this WebRTC Developer Tools Landscape
Is WebRTC going to be free for web developers to set up video calls on web pages?
why does Twilio having pricing 25c per mins for video calls,
is it going to be too expensive for the small guy to mange video calls on web hosting servers?
any advice from anyone deep into WebRTC already?
Some of the comments above are not well informed.
Someone wrote, since the bandwidth needed in case of media relay is higher as well. This is not entirely true, transmission happens between Peers(Browsers), servers are used just for signalling(relaying IP addresses of connecting peers and some more info), you can ROUTE your transmission from central server(for fail overs), but can surely do without it for free.
WebRTC is Free and you can setup the whole thing on your own without having to shell out even a penny. It is a bit hard and mitigating fail-overs is really difficult, but you can certainly do it for free.
Tokbox or Twilio charge money because these tools abstract some very rigid complexities of setting up, running and managing fail-overs in a WebRTC application.
In TokBox's Case:
You don't need to setup STUN, TURN servers, you don't have to worry
about integration with android or IOS clients, they provide a plugin
for IE too, so out of box you get everything and you just have to
concentrate on your application logic rather than WebRTC nuances.
This is a big plus.
Both RELAY and ROUTED schemes came in the box hence you can write
fail-over scenarios if RELAY communication fails. Although there are
some good JavaScript based frameworks that do this in a much cleaner
manner.
It adds slew of other goodies which help in building android and IOS
clients without any pain.
STUN or TURN Servers are used only for Signalling Purpose, and this signalling happens before any actual transmission. This signal is very small and carries the IP address of both the browsers(machines running browsers). For Transmission the communication is done between Browsers(Peer to Peer) themselves, so no server is involved.
Your relay is not happening from a central server so you don't have
to pay for the outgoing bandwidth cost.
To Setup Turn Server,
Use this server, build it and put it into a Rackspace/Amazon Web
Services instance and you are Good with your TURN
Server. That is It, setup your application and have fun with WebRTC
for FREE.
rfc5766-turn-server
If you wish to Use some more free framework to ease yourself more, check out: EasyRTC and PeerJS
Enjoy Developing with WebRTC....
Twilio developer evangelist here.
Your link at the end of your question points to our WebRTC page, which currently talks about the product Twilio Client. Twilio Client briefly is a way that, using WebRTC within browsers and mobile applications you can make phone calls to real phone numbers. This product does not allow you to conduct video calls.
Twilio Client has a cost because of the ability to call out from a browser to a telephone number. The cost is not in the WebRTC portion, but delivering those minutes to the other leg of the call.
Notably, it's not 25 cents ($0.25) a minute, instead it is just a quarter of a cent ($0.0025) a minute.
With regards to video calls with WebRTC, you can now access the public beta of Twilio Video, a platform to make setting up WebRTC calls much easier.
Twilio Video costs for the signalling infrastructure and you can see the prices here. If a WebRTC connection requires a TURN server to relay the media, that also costs per gigabyte of transfer. Usage of the STUN server is free, the costs for the TURN relay are available here.
Please get in touch with me at philnash#twilio.com if you have any other questions about WebRTC.
WebRTC is a technology placed in a browser. It requires backend infrastructure to support it - specifically, STUN and TURN servers as well as signaling servers.
This boils down to the fact that you pay for WebRTC - same as you pay for hosting your website on a server. The price is higher, since the bandwidth needed in case of media relay is higher as well.
To understand more about WebRTC and how it works (as well as why there's a price tag associated with services such as Twilio for it), you can check this free report: https://bloggeek.me/webrtc-business-people/
WebRTC is already free for developers to use. When we added WebRTC to our product, we used this example code, which made it very simple to build a WebRTC client:
https://shanetully.com/2014/09/a-dead-simple-webrtc-example/
Google and Mozilla provide free STUN servers, and it is easy to set up a TURN server. Most clients will be able to connect via STUN, so you won't end up using too much bandwidth on your TURN server.
To set up your own TURN server, coturn seems to be the easiest to set up:
https://github.com/coturn/coturn
Make sure you read the "WEBRTC USAGE" section in the README.turnserver file.
"STUN or TURN Servers are used only for Signalling Purpose, and this signalling happens before any actual transmission. This signal is very small and carries the IP address of both the browsers(machines running browsers). For Transmission the communication is done between Browsers(Peer to Peer) themselves, so no server is involved."
if that is the case, then you should be able to do this on a standard web server using Java/php. PHP will get the IP address of the guys connected to it. Then its just a matter of storing them in MySQL, then making a javascript that would run when the user go to that page in the site.
I've been looking for a solution around using a VPS because running a dedicated server for signaling is like golfing with a Ferrari instead of a golf cart. I still don't think node is efficient. Its single threaded. so node's fararri can only go 5mph.
Since they went to the web site, the php service already can get their ip address what else does it need? All of the above solutions so far require you to pay for a dedicated app to run on a server connected to the web separately for what 5k of data? What a waste of electrons.
But I'm going to start a new thread that is going to be based on getting webrtc without the buy a "VPS" because we want a VPS-less solution.
Many applications use the following model:
Browsers or other clients interact with application servers.
Application servers (web servers or RPC servers) interact with data store servers (SQL servers or non-SQL storage).
For internet applications, they need application servers because they must keep simple feature on data servers for performance. But I can't see why they need application servers on intranet.
For example, can we develop an Adobe AIR application, which directly connect to a PostgreSQL server? I guess we can deploy a center PostgreSQL server which has many stored procedures and set strict permission, and let the Adobe AIR application fetch (and modify) data only by invoking the stored procedure.
Why don't the most of applications choose a simplier solution?
In general, there is no reason why you couldn't get an independent application to talk to a PostgreSQL server directly. Some applications do this and it works fine.
I'm not familiar enough with Adobe AIR to say whether it's possible in this context. In principle, if you can get a PostgreSQL driver, or if you can write your own using TCP sockets (the PostgreSQL network protocol is documented in details in the official documentation), you could certainly connect directly.
This being said, having a form of application server between the end-client and the database server isn't purely for performance.
Web-based development allows the SQL queries to be controlled by the server. Instead of exposing complete SQL access, you expose the features that the client can use. If you need to tweak the queries later (bug, change of data structure, ...), you can do this rather centrally on your application server, without having the need to deploy a new version of the client to each user.
Of course, you can do some abstraction like this user server programming directly, but this isn't suitable for all applications. This may depend on what other features your application needs, for example if it needs to make use of a library programmed in another language. You can use some procedural languages bindings, but it's not always suitable: pl/Python is an "untrusted" language (which may cause security problems) and pl/Java needs a external add-on, for example.
In addition, not all applications are ultimately reserved for intranet usage nowadays. It often makes sense not to restrict yourself to intranet usage when you start designing an application.
I initially started with a direct access design and quickly found it useful to move to an application server where I talked to the DB via web services. Reasons included:
Handling DB restart, local connection loss, client IP address change, etc is much easier when you're talking to the DB over a stateless protocol like HTTP. This is more of an issue for remote workers.
Transactions are clearly demarcated and isolated in server-side transactional methods (I used EJB3 and container managed transactions)
It's much easier to add new clients like a phone app as they can share more of the code and business logic. Stored procedures in the database are very useful, but can be limited and occasionally frustrating.
Some tools/languages don't have built-in tools for talking to PostgreSQL directly, but can easily talk to a RESTful web service with XML or JSON request/response format.
DB admin is easier if you're dealing only with a single application server connection pool
The main downside is of course the extra layer means extra work and extra maintenance.
You can, but...
Browser languages/libraries tend to have poor database support
What happens when someone wants to use this application remotely?
If you're not talking about browser-based applications, then that is exactly what many do. There are plenty of traditional installed client applications talking to a backend database either directly or via a wrapper (odbc/jdbc).
I am attempting to build a client/server game architecture and would like to begin testing the game using my local Mac as the server. I have found several articles on Bonjour, but that is for local network traffic only. My goal is to make this application work over the Internet, connecting to a hosted application on a static address to facilitate turn data. However, I'm at a loss as to which Cocoa APIs to use for this purpose. Should I use NSConnection, NSStream subclasses, or good 'ol C sockets and whatnot. The game state is already built in Objective-C and is ready to be set in motion once I have the server facilities ready. Any insight?
NSConnection, NSStrean and C sockets are build for different needs. You need to specify the needs of your game and the kind of service in order to get more help. If you want to develop a client-server application that relies on the Internet and not on the local network, Bonjour will not be able to help.
C sockets, and Cocoa APIs that wrap around them are intended to operate on an open network stream between the client and the server. The advantage of having an open stream is that you can have the server send data to the client without the client having requested for it. NSURLConnection in Cocoa works differently. With it, you can perform HTTP requests and receive formatted responses from a server.
If your application is based on HTTP requests, I recommend you take a look an NSURLConnection, or AFNetworking, as a 3rd party alternative. If your application relies on open streams, I recommend you take a look at CFNetwork from Apple (C wrapper around BSD sockets that originates from the days Macs had Carbon, with great performance, stability and versatility) and GCDAsyncSocket, a 3rd party library wrapped around BSD Sockets, supports Crand Central Dispatch, is Objective-C ready, and does the job wonderfully.
I hope I helped.
I suggest you to use sockets, since they're not hard to use and are a standard way. I've even written an asynchronous wrapper class around BSD sockets: https://github.com/H2CO3/TCPHelper
This is for simple, one-to-one TCP protocol connections, supporting both IPv4 and IPv6. You can send and receive raw NSData and possibly build a protocol around it.
Foundation classes such as NSURLConnection are not particularily for this purpose; rather than to interact with standard HTTP servers (I suppose you don't want to implement a full HTTP server for a game).
NSNetServices may suit you just like CFNetwork, but the latter is a bit harder to use. If you'd like to use Foundation classes, I'd recommend NSNetServices.
Hope it helps.
There are many different ways to accomplish this. It really depends on how you'll be passing the data and what it will be used for.
First, I would setup a hostname that you can use for development purposes with your game. You can use anything like http://dyn.com/dns/ to setup this for your Mac. Then you can enable a compiler setting to switch out the development / production URL's.
Next, I would recommend using TCP sockets for your game (using CocoaAsyncSocket - https://github.com/robbiehanson/CocoaAsyncSocket). This method should work fine your your use case. Since you are doing turn-based data (and since all of that data is vital) I would not recommend using UDP sockets (but those would work if you were solely passing position, video, or audio data where a dropped packet might not matter).