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
Related
I'm in the process of porting an application to ChromeOS with the requirement that it should look and feel as native as possible. This means in particular that it should allow things such as multi-monitor support and USB support.
One possibility would be to implement it as a web application (since we have already a web client), but in this case I would need to add support for native features (again, multi-monitor support and USB device access), so I wonder what needs to be done in this case. My wild guess as a ChromeOS developer newbie is that I would need to extend the code with ChromeOS JavaScript features, and I don't know if this is possible.
Another possible approach would be to write an Android application, since I see that ChromeOS added support for Android applications (in this case I would have to write the code from scratch).
Finally, another option would be to write native code, which could be possible for example relying on a Crouton development environment, and reuse the code of a native C application.
What approach would you recommend to build a ChromeOS application starting from a web application or from a native one?
What approach would guarantee access to native features (multi-monitor support and USB)?
I discovered two APIs which should help with multi-monitor support:
https://developer.chrome.com/extensions/windows
https://developer.chrome.com/apps/system_display
The system.display API allows one to discover and monitor the current monitor layout, while the windows API allows one to create several windows in the same application. By combining these ones I should be able to create a window for each monitor in case I go with the chrome implementation route.
Given that I already have a native implementation for Linux, Crostini (as opposed to Crouton) is also very appealing since it provides a deeper level of integration with virtually no changes to the code and no need to maintain two different versions of the web client, with the only downside that it requires the user to create a Linux environment and manually install the application, also it is not supported on all chromebook devices and on many it will be never supported.
I still need to check what is the performance overhead. Also the level of integration with USB I/O might be higher than the one achievable by using the chrome API.
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.
Recently I have worked using WebRTC and I'm wondering if it would make more sense to implement a Real Time Communication open standard at a native level.
Let's say that instead of a web browser API we have a native API that any native app, including the browser can leverage.
Part of the promise of WebRTC is to have RTC on the browser without plugins but why stop there, why not have RTC on any device with media capabilities without plugins. There are many devices with media capabilities that will not run a web browser, e.g., wearables. It seems to me that the browser itself has become the plugin and I think we need to get rid of it as far as RTC is concerned.
It sounds like OpenWebRTC is going in a similar direction but so far they are only working inside the browser.
Are there open standards for native RTC? So far it looks like RTCWeb is only concerned about the browser.
Are there any projects/initiatives for native implementations of an open standard for RTC?
webrtc definition. webrtc is stuck into two parts, complementary but separated. the W3C consortium is standardizing a JS API for browsers named webRTC. The IETF is standardizing the underlying protocols and what happen on the wire for interoperability, it is named rtcweb.
the IETF's rtcweb group defines everything you need to interoperate with a browser, without being a browser yourself, i.e. for gateways, devices, .... It has been made explicit at the latest meeting in hawaii last november, and there is for example a corresponding draft.
On the client side, the implementation of webRTC JS API is done on top of c/C++ implementations. Those "native" (as in non-browser, C/C++) APIs can be use directly for servers, embeddable devices, gateway, ect, or can also be wrapped in different languages (obj-c, java) to provide "native" (as in mobile native) APIs.
Note that BOTH openWebrtc.io and webrtc.org have a full implementation of webRTC in C/C++ that you can use. openWebrtc provides iOS wrappers, and webkit wrappers (for safari), but do not provide data channel support, ORTC API support, nor compile under windows. webrtc.org supports all desktop OSes, and provide wrappers for both iOS and Android. The build tools are specific to google's chrome though, unlike openWebRTC which uses standard auto tools, github, ...
HTH
Currently there is no effort in this direction. The guys in the webrtc standardization committee have their hands full standardizing just the javascript API. As you know the current spec is not final and is currently still worked on. And now ORTC will generate even more work.
There are many reasons why no one is currently trying to standardize any form of native RTC. Here are some that come to my mind:
What exactly is native? Javascript is native for the browsers. The chrome version of webrtc is in C++ but the OpnWebRTC one is in C. Android developers use mostedly Java, iOS developers use ObjectiveC. Should there be standards for all these languages? That's going to take forever.
As I said standardization committee already have their hands full.
There is still quite a lot of experimentation that goes on with WebRTC. Standardization may prevent this.
The API of the native libraries tend to be very similar to the JS API.
I am looking for a tool capable of generating multiple Xmpp connections to load-test a XMPP server with a secure connection, especially starttls.
For a xmpp plain text authentication I had used jab_simul(followed this tutorial) and tsung both with success.
But I was unable to use the tolls above for the starttls,I peeked into the code of both tools and tried different configurations of the tools.
Another option I am pondering is using a xmpp library like eXmpp and make a specific load-testing tool myself with, instead of altering jab_simul (C software with comments in language i do not understand) or altering tsung(all purpose load-testing tool, so lots of place where you can go wrong).
short-story - I am looking for a tool or advice to stress-testing/load-testing a xmpp server.
We are facing exactly the same challenge right now. After deep consideration we found out that only especially build software can deliver the load we want to test. (Remember, you can configure ejabberd to something very specific :-)
For that we developed a small library called xmpp_talker https://github.com/burinov/xmpp_talker (Apache Licence) which is a kind of xmpp client made as a gen_server. I find it is a very nice starting point to build any kind of load simulation software. There is also echo_worker example included. So, you have good base to start. At the moment xmpp_talker is suited for exmpp 0.9.7. As far as I know in a few days will be out version 1.0.0. (or 0.9.9?) There are many bug fixes (trust me you don't want to know about them). On monday I will release xmpp_talker for exmpp 0.9.8 with proper service interruption handling.
In case you deside to go the same way xmpp_talker could be useful for you.
Added: Here is also great article that is realted to the topic: https://support.process-one.net/doc/display/EXMPP/Scalable+XMPP+bots+with+erlang+and+exmpp
There's also the recently started XMPP benchmarking project called xmppench which aims to be a high-performance benchmarking tool simulating some reasonable use cases of XMPP servers. It's written in C++, based on Swiften and boost.
I am working with an electronics appliance manufacturer to embed LAN based control systems into the products. The idea is to serve up a system configuration/control interface through a web browser so clients never need to install software. We can communicate with the appliance by sending and receiving serial data through the embedded module. Since the appliance can also be controlled from a front panel UI, it creates a challenge to keep a remote web interface in sync with very low latency. It seems like websockets or some sort of Push is what we need for handling real time events from the server to clients.
I am using a Lantronix Mathport AR embedded device server. Out of the box the unit will serve up any custom HTML and java servlets/applets. We have the option to install a lightweight Linux distro if we need more flexibility. I am not sure how to implement any server side apps since the device is not running standard Apache. I believe it is using Boa.
Can anyone guide me in the right direction of how to do this?
Some general info...The WebSocket protocol (draft spec here) is a simple layer on top of TCP. What this means is that, if you already have a TCP server for your platform, implementing the WebSocket is just a matter of hours. The protocol specifies a handshake and two ways of sending data frames.
I strongly suggest you start by reading the 39 pages spec.
As Tihauan already mentioned, start by reading the spec, and also note that there are still some changes ongoing, although websockets is now more stable than it was 1 year ago.
Key point for me was the requirement that websocket data is entirely UTF-8 text, which lends itself nicely to JSON based message definitions.
Our system uses a form of embedded linux, so we then added and made use of the following libraries:
"libwebsockets" from:
http://git.warmcat.com/cgi-bin/cgit/libwebsockets/
"jansson" from:
http://www.digip.org/jansson/
Using the above as support libraries, we created an internal lightweight "client/server" that allowed our other software modules to register for certain, applicable, websocket messages, and respond as needed. Worked great.
Good luck and best regards,
I'm a bit late, but Mozilla posted a guide entitled "Writing WebSocket servers", which literally guides you through writing a websocket server.
You will need to already know how HTTP works and have medium programming experience. Depending on language support, knowledge of TCP sockets may be required. The scope of this guide is to present the minimum knowledge you need to write a WebSocket server.
https://developer.mozilla.org/en-US/docs/Web/API/WebSockets_API/Writing_WebSocket_servers