My API requires reliable and fast internet connectivity (for scraping).
I tested it on my internet connection, the time it took to fetch & process information rounds up to 20 seconds on slow connectivity, 2 seconds on fast connectivity.
Is there any platform or service (preferably free) that best provides reliable and fast internet connectivity, if I wish to deploy?
Just to be honest: The more you pay the more you get.
But it doesn't help when the client (mobile) has slow connectivity.
My opinion: use Azure to host your app and use the nearest region(s).
Related
I'm looking for a tool or set of tools or framework, preferably free, for monitoring different parts of my web app such as the api part, the db connection, the connection to a third-party api which it uses. It doesn't have to be sophisticated, it rather should display "up" or "down" and if it's down then there should be a means of notifying me by email or Slack or the the like. Your suggestions?
I encourage you to give a try to the Community Edition of Pandora FMS. The Network server is focused on network monitoring so it can perform TCP checks to port 80 and even send information and check the output back. See this example of the wiki: http://wiki.pandorafms.com/index.php?title=Pandora:Documentation_en:Remote_Monitoring#TCP_Monitoring
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.
We have developed a 1-1 video conferencing solution using RTMP , RTMFP technologies using Adobe. We are making use of Flash Media Development Server for recording the web streams. We have tested our application locally and it seems to work good.
Thus, We deployed the application to our server but the latency is very high (like 15 seconds). I mean when i talk , the other party gets my voice after 15 - 20 seconds.
Could someone explain me what is the real problem behind this ?
Do i have to purchase this https://secure1.influxis.com/webrootv10/custom_enterprise/index.aspx
Application.xml
http://www.2shared.com/document/PfMTpQsJ/Application.html
Thanks.
you need tool to control the time spent 3G traffic, because I have it limited. Simple System Monitor is not very suitable because after the reconnection of data traffic reset spent ...
Found a program NTM - Network Traffic Monitor - this is exactly what you need for a limited internet! You can specify a limit and when it ends it will automatically shut-up! Cool thing!
http://www.webupd8.org/2010/12/do-you-have-limited-internet-plan-use.html
Simplified, I have an application where data is intended to flow over the internet between two servers. Ideally, I'd like to test at what point the software ceases to function. At what lowerbound limit (bandwidth, latency, dropped packets) do things stop working to test the reliability of the software.
What I thought I would do was the following:
Setup up 3 machines (VMware instances)
Install the 2 applications on two of the servers.
Setup up the 3rd server to sit between the two machines by doing some sort of magic with Routing and Remote Access on Windows 2003
Install either Traffic Shaper XP or NetLimiter to limit the bandwidth
Run something like TMnetSim Network Simulator to simulate a bad connection.
Does this sound like a good idea or are there easier/better ways of doing this? I'm not that comfortable on Linux and my team mates are even less so.
WANem does exactly this. We have used it both in a virtual machine on the desktop and on a dedicated old pc and it worked great. It can simulate all sorts of broken connectivity.
FreeBSDs ipfw has provisions to simulate links with a given bandwith, latency or error rate. You could use that FreeBSD machine as your machine "in the middle" in your above setup.
You probably can also run at least one of the endpoints on the same machine if you want to reduce the amount of servers involved.
Someone actually packaged up the settings and whatnot necessary for the FreeBSD solution to this problem and they call it DUMMYNET.
It simulates/enforces queue and bandwidth limitations, delays, packet losses, and multipath effects. It also implements a variant of Weighted Fair Queueing called WF2Q+. It can be used on user's workstations, or on FreeBSD machines acting as routers or bridges.
It can simulate exactly what you want, and its free and will boot onto commodity hardware. They even have a canned install of it that is small enough to put on a floppy disk (!) that you can download at that link.
Maybe it is time to learn a bit about Linux because adding a 50ms delay on every outgoing packet can be done in typing just one line:
tc qdisc add dev eth0 root netem delay 50ms
For more see the Linux Traffic Control HOWTO
We had a similar requirement some ten years ago - I'll see if I can recall how we managed it.
If I remember, we wrote a socket proxy program which was controlled by inetd on a UNIX box. This socket would accept connections from a client and open equivalent sessions through to the server. It would then loop, passing messages in both directions.
The way we achieved WAN characteristics was to introduce random delays (with upper and lower limits) in both the connection establishment and the passing of data once the link was up.
It also had the feature to drop the link occasionally as WAN links were less reliable for us than local traffic.
I recall we had to make it threaded to stop the delays from affecting reverse traffic on the link.
There is a very good (and free) Microsoft solution for that, we use it for quite some time and it works great, it can very easily simulate every thing(packet loss, low bandwidth, disconnection, latency....)
This is the best solution i found for a windows environment
More information and a download link can be found here: MARCO blog post
this product has gone some evolution and it is now integrated into visual studio as part of the automation testing, but i found the use of the standalone(that is quite hard to find, so keep a local copy) to work much better. keep in mind that you need at least two computers(or VMs) since you need to pass through a network adapter in order for the application to work its magic.