I want to make an app for smartphones that will send and receive UDP packets to and from a publicly accessible server (so not peer-to-peer, and no need for holepunching).
There's bunches of questions about this but they are all old and vague. I want to know if thing changed since.
Does anyone know a list of mobile networks that now allow UDP? And in a more general way, does most network allow UDP (in US, EU and Asia)?
Related
I was looking all over the internet to find a way to capture incoming packets from a certain network interface, then I came across PCAP, TCPDUMP, I believe the most commonly used networking library out there is Boost/Asio, so I wanted to use this library in order to capture traffic, but apparently there is no example for using Raw sockets or other classes to listen for incoming packets to a certain NI, I would appreciate any help or examples on this.
We eventually found out the best option for sniffing incoming packets was Libtins.
libtins is a high-level, multiplatform C++ network packet sniffing and crafting library.
Its main purpose is to provide the C++ developer an easy, efficient,
platform and endianness-independent way to create tools which need to
send, receive and manipulate network packets.
It uses a BSD-2 license and it's hosted at github.
After webrtc passes by a signaling server on Internet, how it works when two machines are running in the same network?
The data will be exchanged only on the network or will it still use internet ?
I am asking this because of our internet is not good, it's too slow. but our local network speed is very fast.. So I would like to know if the internet signal will affect the audio and video conversation.
Thanks a lot!
Depending on the network configuration, the devices should connect directly over the local network. Please note that some browsers, such as Safari, may not share with the signalling service local ICE Candidates unless configured to do so (false concern over sharing network info). The devices must share local ICE Candidates, or else they will still stream via the external network or a TURN server, if available.
I was going through this PubNub WebRTC demo. https://kevingleason.me/SimpleRTC/minivid.html
Which works fine within same network (same browser or different devices across same network). But I tried using it over internet, I am able to connect a call but can not see anything but a black screen. This is the source for same tutorial
https://github.com/pubnub/SimpleRTC
I have gone through many such application, such as AndroidRTC
and I face same problem (black screen after connection over internet). I am unable to figure out why, any help is appreciated.
You need some sort of signaling mechanism (PubNub, Firebase, or your own software [nodejs seems the preferred choice these days]) to get the webRTC API communicating P2P on your local network. To get webRTC to work from one network to another you need a STUN server/service. Google provides free stun servers (stun:stun.l.google.com:19302). To get webRTC to traverse strict firewall settings and complicated networks you need a TURN server/service like xirsys.com.
This article covers it all ...
http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/
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.
For the past two week I have been unsuccessfully trying to implement udp hole punching, but I'm not sure why. I understand that the algorithm for hole punching is not guaranteed to work, but I believe it should work in my test case because I have noticed that once I bind my socket on my home-network, the port is the same to the outside world as it is locally, and stays that way for all connections made from this socket. Any help after reviewing my trials would be appreciated.
I have three computers, my osx desktop, my iPhone, and my amazon ec2 ami.
on the desktop I've built a cocoa app which uses the GCDAsyncUDPSocket library to bind a port and contact the ec2 server, where a java app using apache's mina library stores the sockets external ip/port and associates it with a username passed in the payload.
the iphone, which is on the AT&T network runs an app which uses the same GCDAsyncUDPSocket library to contact the ec2 server with the same username, which then the ec2 does a lookup for the username, finds the desktops info and informs the desktop of the iphones address and the iphone of the desktops address.
now the iphone & the desktop know about each other they start shooting off packets at each other in hopes to get a punched hole.
in theory this should work, but maybe I am missing something about mobile networks that would make this difficult? But then again running a simple udp echoer on a 4th external computer to manually msg the desktop did not work either, so maybe its my router, but I don't see how that could be as all my tests show that the port the desktop asks for is the same one assigned by the router.
I've been at this for nearly two weeks with little progress and any tips would be appreciated!
"once I bind my socket on my home-network, the port is the same to the outside world as it is locally"
I highly doubt that. To traverse NAT given peers A and B which have sent datagrams to a 3rd party: S you need to send datagrams from A to B and vice versa using their public IPs as seen by S and their port as seen by S (i.e. not the port A, B are bound to from their point of view).