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.
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I am trying out the WebRTC examples of MuazKhan. It is working perfectly fine when the broadcaster is on AWS/Azure(or any other network) and the receiver is through my phones 4G network. But as soon as I switch to my broadband from 4G, the video stream is unable to connect and I the video player keeps on trying. Therefore I assumed that the problem is of the router NATting and will be resolved if I use a tried-n-tested TURN service such as Xirsys.
Sadly, even after using their TURN servers, I still am blocked with my broadband connection issue as mentioned above. Here are a few queries that I wanted to discuss:-
This issue seems to be due to NATting through my broadbands routers. Shouldn't using the TURN server solve it?
How can I verify if even the TURN servers are getting used and its not just the STUN servers.
Can this issue be due to the Signalling Server?
Do i need to enable some specific protocols in my router to make this work?
What else could be responsible for the issue that I should debug?
My case is the following: I will install some prototypes connected together with wifi (one device is the AP) in a distant site. I'd like to be able to remotely connect to my network for avoiding to travel.
As there will be 4G networks available there, I was wondering if a simple solution could be to have a smartphone connected to my wifi network, and use it to forward ssh client requests through 4G connection.
I have no idea if it's possible, how/depending on network operators, if there are standards and software (preferably android) for doing so.
Sorry it this question remains a bit vague, but any pointers would be appreciated.
There are also some 4G boxs that provide internet as wifi AP. I could change my network to connect to it, but I have no idea if I can reliabily get an address to this.
Thanks for your help.
A VPN server might be an option. Like DD-WRT on a router, or your own dedicated VPN server.
Is there a simple guide from where I can start creating a stun / turn and signaling server ?
I spend over a week searching for those things and couldn't find any guide where I can say:
okay, I am on the right track now - this is clear.
So far, everything is so abstract without any examples.
This is what I'm trying to achieve: a simple video stream on my local network where I'll have a server with installed usb camera on it, and an application on my iis which will connect to the usb camera and stream it to the clients, and every time when a client opens the application, will see the video stream from the server camera.
Note: since I want to use it on my local network do I really need a stun/turn server, or is there a guide that shows how to avoid it ?
Media streamed over dedicated servers HTTP/HTTPS rarely needs a NAT traversal solution. Instead, just have your web server with camera attached, on the public Internet or behind your NAT with port-forwarding enabled.
There are LOTS of streaming media solutions available as open source, free downloads, or commercially sold. A good list is here:
https://en.wikipedia.org/wiki/List_of_streaming_media_systems
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/
I have a piece of software I have written that talks to a web service over it's Wifi connection on a Casio WinCE handheld.
Unfortunately the Wifi network and the network created by WMDC (or ActiveSync) to host the debugger connection clash, they are both 192.168.55.0/24. So I can't have them both connected at the same time.
Does anybody know if you can reconfigure WMDC to use a different address range?
I've hunted through the registry but can't see anything obvious, and Google is not turning up anything useful.
Thanks,
James.
Is the development PC on teh same network as the WiFi connection? If so, I'd abandon ActiveSync/WMDC altogetehr and just use ethernet debugging over the same connection that it's using for the web service calls.