does anyone know if its possible to host the receiver app for chromecast in a remote host, such as heroku? Is it only possible to find my chromecast device if its in the same wi-fi network?
You've got two separate questions here.
1) Yes, the receiver app can be hosted anywhere, and the vast majority of them are on a remote host rather than served locally. When you register your app with the Chromecast SDK console, you will have to submit, with the appID, the URL where the receiver app is located. Keep in mind that if your app is 'published' (available to all rather than just to those Chromecasts you whitelist), then your host much be https:// enabled, whether it's local or remote.
2) I think you're asking whether or not a sender app can find a device that's not on the same wi-fi network? Generally, no ... but at Google I/O this summer they did discuss a feature that would allow people not on the network but close enough to the Chromecast (i.e. in the same room) to communicate.
If I'm misunderstanding question two, please clarify.
Related
So I've seen projects like trojan-go,v2ray-go
They are making their own proxy protocols in user space level. I am trying to do the same thing but I don't know how to redirect network traffic from clients to proxy server.
Basically I don't understand how those tools (or any tool) can redirect internet traffic of the device to a certain server, so when the packets are going to the internet they go to the proxy server first instead of their destination ip address.
How can I do it in golang without dealing with netlink and Iptables?
I know apps like wireguard do this by dealing with layer 3-4 stuff using netlink API but I need to know how apps do it without adding a new network interface.
Anybody here with some experience in websockets and webRTC using TURN/STUN servers?
Requirement:
Send real-time video feed from local IP to browser in an external network and I need some help implementing via raspberry pi 3b+. My camera source is android device, and using 3rd party apps I am able to generate the video feed over local network. Using the same app I can stream via Youtube Live,but getting a latency of about 2 secs in ultra low latency mode and dvr enabled. And I am trying to reduce the latency of the stream.
Q1. Do the semi-public TURN server provide a one to one peer. Or anyone can just jump into the URL and view and override what I am streaming?Please provide few list of service providers.
Just for information there would be 1-2 users browser connected at max.
Q2. Do I need Janus gateway to send webRTC/websockets data into the TURN/STUN server? Since my raspberry is connected to a different network and I cannot port forward due to carrier constraints.
Q3. Do I need both STUN/TURN servers or do I even need webRTC instead of websockets to send my video stream over the internet. Is websockets not sufficient?
Q4. Since we are not implementing over local network do we need to install coTURN too on raspberry pi?
Q5. Is there any android app that can publish the data from camera to websocket/werRTC server with a public ws URL?
Any help would be really helpful.
Q1. TURN servers relay media. They do this by allocating for every connecting peer a relay port between 49152–65535. This relay port will then be used to transmit the media to the second peer. The peers will know which relay ports to use automatically since this is part of the ice gathering process. To get back to your question: Other Peers cannot write to that relay port, it is 1 to 1 with handshakes, there is no chance of someone else overwriting it.
Q2. You definitely do not need a Janus Gateway to use TURN. TURN and STUN will probably work fine for NAT-Traversal without port forwarding.
Q3. You need at least a TURN server (but you ideally want to use 1 STUN server and 1 TURN server). STUN will work in most cases, but will fail if there are firewalls or complicated NATs, which block inbound udp connections. TURN is just the fallback for those cases.
Needing WebRTC? For just streaming videos, it depends on the use case. A sequence of images can be transmitted over websockets, they can handle Blobs fine. But you won't have a very fluent, high fps AND high resolution video stream this way. And of course, I know of no usable way to transmit audio over websocket.
Q4. The raspberry pi is a Peer that transmits media? Peers do not need a local TURN server installation, you will only need 1 TURN server (which should not be behind a NAT, probably running on some web server). The TURN server is a separate instance.
EDIT
For your private testing and development purposes, you may use https://numb.viagenie.ca/ . I don't know much about commercial turn server hosters, except that some exist. For someone who owns a v-server or root server, installing coTURN may be an option, this Tutorial might be helpful. To check if the server is working, I also found this snippet to be very useful.
END EDIT
Q5. There is no android app that publishes webRTC streams to a ws URL since websocket
messages are used by webrtc only for signalling (that is, telling peers their host candidates, those are the IP adresses and ports learned by the ice gathering process, this includes the TURN and STUN ip and port combinations).
I am currently writing an app that is planned to control a machine. The machine is controlled by a Raspberry Pi, which offers an API (via flask) to the local wifi. The app on the other hand is also connected to the same wifi and accesses the API.
To make sure that not everybody who downloads the app and is connected to the wifi, can control the machine, I setup some basic authentication.
My next step was actually to switch to https with a self-signed certificate. But the machine(/the raspberry pi) and the app need to be in the same wifi to communicate. So there are actually no intermediaries in the communication. This again makes me wonder if there is any possibility of a man-in-the-middle-attack and if I really need https communication.
So my question is: do I need https here?
A subjective answer. First you have to decide what is the risk to your machine if someone/thing gets control of it. For most consumer applications, within the household maybe that risk is low (maybe not - what about an irrigation controller or heater?). Then why and with what probability would someone WANT to hack in (maybe if your machine is a best seller across the globe it might be a fun target).
You might be surprised at how many devices are on a normal households wifi - dozens at least. Furthermore - while most consumer devices don't rely on inbound access (most use a website to bounce control/commands through) there are probably a lot more inbound (from the internet) ports that have been opened through firewalls than you imagine.
So - I do think there are many opportunities for MITM in a normal household wifi. Whether that would be a concern in early product development - that's up to you.
This SO answer: Is it possible to prevent man-in-the-middle attack when using self-signed certificates? might be useful when actually implementing.
I would know if there is a way to determine if the application connected using 3G or WIFI network.
You're not going to be able to determine this on the server side only. The only thing you can check is the HTTP user agent, which will help you figure out the device. However a connection is just a connection, there's no way to determine how it is connected without the device telling you. If you are developing a local mobile application then you can either do different logic on the device or send additional information about the device's connection.
I want to implement a P2P photo sharing application.Scenario is like this:
A is online and he would like to share his photos with B. Through some server, B gets A's IP address and access A's photos directly.
Is it possible to implement using WebRTC or Websocket ? Please give me some inputs,
Thanks
I implemented P2P file transfer on websockets with very small nodejs server. But it works fine only on Chrome thanks to "download" tag. I also have to have middleware server so it's not STRICTLY P2P. My node.js server never has full file it just transfer current file chunk from one websocket connection to another.
I hope WebRTC will help you to implement what you wish more smouthly, without any middleware.