I was looking for live streaming platforms where you can broadcast yourself and get some money for it. Checked out Streamup, YouNow and AmberDog. The last one was the easiest to use in my perspective but when the stream is loaded, it asks the following:
Peer assisted networking
amber.dog may use peer assisted networking. Do you want to allow
access to your upload bandwidth?
What the hell is that? I haven't seen such a message in any other live-stream platform. Should I allow or deny it?
They are using RTMFP protocol for live-streaming. No harm.
The stream is sent from one peer (source) to another (origin) directly and the server only organizes the graph - who sends what to whom. In this case, by agreeing, you only allow the stream coming to your flash player to be sent over to the next watcher. This is basically the future of live-streaming.
Related
I need to make TCP based decentralised chat app for local network. By decentralised I mean there is no central server. Each entity on a network should have server/client architecture. When app starts it should check which user is online ( already running the app ). My question is how can i check that? Can i do it by trying to connect via connect() function from socket library? I'm new to programming, especially socket programing, so if it's a dumb question sorry in advance.
You should definitely study how other decentralized applications do this. There are lots of techniques.
Each instance of the application should, as part of its server functionality, track the addresses of other instances of the application. Each instance should, as part of its client functionality, keep track of a few instances it can connect to. Prefer instances that have been around for a long time.
The software should include a list of servers that have been running for a long time and are expected to typically be available. You may wish to include a fallback method such DNS, maintained by anyone willing to keep a list of well-known servers offering access through a well-known port. The fallback method can also be IRC or HTTP.
If you want to stay decentralized, you might want to try multicasting or broadcasting a request packet to all hosts on the network to discover other instances of your chat application.
Something similar has been implemented in Pidgin, named Bonjour. It works quite nicely and provides chatting capabilities on a local network. More specifically, it is defined as a Serverless Messaging part of XMPP.
If you are looking for code examples, have a look at one of my projects where I use multicast to discover hosts on the local network that provide a specific service: Headers and implementation.
We're writing a streaming service using wowza as our streaming server with a flash applet on the client. A recent concern we've had is a single client accessing the same stream from multiple tabs. In this scenario, we want to save bandwidth by sending the client only one copy of the stream.
How is this problem typically handled?
I think this is not possible for multiple technical reasons. This is my personal opinion, so experts can correct me if I am wrong)
The flash players on multiple tabs are not aware of each other and they can't be associated with each other on Wowza Streaming Engine side either. Wowza would see 2 different clients from the same IP, but that might be two different devices behind the same router.
The browser tabs are usually isolated from each other in the browser implementations for security and stability reasons, not allowing cross-browser data sharing
The 2 players are not in perfect sync, started at different points in time and lags, buffering might also introduce time shift
I haven't seen any attempt to reduce bandwidth in this scenario. There are solutions to prevent users from playing multiple streams in parallel though, which I would not like to link here, but do a search for "wowza forbidding concurrent stream" and you'll find it.
I think what you thought of is multicast streaming, where network packets are broadcast over the network and every player receives the same stream. This is used in IP TV systems, but that is not possible over internet, only on intranet.
I want to setup a personal videoconferencing service for my family, friends and myself. The main problem I have with current options is that they are either closed-source and centralized (GG hangouts, skype) or open-source but not working in corporate environment or in hotels (due to strict firewalling rules and the "Skype is going through, if you want VOIP use that" kind of netadmin reaction).
I have two solutions then. Either setup a STUN/TURN relay server and use XMPP and SIP as I used to, but that would require my friends to setup that too. Or setup a whole VOIP server. 2 solutions come to mind: SIP and XMPP. Though to my knowledge, each of them ultimately uses the (S)RTP/RTCP protocol.
And that's the problem. Out of the specific signaling part used by the two of them, I really can't figure out the difference between them, their typical use case.
I think you're right in that as far as setting up a video conferencing system XMPP and SIP are equivalent. They both are signalling only protocols and the media sessions they set up typically use RTP (although they can both be used to set up any kind of session you want but RTP is the norm).
The biggest problem is also going to be the one you mention about getting video streams out of a corporate firewall. Skype overcomes this obstacle by sending it's media over an SSL connection and is thus able to get through firewalls. Theoretically you could do the same with RTP and in the past I once used openvpn connections with a SIP client to test some audio calls. My experience wasn't great as the audio was very choppy, assumedly as a result of all the extra packaging that is required to get the high volume of small audio packets from one end to the other. That was nearly a decade ago though so perhaps with the better CPU and bandwidth resources available now it would work better.
Personally I think I'd stick with Skype as it's going to be a big hassle to set up your own system. If you were to go ahead with your own the first option I would try would be Asterisk combined with openvpn so that if the clients were behind a firewall or had NAT issues they could connect over it.
At the bottom, it's all about socket communications. If there is some way to get the ip of the both users, why can't the connection be directly setup between the users instead of having to go thru a server in the middle?
My 2 cents:
No one out there forces us to have a server based real-time communication model. Infact XMPP have an extension called "Serverless Messaging" which defines how to communicate over local or wide-area networks using the principles of zero-configuration networking for endpoint discovery and the syntax of XML streams and XMPP messaging for real-time communication. This method uses DNS-based Service Discovery and Multicast DNS to discover entities that support the protocol, including their IP addresses and preferred ports.
P2P chat applications have been for over a decade now. Having a server in the middle is purely a decision dependent upon your application needs. If your application can live with chats getting lost while the user was transitioning between online/offline status, then you can very well have a direct P2P model going. Similarly, there are a loads and loads of advantages (contact list management, avatars, entity discovery, presence authorization, offline messages, ....) when it comes to choosing a server based messaging model. If you try to have all this right inside your P2P based clients, they might die or under-perform because of all the work they will need to perform by themselves.
"WebSockets" were not designed for P2P/Serverless communication, rather they were designed to provide a standardized PUSH semantic over stateless HTTP protocol. In short, "WebSockets" is a standardized way replacing hacky comet, long-polling, chunked-encoding, jsonp, iframe-based and various other technique developers have been using to simulate server push over HTTP.
Named WebSockets (if someday it is fully and widely supported) could be the solution.
http://namedwebsockets.github.io/spec/
Named WebSockets are useful in a variety of collaborative local device
and local network scenarios: Discover matching peer services on the
local device and/or the local network.
Direct communication between users is possible in Peer To Peer (P2P) networks. In P2P each participant can act as client as well as server. But for P2P networks you need to write a separate program to make the communication possible.
Web Sockets let you leverage existing common browsers as clients. All depends on what is the purpose of your application and how you want to deploy it.
If there is some way to get the ip of the both users
You nailed the answer right in your question.
Most machines I use have IP address of 192.168.0.10 (or similar from 192.168. private network) and are deep, deep behind several layers of NAT. With the end of free IPv4 address pool and IPv6 nowhere near sight, this is the reality most users live. Having a stable intermediary of known, routable address helps a ton working around this issue.
WebSockets don't allow the socket to listen for connections, only to connect as a client to a server (not reverse). Technically they could make it allow this, but as far as I understand the spec doesn't currently (nor is it expected to) allow listen functionality for WebSockets.
The new WebRTC (http://www.webrtc.org/) spec looks like it might support peer-to-peer connections. I have not played with WebRTC at all so I'm not in a position to comment on it. I think it would be a bit more involved than WebSocket stuff. Maybe someone who knows WebRTC better can chime in. (Also apart from the latest version of Chrome I'm not sure if any of the other browsers really support WebRTC yet).
I am developing an iPhone application that allows to basically click through a series of actions. These series are predefined and synced with a common configuration server.
That app might be running on multiple devices at the same time. All devices are assumed to have the same series of actions defined on them. All devices are considered equal, there is not a server and multiple clients or something like that.
(Only) one of these devices is used by a person at any given time, it is however possible that the person switches to a different device at any given time. All "passive" devices need to be synchronized with the active one, so that they display the same action.
The whole thing should happen as automatically as possible. No selection of devices, configuration, all devices in the same network take part in the same series of actions.
One additional requirement is that a device could join during a presentation (a series of actions) and needs to jump to the currently active action.
Right now, I see two options to implement the networking/communication part of that:
Bonjour. I have implemented a working prototype that can automatically connect with one (1) other device in the network and communicate with that. I am not sure at this point how much additional work the "multiple devices" requirement is. Would I have to open a set of connections for every device and manually send the sync events to all of them? Is there a better way or does bonjour provide anything to help me with that? What does Bonjour provide given that I want to communicate with every device in the network anyway?
Multicast with AsyncUdpSocket. Simply define a port and send multicast sync events out to that port. I guess the main issue compared to using bonjour with tcp would be that the connection is not safe and packets could be lost. This is however in a private, protected wlan network with low traffic if that would really be an issue. Are there other disadvantages that I'm not seeing? Because that sounds like a relatively easy option at this point...
Which one would you suggest? Or is there another, better alternative that I'm not thinking of?
You should check out GameKit (built in to iOS)--they have a lot of the machinery you need in a convenient package. You can easily discover peers on the network and easily send data back for forth between clients (broadcast or peer to peer)
In my experience Bonjour is perfect for what you want. There's an excellent tutorial with associated source code: Chatty that can be easily modified to suit your purposes.
I hobbled together a distributed message bus for the iphone (no centralized server) that would work great for this. It should be noted that the UI guy made a mess of the code, so thar' be dragons there: https://code.google.com/p/iphonebusmiddleware/
The basic idea is to use bonjour to form a network with leader election. The leader becomes the hub through which all the slaves subscribe to topics of interest. Then any message sent to a given topic is delivered to every node subscribed to said topic. A master disconnection simple means restarting the leader election process.