I would like to create a web based p2p application between two nodes. A website shows a list of nodes. When a user (say A) clicks on a node, it must setup a p2p chat like connection between the two. It goes like this : once connection is established, node A can send a query message to B. Once B receives the query message, B must respond with the correct answer, (if A queries : RETRIEVE x.txt, B's response must be the contents of x.txt) I would like to be pointed into the right direction regarding the proper tech / protocols to be used. Thank you😀
Firstly, if you want to reach all the nodes, you need to collect their information so that when you click, you can connect to it.
Secondly, if you want to connect to these nodes, you need to do NAT traversal so that they can connect to each other.
Thirdly, you may want a reliable connection, so you need reliable UDP.
According these, you need the following protocols:
Creating a center controller like tracker to collect infomations
Creating NAT traversal, like nat-pmp and upnp, and it's better if you can build a ICE in the center controller
When click to some node, using udp to connect to it
If you want the connection reliable, you may also need to do reliable UDP, like QUIC, kcp or libutp.
Related
I am trying to make 3 peer video chat using Webrtc in Flutter. A needs to see B and C. B needs to see A and C. C needs to A and B. I also use socket.io for signalling server. I used 2 peerconnection. two peer connection is successful. When I tried to connect the third one and tried to pass offer I got an error.
to execute 'createAnswer' on 'RTCPeerConnection': PeerConnection cannot create an answer in a state other than have-remote-offer or have-local-pranswer.
is my approach is bad for multiple peer connection?
A mesh of 3 users means each user sets up two connections, one to each of the other two users. At each client's end, these are two entirely different RTCPeerConnections, and you can't reuse candidates between them, as each candidate contains port numbers allocated specifically for the media and the target it is to be sent to.
If you know how to set up one connection, you know how to set up two. Just keep things separate.
just set 1 connection and try to use the direction: many to many
Assume I'm working on a multiplayer online game. Each group of players may start an instance of the game to play. Take League Of Legends as an example.
At any moment of time, there are many game matches being served at the same time. My question is about the architecture of this case. Here are my suggestions:
Assume we have a cloud with a gateway. Any game instance requires a game server behind this gateway to serve the game. For different clients outside the cloud to access different game servers in the cloud, the gateway may differentiate between connections according to ports. It is like we have one machine with many processes each of them listening on a different port.
Is this the best we can get?
Is there another way for the gateway to differentiate connections and forward them to different game instances?
Notice that these are socket connections NOT HTTP requests to an API gateway.
EDIT 1: This question is not about Load Balancing
The keyword is ports. Will each match be served on a different port? or is there another way to serve multiple services on the same host (host = IP)?
Elaboration: I'm using client-server model for each match instance. So multiple clients may connect to the same match server to participate in the same match. Each match need to be server by a match server.
The limitation in mind is: For one host (=IP) to serve multiple services it need to provide them on different ports. Match 1 on port 1234. So clients participating in match 1 will connect to and communicate with the match server on port 1234.
EDIT 2: Scalability is the target
My match server does not calculate and maintain the world of many matches. It maintains the world of one match. This is why each match need another instance of the match server. It is not scalable to have all clients communicating about different matches to connect to one process and to be processed by one process.
My idea is to serve the world of each match by different process. This will require each process to be listening on a different port.
Example: Any client will start a TCP connection with a server listening on port A. Is there is a way to serve multiple MatchServers on the same port A (so that more simultaneous MatchServers won't result in more ports)?
Is there a better scalable way to serve the different worlds of multiple matches?
Short answer: you probably shouldn't use proxy-gateway to handle user connections unless you are absolutely sure there's no other way - you are severely limiting your scaling ability.
Long answer:
What you've described is just a load balancing problem. You can find plenty of solutions based on given restrictions via google.
For League Of Legends it can be quite simple: using some health-check find server with lowest amount of load and stick (kinda like sticky sessions) current game to this server - until the game is finished any computations for particular game are made there. You could use any kind of caching mechanism to store game - server relation for subsequent requests on gateway side.
Another, a bit more complicated example could be data storage for statistics for particular game - it's usually solved via sharding which is a usual consequence of distributed computing. It could be solved this way: use some kind of hashing function (for example, modulo) with game ID as parameter to calculate server number. For example 18283 mod 15 = 13 for game ID = 18283 and 15 available shards - so 13th server should store/serve this data.
Main problem here would be "rebalancing" - adding/remove a shard from cluster, for example.
Those are just two examples, you can google more of them using appropriate keywords. Just keep in mind that all of this is just a subset of problems of distributed computing.
I'm trying to build instant messaging app. Clients will not only send messages but also often send audios. And I've decided to use websocket connection to communicate with clients. It is fast and allows to send binary data.
The main idea is to receive from client1 message and notify about it client2. But here's the thing. My app will be running on GAE. And what if client1's socket is opened on server1 and client2's is opened on server2. This servers don't know about each others clients.
I have one idea how to solve it, but I am sure it is shitty way. I am going to use some sort of communication between servers(for example JMS or open another websocket connection between servers, doesn't matter right now).
But it surely will lead to a disaster. I can't even imagine how often those servers will speak to each other. For each message server1 should notify server2, server2 should notify client2. But things become even worse when serverN comes into play.
Another way I see this to work is Firebase. But it restricts message size to 4KB. So I can't send audios via it. As a solution I can notify client about new audio and he goes to my server for it.
Hope I clearly explained the problem. Does anyone know how to solve it? Or maybe there are another ways to build such apps?
If you are building a messaging cluster and expect communicating clients to connect to different instances of the server then server-server communication is inevitable. Usually it's not a problem though.
First, if you don't use any load balancing your clients will connect to the same server 50% of time on average (in case of 2 servers).
Second, intra-datacenter links are fast and free in all known public clouds.
Third, you can often do something smart on the frontend to make sure two likely to communicate clients connect to the same server. For instance direct all clients from the same country to the same server using DNS load balancing.
The second part of the question is about passing large media files. It's a common best practice to send it out of band - store on the server and only pass the reference to it. Like someone suggested in the comment, save the audio on the server and just send a message like "audio is available, fetch it from here ...". You don't need to poll the server for that. Just fetch it once when the receiving client requests it.
In general, it seems like you are trying to reinvent the wheel. Just use something off the shelf.
Let all client get connected to multiple servers and each server keeps this metadata
A centralized system like zookeeper stores active servers details
When a client c1 sends a message to client c2:
the message is received by a server (say s1, we can add a load balancer to distribute incoming requests)
s1 will broadcast this information to all other servers to get which server the client c2 is connected to OR a better approach to use consistent hashing which decides which server the client can connect to & in this approach message broadcast is not required
the corresponding server responses to server s1 (say s2)
now s1 sends the message m to s2 and server s2 to client c2
Cons of the above approach:
Each server will have a connection with the n-1 servers, creating a mesh topology
Centralized system (zookeeper) becomes a single point of failures (which is solvable)
Apps like Whatsapp, G-Talk uses XMPP and TCP/IP.
I'm looking for the name of a protocol and example code that permits handing off IP/port connections to establish unmediated P2P after introduction through a server.
Simple example:
You and I both start chat programs that connect to chatintroduce.com (fictional server). I send you a "Hi! Wanna chat?" message. It doesn't get sent. Instead my chat program tells chatintroduce to send your chat program a request for connection. You respond to a prompt and your chat program tells chatintroduce to broker the connection. Chatintroduce establishes an initial two-way connection between us. Now, this final step is important, chatintroduce releases control and our two chat programs now talk directly to each other without any traffic through chatintroduce.
In other words, I construct packets which have your IP address and you receive them without interference from firewalls, NATs or any other technologies. In other words, true peer-to-peer connection independent of intermediate server.
I need to know what search terms to use to find appropriate technology. An RFC name would suffice. I've been searching for days without success.
I think what you are looking for is TCP/UDP hole punching which typically coordinates the P2P connection using a STUN server to determine the "capabilities" of the firewalls (e.g. is it a full cone nat? symmetric?).
https://en.wikipedia.org/wiki/Hole_punching_(networking)
We employed this at a company I worked for to create a kind of BitTorrent that could circumvent firewalls for streaming video between two peers.
Note that sometimes it is NOT possible to establish a connection without the intermediary.
What you are looking for is ICE protocol. RFC 5245. This protocol is used for connecting two peers through NAT traversal. There are some open source libraries and also some proprietary libraries for this. You can search google with ICE implementation.
You will also need to read about some additional protocols. These are used with ICE protocol. They are STUN and TURN.
For some cases you can't make P2P call 100% time. You will have to use a relay server. Like if the NAT combination of two peers are Symmetric vs Symmetric/PRC. That relay server is called TURN server.
Some technique like Port forwarding and TCP/UDP hole punching will help you to increase P2P rates.
See this answer for more information about which combination of NAT will require a relay server and which don't.
Thank you. I will be looking further into ICE, STUN, TURN, and hole-punching.
I also found n2n which looks like almost exactly what I wanted.
https://github.com/meyerd/n2n
http://xmodulo.com/configure-peer-to-peer-vpn-linux.html
With n2n, one makes a VPN with a super node that all other edge nodes know.
But once the introductions are made, the super node can be absent.
This was exactly what I wanted. I hope it works across platforms (linux, MacOS, Windows).
Again, I am still researching before implementation, so your advice was very important to me.
Thank you.
Use PJNATH. Its open source.
http://www.pjsip.org/pjnath/docs/html/
There is not much open source on NAT Traversal. As far as I know PJNATH is good.
For server you can use Google's Open source STUN and TURN server.
I am still learning about SIP and all its protocols, specifically trying to integrate PJSIP into an iPhone application to make p2p calls.
I have a question about a peer 2 peer connection using PJSUA. I am able to
make calls perfectly to other clients on my local network by calling directly using the URI:
sip:192...*:5060
I am curious if this will work for
making direct calls to other SIP URIs that are not on the local
network without using server configuration - if not this way, is there another way of making p2p calls without server configuration?
thanks in advance,
You can make calls without server configuration, as a general principle, but something needs configuring. As mattjgalloway points out in the comments below your question, the most robust solution is a can of worms involving ICE which provides a kind of "umbrella" protocol for things like STUN.
Last time I touched this issue, I had the requirement that I couldn't use internet-based SIP servers to help. I came up with the idea of a registry of sorts: your client can define a bunch of "address spaces" with particular routing requirements. For SIP URIs in your LAN, you define no routing; for URIs in your company's VPN-accessed network, you define a route passing through your VPN connection; for everything else you define a route through your internet router.
By "define a route", I mean that when you place a call to a URI in some particular address space, you store what IP will go into a Contact header, what Route headers you might need, and so on.
Thus, the process of making a call becomes:
Look up in the set of address spaces for a match.
Ask that address space for the suitable bits needed to make a workable INVITE (appropriate Contact header details, Route headers, etc.)
Construct a normal INVITE, mutating as necessary for the previous step.
Send the INVITE as normal.
This essentially reproduces half of what ICE would give you, in a manually administrated form. "Half", because this ensures that one SIP agent can make calls such that the SIP routing all works. The missing half is you still need some kind of registrar somewhere, and each agent in your contact list needs to have the necessary setup to receive incoming calls. (If an agent's behind a NATting internet router, the router would need to either run a SIP proxy, or forward ports 5060, 5061 to a particular machine (which might be an agent, or a proxy serving the LAN's agents).
It is, indeed, a large can of worms.
The basic issue is to solve the problem of getting transport ports anywhere on the internet for multimedia traffic.
Many companies/experts have tried to solve this situation. A possible way out of is to buy a domain and setup a basic registrar using YATE or Asterisk on an address accessible from the internet and configure it to also use ICE as needed. Your iphone application at both ends could register automatically to it upon start. Then make P2P calls.