Please let me know what architecture do VoIP applications use, P2P or Client-Server?
Thank you.
Some of each in general. There are three protocols involved, usually. One of them, for example SIP, is used to establish the connection. you need a server for that because someone has to establish the original connection; that means advertising availability and such. The other two are essentially always RTP and RTCP -- "real time protocol" and "real time control protocol", and those are better P2P, because you want fast transmission with no intermediate bottleneck.
There's a nice article on the whole discussion here.
There's usually some kind of "presense server": devices register ("I exist here!") and calls are established via the server (when you say "I want to connect to device (555) 555-1234" that connection request is routed via presence servers).
After the call is established and the real-time voice/media is streaming, that traffic is usually peer-to-peer (bypassing any central server), except if there's a complication like both devices being behind firewalls.
Related
I recall reading an article about a proposed way to do this. If I recall correctly, the researchers successfully created a connection to a client on another network without port forwarding by sending HTTP packets to each other (Alice pretends that Bob is an HTTP web server while Bob pretends Alice is a web server).
I'm not sure if that makes sense, but does anyone know where I can find the article or does anyone have any other ideas how to connect two clients together without a central server or port forwarding?
Is it even possible?
Edit: I would know the IPs of both computers and port the program listens on.
It is possible. I see at least 2 parts to your question. (It is not going to be HTTP packet. It is a lot more complex than that.)
First off, I believe you might be talking about a concept called decentralized P2P network. The main idea behind a decentralized peer-to-peer network is the fact that nodes conjoint in such a network will not require central server or group of servers.
As you might already know, most common centralized peer-to-peer networks require such centralized system to exchange and maintain interconnectivity among nodes. The basic concept is such, a new node will connect to one of the main servers to retrieve information about other nodes on the network to maintain its connectivity and availability. The central system gets maintained through servers constantly synchronizing network state, relevant information, and central coordination among each other.
Decentralized network, on the other hand, does not have any structure or predetermined core. This peer-to-peer model is also called unstructured P2P networks. Any new node will copy or inherit original links from the "parent" node and will form its own list over time. There are several categories of decentralization of such unstructured networks.
Interestingly enough, the absence of central command and control system makes it solution of choice for modern malware botnets. A great example could be Storm botnet, which employed so-called Passive P2P Monitor (PPM). PPM was able to locate the infected hosts and build peer list regardless whether or not infected hosts are behind a firewall or NAT. Wikipedia's article Storm botnet is an interesting read. There is also great collaborative study called Towards Complete Node Enumeration in a Peer-to-Peer Botnet, which provides excellent conceptual analysis and techniques employed by Storm botnet network.
Second of all, you might be talking about UDP hole punching. This is a technique or algorithm used to maintain connectivity between 2 hosts behind NATed router/gateway using 3rd comment host by means of a third rendezvous server.
There is a great paper by Bryan Ford, Pyda Srisuresh, and Dan Kegel called Peer-to-Peer Communication Across Network Address Translators.
As answered, a peer-to-peer connection requires establishment of a connection between two (presumably) residential computers, which will necessitate punching holes through both of their firewalls. For a concrete example of hole punching, see pwnat: "The only tool to punch holes through firewalls/NATs without a third party". The process, put simply, goes like this:
The "server" (who doesn't know the client's IP address, but the client knows the server's) pings a very specific ICMP Echo Request packet to 1.2.3.4 every 30 seconds. The NAT, during translation, takes note of this packet in case it gets a response.
The client sends an ICMP Time Exceeded packet to the server, which is a type of packet that usually contains the packet that failed to deliver. The client, knowing in advance the exact packet that the server has been sending to 1.2.3.4, embeds that whole packet in the Data field.
The NAT recognizes the Echo Request packet and happily relays the whole Time Exceeded packet, source IP and all, to the correct user, i.e. the server. Voila, now the server knows the client's IP and port number.
Now that the server knows the address, it begins to continually send UDP packets to the client, despite the fact that the client's NAT did not expect them and will therefore ignore them all.
The client begins sending UDP packets to the server, which will be recognized by the server's NAT as a response to the server's packets and route them appropriately.
Now that the client is sending UDP packets to the server, the server's stream of UDP packets starts getting properly routed by the client's NAT.
And, in 6 easy steps, you have established a UDP connection between a client and a server penetrating two residential firewalls. Take that, ISP!
I know p2p software like Skype is using UDP hole punching for that. But what if one of the clients is a web browser which needs to download a file from another client (TCP connection instead of UDP)? Is there any technique for such case?
I can have an intermediate public server which can marry the clients but I can't afford all the traffic between these clients go through this server. The public server can only establish the connection between the clients, like Skype does, and that's all. And this must work via TCP (more exactly, HTTP) to let the downloading client be a web browser.
Both clients must not be required to setup anything in their routers or anything like that.
I'll plan to code this in C/C++ but at the point I'm wondering if this idea is possible at all.
I previously wrote up a very consolidated rough answer on how P2P roughly works with some discussion on various protocols and corresponding open-source libraries. You can read it here.
The reliability of P2P is ultimately a result of how much you invest in it from both a client coding perspective and a service configuration (i.e. signaling servers and relays). You can settle for easy NAT traversal of UDP with no firewall support. Maybe a little more effort and you get TCP connectivity. And you can go "all the way" and have relays that have HTTPS listeners for clients behind the hardest of firewalls to traverse.
As to the answer of your question about firewalls. Depends on how the Firewall is configured. Many firewalls are just glorified NATs with security to restrict traffic to certain ports and block unsolicited incoming connections. Others are extremely restrictive and just allow HTTP/HTTPS traffic over a proxy.
The video conference apps will ultimately fallback to emulating an HTTPS connection over the PC's configured proxy server to port 443 (or 80) of a remote relay server if it can't get directly connected. (And in some cases, the remote client will try to listen on port 80 or port 443 so it can connect direct).
You are absolutely right to assume that having all the clients going through a relay will be expensive to maintain. If your goal is 100% connectivity no matter what type of firewall the clients is behind, some relay solution will have to exist. If you don't support a relay solution, you can invest heavily in getting the direct connectivity to work reliably and only have a small percentage of clients blocked.
Hope this helps.
PeerConnection, part of WebRTC solves this in modern browsers.
Under the hood it uses ICE which is an RFC for NAT hole-punching.
For older browsers, it is possible to use the P2P support in Flash.
I'm developing chat application using app.js which is webkit+node.js framework.
So i have node.js plus bridged web browser environment on both sides.
I want to make file transfer feature somewhat similar to Skype one.
So, initial idea is to:
1.connect clients to main server.
2.Each client gets ip of oposite ones.
3.Start socket or websocket server on both clients and connect to each other.
4.Sender reads the file and transmits it to the reciver.
Question are:
1.Im not really sure that one client can "see" the other.
2.file is a binary data, but websockets are made for text messages so i need some kind of coding/decoding stuff. I thought about base 64 but it has 30% of "overhead" information. So i need something more effitient (base 128?).
3.If it is not efficient to use websocket should i use TCP sockets instead? What problems can appear if i decide to use them?
Yeah i know about node2node and BinaryJS, i just dont know should i use them or not. And i really what to do something myself.
OK, with your communication looking like this:
(C->N)<->N<->(N->C)
(...) is installed on one client's machine. N's are node servers, C's are web clients.
This is out of your control. Some file sharing apps send test packets from the central server to clients, to check whether ports are open and NAT rules are configured correctly, etc. Your clients will start their own servers on some port, your master server can potentially create a test connection to these servers to see whether they're started correctly and open to the web, BEFORE telling other clients that they can send files.
Websockets are great for status messages from your servers to the web GUIs and general client-to-client communication. For the actual file transfers, I would use TCP sockets, see the next answer. On the other hand base64 encoding is really not a slow process, play with it and benchmark its performance, then decide with some data to back up your decision.
You could use a combination: websockets from your servers to the web GUIs, but TCP communication between the servers themselves. TCP servers (and streams) aren't hard to set up in Node, I see no disadvantages. It might actually be less complicated than installing node2node on those servers, since TCP is already built-in.
I'm developing a SIP mobile softphone, customer needs a complete hiding of SIP messages from softphones to SIP servers as VOIP calls are regionally prohibited, however using TLS connection was not sufficient since the message headers are easily recognized as a SIP message. What are the best common alternative?
what about openvpn, IPSec tunneling?
Transmitting SIP over TLS means the SIP headers will only be viewable if someone is able to compromise your TLS keys, i.e. it's highly unlikely unless some national security agency is on your case.
What you might be encountering is port 5061 being blocked since it's the default and therefore well known SIP TLS port. To get around that simply use a different port for your SIP TLS connection. As far as anyone viewing the traffic goes if it's not suing port 5061 they won't have any idea that SIP is being used in your TLS stream.
Of course you also need to consider the RTP traffic which is what will carry the audio part of the call once SIP has set it up. There are no standardised ports for RTP but some popular VoIP softswitches do use certain ranges by default. For example Asterisk uses UDP 10,000 to 20,000. To work around that you'd really need to use SRTP but that's going to be harder to set up since not that many SIP user agents and servers support it. It will also be easier to detect for someone watching your traffic since even without knowing the contents the profile of RTP packets would be detectable. Still it's likely to need a sophisticated entity monitoring your traffic to detect a VoIP call using SIP over TLS on a non-standard port and SRTP call amongst the general noise of internet traffic.
How can an application be designed such that two peers can communicate directly with each other (assuming both know each other's IPs), but without outgoing connections? That's, no ports will be opened. Bitorrent for example does it, but multiplayer games (as far as I know) require port forwarding.
I'm not sure what you mean by No Outgoing Connections, I'm going to assume like everyone else you meant no Incoming Connections (they are behind a NAT/FW/etc).
The most common one mentioned so far is UPNP, which in this context is a protocol that allows you as a computer to talk to the Gateway and say forward me this port because I want someone on the outside to be able to talk to me. UPNP is also designed for other things, but this is the common thing for home networking (Actually it's one of many definitions).
There are also more common and slightly more reliable ways if you don't own the network. The most common is called STUN but if I recall correctly there are a few variants. Basically you use a third party server that allows incoming connections to try and coordinate a communication channel. Basically, what you do is send a UDP packet to you're peer, which will open up you're NAT for a response, but gets dropped on you're peer's NAT (since no forwarding rule exists yet). Through the connection to the intermediary, they are then told to do the same, which now opens up their NAT, and matches the existing rule in you're NAT. Now the communications can proceed. Their is a variant of this which will allow a TCP/IP connection as well by sending SYN and SYN-ACK messages with some coordination.
The Wikipedia articles I've linked to has links to the relevant rfc's for these protocols on precisely how they work. Essentially it comes down to, there isn't an easy answer, as this is a very network centric problem.
You need a "meeting point" in the network somewhere: the participants "meet" at a "gateway" of some sort and the said "gateway function" takes care of the forwarding.
At least that's one way of doing it: I won't try to comment on the details of Bittorrent... I am sure you can google for links.
UPNP dealt with this mostly in the recent years, but the need to open ports is because the application has been coded to listen on a specific port for a response.
Ports beneath 1024 are called "registered" because they've been assigned a port number because a company paid for it. This doesn't mean you couldn't use port 53 for a webserver or SSH, just that most will assume when they see it that they are dealing with DNS. Ports above 1024 are unregistered, so there's no association - your web browser, be it Internet Explorer/Firefox/etc, is using an unregistered port to send the request to the StackOverflow webserver(s) on port 80. You can use:
netstat -a
..on windows hosts to see what network connections are currently established, including the port involved.
UPNP can be used to negotiate with the router to open and forward a port to your application. Even bit-torrent needs at least one of the peers to have an open port to enable p2p connections. There is no need for both peers to have an open port however, since they both communicate with the same server (tracker) that lets them negotiate and determine who has an open port.
An alternative is an echo-server / relay-server somewhere on the internet that both peers trust, and have that relay all the traffic.
The "problem" with this solution is that the echo-server needs to have lots of bandwidth to accomodate all connected peers since it relays all the traffic rather than establish p2p connections.
Check out EchoWare: http://www.echogent.com/tech.htm