What is the purpose for using sip with VOIP ? is it just to know the remote ip address ?
If i know (by server) the remote IP address ,then established a direct TCP socket connection for call negotiation, and send the media over RTP protocol ,so am i still need sip protocol ? or how can sip help me here?
The Session Initiation Protocol does rather a lot more than find out a remote IP address/port/transport triple.
It lets two parties
negotiate the media streams (including codecs and transports) and
establish commonly understood extensions to the protocol.
It also describes how to build scalable infrastructure (proxies, using SRV and NAPTR records, back to back user agents), location services and a host of other details that go into making a voice (or any other kind of) call to arbitrary third parties.
then you have implemented a sip alternative. sip (session instanciation protocol) does just control the "phone call". if you want to do that on your own, why not? the only problem would be that there are many sip clients and just one (or few) clients using your protocol.
SIP allocates a IP:port to a voip call. The RTP flows (one for each direction) will then use this IP:port as a destination address. If you have only one static RTP flow to send to your server, it may be useful and ok to do what you said.
Otherwise, if there are many clients, or if your system has to change a lot, it's better to use a polished protocol which will dynamically allocate ports and establish your sessions.
nobody forces you to implement a standard.
e.g. why do you implement the media stream in RTP? most likely because you already have code that "talks" RTP (e.g. a library, or a raedymade application).
the nice thing about standards is, that it will work "out-of-the-box" with all other applications implementing the same standard.
if it is an open standard, there's another nice thing: other people have already spent a lot of brain power into getting the implementation right. you don't need to fall into the same problems.
Related
I'm writing a server that, among other things, needs to be constantly sending data in different multicast addresses. The packages being sent might be received by a client side (an app) which will be switching between the mentioned addresses.
I'm using Perfect (https://github.com/PerfectlySoft/Perfect) for writing the server side, however had no luck using the Perfect-Net module nor using CocoaAsyncSocket. How could i implement both the sender and the receiver using swift? Any could snippet would be really useful.
I've been reading about multicasting and when it comes to the receiver, i've notice that in most languages (i.e. java or c#) the receiver often indicates a port number and a multicast ip-address, but when is the connection with the server being made? When does the socket bind to the real server ip-address?
Thanks in advance
If we talk about the TCP/IP stack, only IP and UDP support broadcasts and multicasts. They're both connectionless, and this is why you see only sending and receiving to special multicast addresses, but no binds and connects. You see it in different languages because (a) protocols are language-agnostic and (b) most implementations put reasonable efforts in trying to be compatible with BSD sockets interface.
If you want that true multicast, you'll need to find a swift implementation of sockets that allow setting options. Usual names for this operation is setsockopt. Multicast sender side doesn't need anything beyond a basic UDP socket (I suggest using UDP, not IP), while sender needs to be added to a multicast group. This Python example pretty much describes it.
However, it's worth noting that routers don't route broadcasts and multicasts. Hence you cannot use it over internet. If you need to use internet in your project, I'd advise you to use TCP - or websockets if your clients are browsers - and send messages to "groups" of them manually.
I guess you actually want Perfect-Kafka or Perfect-Mosquitto - Message Queue which allows a server to publish live streams to the client side subscribers. Low-level sockets will not easily fulfill your requirement.
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.
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 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.
Is it possible to multiplex sa ocket connection?
I need to establish multiple connections to yahoo messenger and i am looking for a way to do this efficiently without having to hold a socket open for each client connection.
so far i have to use one socket for each client and this does not scale well above 50,000 connections.
oh, my solution is for a TELCO, so i need to at least hit 250,000 to 500,000 connections
i'm planing to bind multiple IP addresses to a single NIC to beat the 65k port restriction per IP address.
Please i would any help, insight i can get.
**most of my other questions on this site have gone un-answered :) **
Thanks
This is an interesting question about scaling in a serious situation.
You are essentially asking, "How do I establish N connections to an internet service, where N is >= 250,000".
The only way to do this effectively and efficiently is to cluster. You cannot do this on a single host, so you will need to be able to fragment and partition your client base into a number of different servers, so that each is only handling a subset.
The idea would be for a single server to hold open as few connections as possible (spreading out the connectivity evenly) while holding enough connections to make whatever service you're hosting viable by keeping inter-server communication to a minimum level. This will mean that any two connections that are related (such as two accounts that talk to each other a lot) will have to be on the same host.
You will need servers and network infrastructure that can handle this. You will need a subnet of ip addresses, each server will have to have stateless communication with the internet (i.e. your router will not be doing any NAT in order to not have to track 250,000+ connections).
You will have to talk to AOL. There is no way that AOL will be able to handle this level of connectivity without considering cutting your connection off. Any service of this scale would have to be negotiated with AOL so both you and they would be able to handle the connectivity.
There are i/o multiplexing technologies that you should investigate. Kqueue and epoll come to mind.
In order to write this massively concurrent and teleco grade solution, I would recommend investigating erlang. Erlang is designed for situations such as these (multi-server, massively-multi-client, massively-multithreaded telecommunications grade software). It is currently used for running Ericsson telephone exchanges.
While you can listen on a socket for multiple incoming connection requests, when the connection is established, it connects a unique port on the server to a unique port on the client. In order to multiplex a connection, you need to control both ends of the pipe and have a protocol that allows you to switch contexts from one virtual connection to another or use a stateless protocol that doesn't care about the client's identity. In the former case you'd need to implement it in the application layer so that you could reuse existing connections. In the latter case you could get by using a proxy that keeps track of which server response goes to which client. Since you're connecting to Yahoo Messenger, I don't think you'll be able to do this since it requires an authenticated connection and it assumes that each connection corresponds to a single user.
You can only multiplex multiple connections over a single socket if the other end supports such an operation.
In other words it's a function protocol - sockets don't have any native support for it.
I doubt yahoo messenger protocol has any support for it.
An alternative (to multiple IPs on a single NIC) is to design your own multiplexing protocol and have satellite servers that convert from the multiplex protocol to the yahoo protocol.
I'll trow in another approach you could consider (depending on how desperate you are).
Note that operating system TCP/IP implementations need to be general purpose, but you are only interested in a very specific use-case. So it might make sense to implement a cut-down version of TCP/IP (which only handles your use-case, but does that very well) in your application code.
For example, if you are using Linux, you could route a couple of IP addresses to a tun interface and have your application handle the IP packets for that tun interface. That way you can implement TCP/IP (optimised for your use-case) entirely in your application and avoid any operating system restriction on the number of open connections.
Of course, it's quite a bit of work doing the TCP/IP yourself, but it really depends on how desperate you are - i.e. how much hardware can you afford to throw at the problem.
500,000 arbitrary yahoo messenger connections - is your telco doing this on behalf of Yahoo? It seems like whatever solution has been in place for many years now should be scalable with the help of Moore's Law - and as far as I know all the IM clients have been pretty effective for a long time, and there's no pressing increase in demand that I can think of.
Why isn't this a reasonable problem to address with hardware plus traditional solutions?