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.
Related
I have a DID from nexmo with voice support and I added SIP forwarding to one of my freeswitch servers (B). I have two freeswitch servers A (asia) and B (europe), If a caller calls from Asia, how do I make sure it gets routed to A server so its close to the user?
Based on my research I might need to do lua scripting for dynamic dialplan with redirect. But even this, I'm not even sure how to get the caller's location so I can decide if to which server to route the call.
Is there an easy way to do this?
what do you know about the caller?
If it's the IP address, then a lookup in WHOIS database should work. Also there are projects like freegeoip.net that deliver you the geographical location for an IP address.
Otherwise, if caller's phone number is only know, then you need to have a database of all country codes around the world, and build your lookup logic based on that.
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 think about the following setup, but I do not know how to connect the main parts.
One the one side there is a Kamailio SIP server. This server provides VoIP connectivity within a certain network (non public intranet).
On the other side there is a SIP provider. This provider provides a single telephone number from the PSTN. Let's say the number is 0034-443322.
Both components are working fine so far.
I want to use that number as a dial-in to my private network. A user with number 8282 in my network should be reachable via 0034-443322-8282 from the outside world. Outgoing calls aren't necessary.
How to reach my goal? I don't know what to look for :/ Any ideas are very welcome :)
kind regards
K.A.
If your PSTN gateway can be reached by dialing the full number (including the extension), simply let the gateway forward every incoming call to your Kamailio instance which will forward the call to the appropriate user. For that, you need to create your users (known as subscribers in Kamailio) and they need to register to your Kamailio instance so that they can receive incoming calls. Regarding mapping extensions to users, you can simply let the extension be the username; or you can add extensions as aliases of the subscribers.
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.
Here is my problem. I do distributed calling with 10-15 Asterisk boxes. They each have a different IP address. My provider wants me to use IP based authentication, which can be a pain (the IP's change on a frequent basis).
I was thinking of setting up one Asterisk box to relay calls from all the others. That way I'd just need to set up the SIP trunk to the provider on this one box/IP.
Can anyone tell me anything special I would need to do to set this up - either in the dial plan or otherwise? I do not want all the media to run through the central box, I merely want it to set up the calls and allow the other boxes to talk directly to the provider once the call is set up.
Thanks in advance for any suggestions.
I believe a sip proxy like openser/kamailio or opensips would fit in best here. You could set the proxy up to relay all of the sip signalling to your cluster of asterisk boxes.
Here's an example for doing this with Kamailio.
Also:
I do not want all the media to run through the central box, I merely want it to set up the calls and allow the other boxes to talk directly to the provider once the call is set up.
Make sure that's groovy with your provider. A lot of ITSPs expect to signal and send media to pre-defined IP addresses. Generally they're talking to an SBC, which is expected to anchor the media and send it elsewhere..