Cheap Point-To-Point VoIP Through NAT? 35
An anonymous reader asks: "70% of my phone bill comes from calls to a few colleagues. We all have 'broadband' internet access (at least 100 kbit/s upstream) and are behind NATs, so we can share our access with the rest of our house-mates. The OS most used is Linux. In order to lower our phone bills I'm looking for a Point-to-Point audio tool which enables you to pass relatively easily through the NATs. I've had a look at Speak-Freely, which is quite nice as it sports things like GPG-encryption. But it uses two UDP and one TCP ports which is a bit much and not very NAT friendly. I wouldn't like to use commercial tools with central servers like Skype. What would be ok is to use a webserver to serve as a kind of starting point where you would update your IP address and ports. But it should be possible to give your mom and pop webhoster to set up or even better just a cgi-script which interacts with the clients via http or https. The audio data itself shouldn't be routed over a server (what a waste of bandwidth). Thanks for all ideas."
openvpn (Score:2)
open source, cross platform
here [sourceforge.net]
Re:openvpn (Score:2)
Re:openvpn (Score:1)
VPN: A different, and useful, approach (Score:2)
Don't try and fight with NAT's, wonky clients, etc., just VPN the lot together and make it all look like a simple little network. Takes the whole question and approaches it from an entirely different, and sound, angle. That's not flaky; that's inspired.
Heck, not just chat but file sharing, white boarding, remote printing, and everything else between these folks will then be trivial too, probably their next request anyway.
OpenVPN is pretty easy to set up, even
IPv6 and Teredo (Score:3, Informative)
Use Teredo and whatever protocol you like.
Teredo is a way to give yourself a realworld IPv6 address, even though you are stuck behind NAT (and without cooperation from the NAT device, like uPnP requires).
Basically Teredo tunnels IPv6 packets over UDP, and relies on the fact that most NAT's reuse the same source port for all udp packets that you send that have the same source address internally.
All your application only need to support IPv6. There are Teredo implementations for Linux and FreeBSD [simphalempin.com] and Teredo is built into Windows SP2 [microsoft.com]. Teredo also supports two people both behind NAT to talk to each other directly in almost all common circumstances.
So go add IPv6 support to your applications, and recommend your users use Teredo to defeat NAT!
Re:IPv6 and Teredo (Score:3, Informative)
Um, no, it's built into the Advanced Networking Pack for Windows XP [microsoft.com] - which is not installed by default.
Re:IPv6 and Teredo (Score:1)
Re:IPv6 and Teredo (Score:2)
Re:IPv6 and Teredo (Score:2)
Quotes:
and
-- IPv6 [microsoft.com]
Re:IPv6 and Teredo (Score:2)
I would really like to see a 6to4 gateway function become a standard vendor feature on popular mass-market routers like the Linksys WRT54G. Since most DSL and cable modem ISPs still give their cu
Re:IPv6 and Teredo (Score:2)
My biggest problem at the moment is that Linux doesn't do particularly good source address selection for IPv6 addresses, in fact it uses the most recently added address to an interface, which if you have 6to4 *and* a slow, laggy tunnel which takes ages to initialise, then all the source addresses on your packets will be via the slow, laggy tunnel. Gnrrg.
I
Re:IPv6 and Teredo (Score:2)
IPv6 has the advantage that it pushes some of the route selection back to the application where the user can control it. IPv6 also has the disadvantage that it pushes some of the route selection back to the application where the user
Re:IPv6 and Teredo (Score:2)
what's wrong with skype? (Score:2)
Re:what's wrong with skype? (Score:2)
The nice thing about it is that it busts NAT like it wasn't even there, and it "just works."
Re:what's wrong with skype? (Score:2)
The voice is point-to-point; the signalling and control channels are P2P (bouncing off other hosts) to get around NAT.
Asterisk..... (Score:1)
VoIP over NAT (Score:5, Informative)
Having said that, where've you been for the last couple of years? There are free registrars that let you use rfc compliant VoIP like SIP: FWD [freeworlddialup.org], IPTel [slashdot.org]. You register there, but you communicate directly between your internet connections. This is really something like web page with your IPs, but automated. Kphone or Linphone are good for it on Linux.
You have to set up some kind of NAT traversal. You can set up port forwarding on the NAT and/or use STUN server.
Also, Skype isn't communicating via server. Skype only authenticates with server, but communication more or less is point to point. When the Skype client is unreachable directly, you communicate with it via third party (i.e. any Skype client with externally open ports). And the communication is encrypted with AES in order to avoid snooping by your
There's also teamspeak which requires extrenally running server (there are some servers publically available) but works like a charm with every kind of NAT, because all the communication goes thru server.
Robert
Re:VoIP over NAT (Score:2, Interesting)
Maybe something like http://chownat.lucidx.com/ could be integrated into other software.
Re:VoIP over NAT (Score:3, Informative)
Unfortunatelly, there's no way for the clients alone to initiate this transfer. They have to know:
Now, there are some "middleman" servers like STUN that will take care of some of this, but requirement 3 may be impossible to to fulfill.
You see, normally when you send packets through NAT, it rewrites source port and address. In case of Linux, if the port is free on firewall/nat box
Re:VoIP over NAT (Score:2, Informative)
Of course, running an asterisk server gives you a lot more options and is definately the geek thing to do!
skype (Score:4, Interesting)
Skype is nat friendly. All you need to do is forward one port. If you don't, the traffic will still get through by routing through people who are NOT on a nat, encrypted end to end.
I would say that Skype is the most NAT friendly of any of the consumer voice over ip programs, and the voice quality is superior.
Go with Skype; you won't regret it.
Re:skype (Score:2)
Use Freeworld Dialup! (Score:2, Interesting)
SIP based VoIP, Asterisk [asterisk.org] compatible if you want to get fancy, uses STUN to traverse nat'ing firewalls. They even sponsor a few SIP clients so it's all free, and you can buy a cheap hardware SIP phone or interface and make the calls from a real phone instead of a PC.
SipPhone (Score:1)
SipPhone is the BEST option at the moment (Score:2)
1. There are upfront costs for hardware, unless you just go with the free softphone (X-ten lite). The hardware runs around $50.
2. Quality is not so good if you have shoddy upload rates (but this is general downside to VoIP in the real world and not unique to SIPphone).
But the pros are definitely worth the cost:
1. The ability to call other VoIP users in ot [sipphone.com]
linphone (Score:2)
VPN (Score:1)
As a bonus, all calls and any other data between the sites will be encrypted.
Teamspeak. (Score:2, Informative)
Why not (Score:3, Insightful)
FWD (Score:2)
Run your VOIP through... (Score:1)
rat (robust audio tool) (Score:1)
you specify the other end's ip address and single udp port. easy to port-forward.
it doesn't encode end-point data in stream, so rat won't get all confused when the other end identifies itself as a non-routable ip address, as with some protocols.
and with multicast, you can do teleconferencing with multiple people.