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Industry-Standard VOIP Phone Using All Free Software
Ralf Ackermann writes: "Voice over IP on a HardPhone running Linux and
just using Open Source software became real. We have sucessfully installed and tested (interoperability with Cisco
7960 as well as Pingtel xPressa in an environment with a partysip SIP registrar and proxy) the linphone SIP phone on a StrongARM based
TuxScreen.
Here is the link describing the steps for others
to use the setup as well: TuxScreen running SIP. All the infos for setting up a comparable installation can be found on the
URL, please also feel free to ask or drop opinions. Many thanks to the linphone developers as well as to my student Florian
Winterstein (for working on a console linphonec version). The setup (on a StrongARM system) is well suited for PDA (iPAQ) or
wearable environments as well."
Check this out for another linux-based VoIP, standards-compliant (both SIP *and* H323) phone.
It's been out for quite awhile, over a year. My company is a reseller. They're cheap (~ $199 each) and they rock.
Don't sweat the petty things. But do pet the sweaty things.
This is fine for intraoffice communication, but what about the real world? I think that a solution for connection VoIP -> POTS could be easy, with a little knowledge of Linux drivers.
...)
Get an older box (P2 400 or so), with plenty of PCI slots, and preferrably an onboard NIC also. Get some Winmodems equal to the # of pci slots.
WinModems, even in all of their Microsoft-sponsored godless evil towards open source platforms, are basically A/D and D/A converters hooked to a phone jack. It should be relatively simple to talk (no pun intended) to them in software and use one as an interface to POTS. It has all of the neccesary hardware, and writing a sound driver for it shouldn't be too difficult. A brand of WinModem with fairly standard hardware could be decided on by the implementer, and drivers written for that. (Winmodems? Standard?
Client software with available source code could be modified to use those, as well as control the phone-line functions. Just run an instance per WinModem.
Honestly, I think that this could work, and it would be a great hack to accomplish. Anyone fancy a go at it?
Every cloud has a silver lining (except for the mushroom shaped ones, which have a lining of Iridium & Strontium 90)
(shameless plug) Take a look at Speex, an open-source, patent-free speech codec (Speex is to speech what Vorbis is to music). Speex should soon be available in Linphone too!
Opus: the Swiss army knife of audio codec
Open Source VoIP and telephony is tracked at http://www.linuxtelephony.org.
VoIP hardware (PCI cards) is IIRC $79, gateways are $179, drivers have been in the kernel since (again IIRC) 2.2.16. http://www.quicknet.net.
VoIP in the U.S. is almost pointless because the PSTN is too good. No one wants near perfect when perfect is cheap and easy. In the third world, if you can get a phone circuit it averages $1.27 per minute, whereas VoIP through a hop-off provider like Net2Phone (http://www.net2phone.com) runs average about $0.23 per minute or less. Straight IP to IP (like across a VPN from company branch to branch) is just the cost of the ISP (usually flat rate). So VoIP cafes are a popular way for the non-super-rich and powerful to make calls to their relatives in the first world.
Personally, the VoIP calls I have made have an almost imperceptible latency problem and sounded *far* better than any GSM call I have ever heard. Then again, these calls were during business hours so net congestion was not an issue for me.
The Ogg Vorbis has a low bit rate mode that is useful for VoIP telephony, and is grossly better than GSM to my ear.
Finally, VoIP is used by the big players here in the US. Qwest and Sprint use VoIP in preference to ATM due to cost of the equipment (Bits per second/price of hardware. IP is more efficient than ATM due to less overhead). Any cross-country call is VoIP nowadays.
Just my two bits.
-C