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VOCAL: Open Source VoIP Software for Linux
An Anonymous Coward writes: "While most Open Source projects are applications and utilities intended for single users, David Bryan and David Kelly did something different.
They created an infrastructure project -- a VoIP phone system that either can run on a single box attached to a couple of IP phones or can scale up to a network of hosts processing hundreds of
calls between thousands of users. In this informative technical article at ELJonline, Bryan and Kelly detail the 'Vovida Open Communications Applications Library' ('VOCAL') project, a fully functional phone system that can run on either Red Hat Linux or Sun Solaris."
(Funny, this is the second post of mine that has that title)
Yahoo Broadband is offering VoIP Internation Telephony at 7.5 yen/ 3 minutes. Very good deal.
It's very clear as well.
I have been pwned because my
Apache, sendmail, bind... famous open source projects designed for single users. It's a good idea someone came along to do this new innovative infrastructure sorta stuff. Maybe someday we can have a whole inter-network of computers using open protocols.
Instead os using the SII protocol as VOCAL does, you could also use H.323 for example the OpenH323 Gatekeeper, now called The GNU Gatekeeper.
Fast forward to 2002. Microsoft still kind of ships Netmeeting with Windows XP Home, but there are no shortcuts, their documentation discourages you from using it (it also blue-screened my XP machine when I tried running it). Instead, they want you to use Microsoft Messenger, which only seems to want to talk through Microsoft's servers. Yahoo! give you video conferencing, but only through Yahoo! messenger and only on Windows. CU-SeeMe doesn't seem to exist anymore. In fact, I couldn't find any Windows or OSX H.323 implementations.
Instead, now the next thing seems to be SIP (Session Initiation Protocol, which is curiously what Vovida is based on. Well, it's kind of like HTTP, and that's nice compared to H.323's ASN protocols. MSN Messenger seems to be using it. There is Linphone, which is SIP based and works on Linux.
But... how do we do cross platform video conferencing now? Microsoft Messenger may speak SIP, but as far as I can tell, it doesn't let me do machine to machine calls. Even if it did, GnomeMeeting doesn't seem to support SIP (yet?) and Linphone doesn't do video. And MacOSX, as far as I can tell, is almost completely out in the cold; at least, I couldn't find any commercial video conferencing software for it. The closest is the OpenH.323 sample applications, running under X11 on MacOSX. That's not exactly what you can ask average Mac users to use.
So, if I want to do cross-platform video conferencing between Linux, Windows, and/or Macintosh, what software and protocols should I use?
I work for a company that has a (very) new product, called the VoIP Development System (VDS), that is a testbed and diagnostic application for VoIP systems. Apparently, the software is so new that it is not even featured on the front page of our web site.
Anyway, VoIP architecture is, of course, integrated into the software. On a daily basis, the VoIP package development team is coming to us, the senior programmers, and asking for assistance and references for developing various parts of the code, ranging from simple GUI items to items regarding the infinitely more complex network architecture implementation.
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Because of this, I know how difficult and intense the development of VoIP systems is. Kudos go out to the developers for this project. Keep up the good work; you're doing an excellent thing for the open source and free software communities.
Now, whether free software will release a person or company from the cost of buying the hardware to support an extensive network of VoIP systems is another problem, entirely.
Why wasn't a link to the project's actual webpage in the submission? Here it is.
Now all we need is to create some sort of secure database, where people could donate use of thier landline (for local calls in their area code) for a period of time in exchange for credit to make calls to other area codes. It would be similar to ham radio telephone relays. Now all we need is a single combo ip/telephone # so that it would call your computer first (for long distance) and then your home phone. I suppose this could be implemented with dyndns.org or another similar service. Anything to spite qwest!
Asterisk is the VOIP/phonesystem software package for linux, and has been for over 3 years now. It sounds like this VOCAL is a framework for call routing (just like asterisk) but without the POTS gateway abilities.
Also, I have had great luck with my 20 VOIP blasters running in basically a P2P mode with only asking for directions from the phonebook server...
I have yet to impliment a POTS gateway using asterisk because the internet phonejack cards are horribly expensive. Anyone else here doing linux Voip?
Do not look at laser with remaining good eye.
I sure would like to see that in action. If it is really that scaleable, and if it works as well as they say, then this could be some serious competition for Lucent and Nortel platforms which cost a hell of a lot more money.
-- -- Warning. Do not stare directly at the sun.
Now if I could tell this box, "take calles on this line and send them to port 5433 on 192.168.1.23 as a 64k mu-law stream" then I would have 99% of what I need for a VoIP gateway to the telephone company.
How are you going to handle call setup and teardown? There are a multitude of things you have to deal with in IP telephoney that go beyond the functionality provided by protocols like TCP and UDP. All SIP does is provide call signalling. The actual voice stream is handed off to SDP which specifies which voice encoding type to use. Such as G.711 Mulaw which you referred to.
The next great leap in VoIT will come from someone thats got the balls to do ISDN over IP and write some sample code that works and then an RFC. Till then its just a sick game.
What do you think H.323 is? Take a look at the signalling required to setup a call in H.225 compared to Q.931. The only thing you're missing in H.225 is the ACK's and those are provided by the underlying TCP protocol.
The people from cyclades said they looked at doing VoIP but everyone wanted "standards" which they didn't or couldn't squeeze into the RAS box. I don't think they ever thought that it wasn't that hard. My Cisco AS5300 doesn't have any problems with converting incoming ISDN to H.323 or SIP. Try a 2600 even.
The Information Revolution will be fought on the command line.
http://www.cs.columbia.edu/sip lists implementations. There aren't many for Unix-related systems, but our CINEMA sipc tool does run on all common Unix/Linux platforms and supports audio, video and other conferencing functionality. It is not free software.
Most SIP tools allow direct communications. Some may need a proxy server. A proxy server is somewhat similar to an H.323 gatekeeper. The VOCAL set includes this, but there are many others, too, listed at the URL above.
let me start by saying that SIP is very, very good. all of Cisco's IP telephony products are based on SIP now, instead of their previous mucky protocols. many larger vendors are also supporting SIP, as it an RFC and other goodness. VOCAL, which I have had the pleasure of working with recently, is very well designed, and (in my biased opinion) is nice because it's not really "linux centric". we did a test deployment on several FreeBSD systems functioning as a Vmail system, inter-office IP phone calls (to both Cisco SoftPhone clients and actual cisco IP phones) and working with a cisco 3640 router with two VIC-2FXO cards (which provides 4 lines out to the PSTN through our PBX). the mapping is pretty easy from cisco VOCAL, and the VOCAL user agent piece is pretty cool, although right now it's just a very basic CLI tool under windows. We really haven't tried using a unix system as there are few end users at a brokerage firm who actively use unix as a client desktop!
definitely check out the cisco SIP offerings, as well as the excellent vovida project and tools. they have a lot more to offer as well, including some frivolous PSTN gateway stuff using those internet linejack bits. I personally agree with what they've been doing, which is building an enterprise-class IP telephony infrastructure, rather than wasting time on stuff for college kids to avoid phone bills. but then again your needs may differ from ours. YMMV!
EOM