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VOCAL: Open Source VoIP Software for Linux 149

Posted by timothy from the taunting-the-phone-company-dragon dept.
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."
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VOCAL: Open Source VoIP Software for Linux More

VOCAL: Open Source VoIP Software for Linux

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  • In Japan... (Score:3, Informative)

    by ObviousGuy ( 578567 ) <ObviousGuy@hotmail.com> on Wednesday May 15, 2002 @01:05AM (#3522069) Homepage Journal
    (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.

  • yep... (Score:5, Funny)

    by mattdm ( 1931 ) on Wednesday May 15, 2002 @01:05AM (#3522073) Homepage
    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.

    • Indeed, I'm hard pressed to think of one open source project indtended for single user. Then again, maybe I need some sleep...
    • Propably not. I don't think that ISP's will get all excited about increased uploading. They'll just find out ways to block those things, just like they do with p2p software.

  • OpenH323 is also nice (Score:3, Informative)

    by willamowius ( 193393 ) on Wednesday May 15, 2002 @01:09AM (#3522086) Homepage
    Instead os using the SII protocol as VOCAL does, you could also use H.323 for example the OpenH323 Gatekeeper [willamowius.de], now called The GNU Gatekeeper.
    • well i don't know squat about either protocol, but i did notice that VOCAL features:
      H.323 Translator
      The H.323 Translator now supports gateway trunking. Before, it was only supporting NetMeeting as endpoints.
  • Hah! "Most applications" are targeted toward "signle users", what kind of crap is this? What is this site running on?!
  • Seeing as how my company is one of the telecom's that is failing right now, maybe we could use this technology to replace our Sonus Softswitchs. I bet that would save us a whole lot of green. I'm going to have to check it out and see what I can set up.
    • Reducing costs by replacing your telecommunications-grade infrastructure with cheap PCs running this month's patchlevel of Red Hat Linux sounds like a recipe for success to me!
      • I wonder which failing telecom this guy works for. My guess is XO Communications. Or is it Elecrtic LightWave? Maybe Qwest?? Oooh, there's just so many to choose from!
      • Hmm, actually this is exactly the kind of thing Cisco does with their VoIP solution, the call manager runs on win2k servers running IIS to feed the xml to the ip phones, it also does call routing etc. We have had our VoIP solution in place for nearly a year and have not had any downtime yet. We have a cluster of 2 servers running for redundency, but have not needed the secondary to take full load yet.

  • VOCAL (Score:2, Funny)

    by 56ker ( 566853 )
    "'Vovida Open Communications Applications Library' ('VOCAL')" - talk about the award for the most contrived acronym!

  • I was wounrdering (Score:2, Funny)

    by Anonymous Coward
    Community Supported VoIP. then we will have half of the teleco company's calling us pirate's and the other half calling us customers

    man if this is fp the meaning to my life has meaning now.

  • H.323 (Score:2, Informative)

    by zoftie ( 195518 )
    What about open h323. Wasn't that a standard for making reliable phonecalls over packet streams?
    Time to go with industry standard people. Last
    place I worked at, we have implemented H.323 gateway
    and it worked like magic, coupled up with outher
    gateways.
    ... dunno, looks like guys reinventing the wheel all over again.
    p.
    • H.323 works fine, but you need to have some other protocols with it if you want to opereate something approximating to a telephone network (as opposed to a point-2-point connection). In particular this means things like SIP (Session Initiation Protocol).
    • VOCAL can implement H.323, SIP, and MGCP gateways, or a combination of the three. SIP is a new standard that helps to eliminate much of the complexity of H.323, and I think (from having used both H.323 and SIP products) that SIP just slays H.323. while it's still a newer technology, it has much better scalability as well as the ability to be a ubiquitous IP/phone/video service. addressing is simple, it's as easy as sipuser@host.they.are.at. your client (at least in the VOCAL infrastructure) updates a registration server as to your current location, and proxy servers will always move calls in the right direction towards you. Sound like a perfect fit for new Mobile IP networks, anyone? this is a killer app that's just waiting to be implemented...

  • So, what should I do now? (Score:5, Interesting)

    by g4dget ( 579145 ) on Wednesday May 15, 2002 @01:27AM (#3522145)
    H.323 and associated protocols for video conferencing and collaboration have been standardized for a while. They are kind of messy, but there were Windows implementations like NetMeeting, Linux implementations like Open H.323 [openh323.org], and commercial implementations like CU-SeeMe (for Windows and Mac). These things could even talk to one another and to GnomeMeeting.

    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 [columbia.edu], which is curiously what Vovida [vovida.org] 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 [linphone.org], 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?

    • We ha d a go and tried the H.323 to ISDN gateway in our company. It worked like a breeze, right out of the box.

      We were able to connect M$ Netmeeting directly to the server as well as the minimal phone application for windows (yes there is one avail. at Open H.323 Org [openh323.org]).

      I am sory to say that calling the communications prgramm under Linux froze my box completely -- it was probably the soundcard. But when I look an the Gnome or KDE application which are available I think Unix users have a good option to participate.

      When it comes to Mac I must say I have no access to one, so I cannot verify the availability/functionality of any app for MacOS. I do beleive though that under MacOSX the above Unix versions should run very well ?

      When it comes to SIP we do have linphone (Gnome) available as well as a whole rack ot libraries for different languages. All found on Freshmeat Net [freshmeat.net] with the simple query "SIP" .

      No idea about MacOS SIP apps, but the same though as above: MacOSX and Gnome ?
    • Don't forget to check out the speak-freely internet telephone. It's audio only, but is availible for both Unix and Windows. The unix version is via a command-line interface, but there are java and Tcl/Tk front ends. (I gave up and stuck with the CLI)

      Not a polished product to wow the boss, but it works. well.

      http://www.fourmilab.ch/
      apt-get install speak-freely
      • I support speak freely adamently.
        It allowed my wife(in Brasil) and I(in Washington State) to "speak freely" on pun intended.
        And let me tell you it beats the hell out of $0.35/min long distance to Brasil.
        It even had a nice little conference calling ability that was pretty cool.

    • The version of MSN Messenger that ships with XP talks SIP (I'm not sure if the downloadable version of Messenger for earlier operating system does support SIP, I guess not because XP has SIP built in to the operating system).

      Microsoft seems to be taking the view that SIP is the way to go and is down playing H.323.

      Messenger is preconfigured to talk to several different ITSPs (internet telephony service providers) that provide worldwide PC to phone services. I know of one of these CallServe [callserve.com] that have some information on their site.

      IP telephony may not be sexy any more but it is still building rapidly in usage. A lot of "cheap" international phone to phone calls now use IP without the users necessarily being aware of the fact.

      • Microsoft seems to be taking the view that SIP is the way to go and is down playing H.323.

        Ick. H.323 is a dog to operate through NAT. If both parties are using NAT, you have problems getting one side to call the other. I've been able to call out from behind a NAT router to a modem user, but not accept calls coming the other way. It looks like SIP also needs a proxy of some sort.

        For a useful comparison check out this H323 vs SIP [packetizer.com]comparison. Looks like SIP is a lot simpler but less interoperable with things like PSTN.

        Really, these days there's no excuse for protocols that hide IP information in the packet data (that's FTP, H.323, and a ton of others).

        Jon

        • Thanks for the comparison link [packetizer.com]; it is a very thorough analysis, although it feels like the authors are biased towards H.323. They are definately more telephony minded rather than Internet minded, being worried about centralized control and billing for example.

          Since I work for a company that has products using both protocols, any bias I have is hopefully personal rather than commercial. Here are the differences between the protocol that really matter to me:

          • H.323 is definately telecom oriented, being a product of the ITU. If you just want to get a phone call across the Internet, this is the protocol to use.
          • The packetizer.com comparison mentions that H.323 is built on the ASN.1 notation, but it fails to mention that there are multiple options for encoding. Most ASN.1 protocols, such as SNMP, use the Basic Encoding Rules (BER), which are relatively easy to understand and implement. However, H.323 is the only protocol family I know of that uses Packed Encoding Rules (PER). The PER spec is HUGE and hard to understand. As of a couple of years ago, just about all H.323 apps were built on three or four commercial toolkits which were quite expensive because there were very few people who wanted to devote a year of their life to figure out PER. Now that OpenH.323 is out, I suppose the situation is better, but this used to be a huge barrier against getting into the H.323 space.
          • SIP is more Internet oriented, and its design takes advantage of capabilities on the Interent that you don't have in a telephony network. On the Internet, if you want to send queries to a dozen different servers to see which one I'm connected to, its not that big of a deal, but the equivalent search in a telephony network would tie up a lot of resources. Therefore, telephony data tends to be centralized, but Internet data can easily be distributed. This differences shows up in that H.323 favors a central Registration/Authentication Service (RAS), where SIP offers a distributed search mechanism.
          • Since SIP is a text base protocol, a Perl hacker can experiment with SIP services without a toolkit.
          Because of these issues, I think SIP is a better protocol for next generation Internet multimedia. However, SIP is immature, and there are some compatibility problems, and I suspect there are some basic design problems that may continue to plague it. I wonder if we won't see people move onto something that is SOAP based before this is all over. (Of course, now we have to get into the whole SOAP-RPC verses REST debate!)
    • Take a look at hppt://www.versiontracker.com there are several video-conferencing/netphone applications for Mac OS X. However, I don't know what they are based on as I don't have any of them myself.
    • 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!
    • H.323 nearly killed this whole application space. It's overly complex and typical of an offspring from ITU-T. Think of it as the "Telco guys" solution to all this.

      SIP is from IETF, mixes much better with typical Internet scenarios (NAT, firewalls...etc). It's also far easier to code to.

      SIP is the future, it is what is enabling the VOIP dialtone provider boom (it really is a boom, it's just hard to tell).

      For example, right now, in less than 10mins, you can go to www.denwa.com, give them your credit card info, and get a SIP dialtone. That includes a DID number (phone number basically, they offer several area codes), optional voice mail, and pretty much any call feature you can think of (as if you owned your own PBX).

      You can use a software based SIP device to make/receive calls, or you can use a VOIP/SIP enabled phone (like the Cisco 7960 you see in Fox's 24). You can also buy a FXS/FXO device and enable any POTS (Plain Old Telephone System) phone (generally not worth the cost).

      Anyone who pays for a Cable Modem and pays the telco for phone lines should consider dropping the telco completly, and instead signing up for a SIP Dialtone. It really is that easy.

      -Malakai
      • For example, right now, in less than 10mins, you can go to www.denwa.com, give them your credit card info, and get a SIP dialtone.

        In right about 10 minutes, you can go to www.denwa.com and become more frustrated than you've been in a long time.

        The site is horrible. It takes forever to figure out how to get anything out of them, and I never did manage to find out the rates for their service. I am surprised myself at how thoroughly the site pissed me off with its inscrutability - and I make a living at dealing with bad info design, so I'd expect to be inured to it. If they expect to deal with average consumers, and that's the best they can do, they're screwed from the start, no matter how fabulous the technology may or may not be.

  • VoIP Development (Score:4, Interesting)

    by saveth ( 416302 ) <cww@noSpAm.denterprises.org> on Wednesday May 15, 2002 @01:28AM (#3522148)
    <plug type="shameless">

    I work for a company [signalogic.com] 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.

    </plug>

    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. :)
    • I think :

      * Support for Visual Studio development tools: voice and audio software components on host side may be Visual C/C++ or Visual Basic DLLs or OCXs

      tells you all you need to know
    • Apparently, the software is so new that it is not even featured on the front page of our web site.

      Erm, I hope you didn't sign anything in particular before you found out about this product :-D

  • Vovida.org (Score:5, Informative)

    by Hatter ( 3985 ) on Wednesday May 15, 2002 @01:36AM (#3522171)
    Why wasn't a link to the project's actual webpage in the submission? Here [vovida.org] it is.

  • VOCAL (Score:2, Funny)

    by agent0range_ ( 472103 )
    Now, since they went out of their way to have a nice acronym, couldn't they have gone all the way? Change 'Vovida' to 'Vocal' and you have yourself a nice recursive acronym.
  • I used to use SIP at work for a SIP phone program for linux called Siphon. Let me tell you, it was a pain in the ass. First of all Siphon needed sip-0.1.7 but 0.1.7 didn't work for RedHat 7.1, but Siphon didn't work with sip-1.2.0, which was the only one that worked with RedHat 7.1. SIP is part of VOCAL and what I'm trying to say is that VoIP is kind of a mess on Linux, and I'm glad to see it improve.

    I pray for the day when Vovida comes out with better documentation, and perhaps a less memory intensive VoIP package.

  • Replace the Phone Company!!!! (Score:5, Interesting)

    by efficacymanUM ( 540328 ) on Wednesday May 15, 2002 @02:33AM (#3522289)
    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!
    • Who do you think operates the lines that let you connect via the Internet to other computers? The phone company of course.
    • I wrote todd (http://todd.cx [todd.cx]), so speaking from how mine is written, it wouldn't be too hard at all to set it up to have a system where you could just have <area code>.todd.cx point to a bunch of computers in that area code (though you'd have to hope they were telling the truth when they typed in their area code).

      > Anything to spite qwest!
      Qwest runs the main internet backbones you'd have to use for east-coast to west-coast calls :-) They thought of that already.

      Also, it'd be kind of weird to have the possibility of people picking up the phone in their house only to hear two parties they don't know using their phone - 'daddy, there are people talking about bad things on the phone')

    • Re: (Score:3, Informative)

      by account_deleted ( 4530225 )
      Comment removed based on user account deletion
      • But, if you want something similarly cool, check out Vonage [vonage.com]. $39 a month for unlimited long distance, you choose your area code, and it routes all of your calls over your broadband connection. Someone I work with has had it for a month, and it works flawlessly.

        My neighbor has Vonage, and it's okay. It's on a 384K Speakeasy DSL line with a Netopia R7200.

        The main problem is that any other network traffic just kills the phone call - people on both sides suddenly sound like robots and there's lots of dropouts. Just calling up a smallish web page (yahoo.com) is enough to do it for several seconds. So it's sort of a one-or-the-other proposition: Use the computer OR use the phone. Maybe that works in a one-person household, but not otherwise.

        I assume that with a higher-bandwidth connection this would be less of a problem, but having more than 384K upstream is not that common in consumer-world.

        I don't think the problem is with Speakeasy, as they get rock-solid 20ms pings to Vonage.

  • ...while i do not want to sound whiny i do want to say that all i personaly want is just a small app that will allow me to easily connect with my friend in the states and talk (i'm in greece) ...for example when i use ohphone , and i tell it to DISABLE some codecs, it just goes on and uses them. also, it just doesn't respect my preferences and it insists on using a 64kbit codec and all i have is a poor 56k line (more like a decent 33.6k line).

    anyone with some help appreciated. don't tell me to read the man page or the help page, i've read them so many times that it isn't fun anymore.
  • 'Vovida Open Communications Applications Library' ('VOCAL')

    Maybe they started the project only because they had found a cool acronym!...:)

  • At work I've got a cyclades PR4000 hooked up to a E1 (30 line pri isdn link) and the device is a router. The device can take an ISDN call and hand the data off to a port on anything connected to the net but it can't (or won't) do it if its a voice call because it hands it off to an overpriced DPS module that wants to decode the signals as if they were from a modem. With the
    exception of a packet saying "this call came in on port 12 for phone no 99991111 from 1233212232" its got all the bits together to pull this off but no such luck. 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.

    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.

    I also have another toy which is a 3com NBX 100 "IP Phone System". Too bad its an ethernet phone system and not an IP phone system. They claim its "open" but the only thing I've found out about it is they have illegal included gzip and gnu tar in an executable which they aren't providing source for. This from one of the few IT compaines that supported the DMCA. Maybe they had stuff to hide like stealing software. Google for "NBX rant" for more...

    </rant off>

    So I've got this cool device hooked to the phone co and I've got another cool device that hooks to cool phones that sit on my desk and talk over the lan. Will they every talk to each other? I think not.

    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.
    • Mabye you have a differnt NBX but the ones I've used are IP-capable. You can take your phone home with you, hook it up to your DSL. THe phone still thinks you are on your regular extension at work.

      Of course you do have to tell the NBX that this is happening but many people have done it. A quick google serach will turn up plenty of results.

      ISDN over IP? Um, I'm pretty sure the reason ISDN is used for commercial grade vtc is because you get a circuit, unlike IP and the Internet and its crazy packet switching. When I call your polycom box on ISDN it's a lot more stable and less flaky than IP. Eventually IP routing and bandwidth on the Internet might be so great that packet switching is as predictable and reliable as circuit switching, but that is FAR from the case now.
      • The NBX 100 requires an upgrade that cost someware around AU$5000 to turn on IP.

        My local ATM loop is too busy to support the phone over ADSL so thats out.

        My internal ehternet and VPN's all have less jitter than the ATM loop so it seems to me that ISDN over IP would work fine. Infact I've done it using the NBX and relaying over PPP over an SSH tunnel on controlled lines and it works fine. ISDN is 64k data. There is no reason that a typical T1 with QoS can't cope with a few channles of ISDN over IP without anyone noticing but this won't work to call 1/2 around the world but I don't need a solution for that since phone lines work great for that and wholesale rates between the US and Oz are now under US$.015/min its cheaper to pick up the phone than send the data.
    • 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.
      • Call setup:
        1) RAS gets voice call on a specifc channel
        2) RAS opens tcp port to a ivr server
        and connects the data stream to that.
        3) IVR server sends recored mulaw data down the line and the caller hears that a audio
        4) caller sends touchtones down the line to the program on the server which does an FFT on the data to figure out what buttons where pressed.
        5) RAS sends call info (like caller id) to a syslogd somewhere.
        Call teardown:
        Someone hangs up the connection. Other end gets
        hung up too.

        How about dial out?
        1) server decides it wants to make a call
        2) connects to port 20032 on RAS and sends "atdt1234535645"
        3) RAS makes outgoing call
        4) IVR server sends recored mulaw data down the line and the caller hears that a audio
        5) caller sends touchtones down the line to the program on the server which does an FFT on the data to figure out what buttons where pressed.

        Seeing that the device already does 99% of this (100% if the data bit is set on the voice call) and IT WORKS, I don't see why I need all the other nonesense that the protocols give me.

  • How will this newcomer work with asterisk? (Score:5, Interesting)

    by Lumpy ( 12016 ) on Wednesday May 15, 2002 @05:35AM (#3522555) Homepage
    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?
    • PhoneJACK cards [linuxjack.com] are not expensive at all if you consider the fact that they are the ONLY product out there for the low-density VoIP market that has on-card, no license, royalty-free compression codecs. The "royalty" is built intot he hardware cost. There are NO OTHER OPTIONS at this point, if you want to use decent compression with open source code. GSM, despite being available, does have patent issues, and uncompressed audio will not fly on the open net.

      If you want to do VoIP for $20 a port, you will not get the compression - the license fees are higher than that, even before you pay for hardware. The VoIP-Blaster is a nice toy, but it cannot do compression and it cannot ring the phone - so it's basically an outgoing phone interface unit that does not support compression.

      I've played in this space for almost 4 years now. It's not changed much. There is still a lack of good low-density hardware at decent prices with good cross-platfrm support; the biggest reason for this is volume. If the demand were there for 100,000 cards, you could probably get a decent break on volume pricing. To date the demand has never grown sufficiently so I predict the prices will not drop radically.

      And as for PSTN interfacing, you will always have the cost of the real termination point to factor into any system. Even if you can do this cheaply, to match what the phone system does now you will not lower the cost per minute all that much - you can easily get 4-5cents/minute rates in the US, which is a lot cheaper than VoIP if you factor the costs of the new equipment in.

      The place where VoIP makes the most sense is still the third-world. You face internet access issues there, but at 35-50 cents/minute long distance rates in the best cases, you have lots of margin to save serious cash by choosing VoIP, even after paying for the new hardware.

      VoIP is quite interesting, but linux based VoIP and telephony in general has declined in the last few years - mostly due to a lack of new hardware support. That is due to the fact that the US market - the major market for hardware - is still so cheap for real telephony that there is no market driver. At least in my opinion.

      • Funny... the VOIP blaster rings the telephones connected to them fine. and in FACT I have used 1 Voip blaster to create a POTS gateway. I simply scabbed one on a POST line interface to the NEC phone system here and programmed the phone system to auto-pickup and connect into the phone system with an internal dial tone..

        So I dial the Voip blaster on the phone system, get another dial tone, dial 9,1-800-I81-U812 and voila I have a nice connection outside.

        Yup, they work absolutely fantastic for a toy. and do everything we needed here. Oh, the Voip Blaster does have hardware compression and echo-cancelling.. fobbit only does a stream to stream connect. kinda neat that way...

        you really should try and get a couple of VoIP blasters, they coupled with fobbit or fobbit+openH232 make a very viable VOIP solution that is invisible and completely operated from the $35.00 analog telephone plugged into it. (and yes when you call it it rings... no difference than a normal phone operation outside dialing extension numbers only or ip addresses.
        • you really should try and get a couple of VoIP blasters, they coupled with fobbit or fobbit+openH232 make a very viable VOIP solution that is invisible and completely operated from the $35.00 analog telephone plugged into it.

          I thought VoIP Blaster had been discontinued by the manufacturer. Have they changed their minds?
          • Yes it has been discontinued by Creative, the manufacturer is still happily making them. Inno Media still sells them like hotcakes outside the USA. Besides, I am getting mine from clearing houses, ebay (when you find sane prices... ebay is getting insane, but that's another topic) and I personally cleared out creative when the story first ran here on slashdot. My first test was to add "free" phone service to remote Z ends on my T-1's and fiber backbone, it works perfectly, even cisco technicians and MCSE's can use it! (sorry about the cisco dig, but recently having to have to explain TCP/IP routing to one has jaded me) I have further expanded it to include my relatives on the internet with my personal servers acting as their phonebook server. The VoIP blaster uses on average 18-30Kbps of bandwidth with latency of 250ms going un-noticed unless you are actually looking for it... so the dial-up relatives can use it too. Basically the Voip blaster is the absolute best Viop product available to date that uses a computer, Granted, you can spend $$$$ on a real voip phone but then I lose the flexibility of my current VoIP network.. (I.E. I control my phonebook server and I can run peer to peer, regular VoIP phones need more infrastructure to work properly, and need buckes of money.)

            I personally think that creative dropped it because of either telco pressure or they were afraid of getting sued.

  • Competition for Nortel / Lucent (Score:4, Interesting)

    by z_gringo ( 452163 ) <z_gringo@NosPam.hotmail.com> on Wednesday May 15, 2002 @05:50AM (#3522573)
    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.
  • Open Source: Data, Analog voice, VOIP, PBX... here. [asteriskpbx.org]

  • SIP tools (Score:3, Informative)

    by leonia ( 246522 ) on Wednesday May 15, 2002 @08:06AM (#3522876) Homepage
    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.
  • I have no more than a most basic understanding of how trademark law works, however, since this is a computer telephony-type project, they might want to think about picking a new name before they get sued. This two-bit company I worked for a couple years back registered a very similar word as their trademark for a digital recording system: VOCALS [allianceresearch.com].

    Note that I make no judgement on the hows and whys of the way that whole process works, but I've seen enough news stories of this sort of thing that they might want to consider coming up with something else. If they were wildly and completely different, I wouldn't even bring it up, but one is VoIP and the other is a digital voice-recording system that ties into the phone system on one end and a database & data-collection application on the other-end...

  • While most Open Source projects are applications and utilities intended for single users

    What is this AC thinking? What is timothy thinking posting this without editing it?

    Gee, just off of the top of my head: samba, bind, apache, open ssh, sendmail, mysql, and postgresql.

    None of those are single user applications and I think slashdot.org is using a couple of them.

  • The article seems to show a fairly simple state model for Vovida.

    I love simplicity, but worry about feature completeness and extensibility, too.

    Does anyone knowledgable know how Vovida compare with Bayonne [gnu.org]?

  • I live in a small town in Illnois where Verizon has graciously decided to charge us a connect fee for Every call... even if it never gets past our local loop... SO, What about taking this technology to build "In-town" phone services, connecting out-of-town calls for the expected rate?

    Is this something that could be explored?
  • Comment removed based on user account deletion

  • VOCAL Posting (Score:2, Interesting)

    by vocal_scribe ( 579597 )
    I would like to clarify a few points about the recent Vovida posting:

    VOCAL was created by a group of about 50 developers who worked for Vovida Networks, not just David Bryan and David Kelly. David and David wrote the article that appeared in Embedded Linux Journal and that's why they're mentioned in the posting. All in all, VOCAL represents over 100 man months of development.

    Vovida.org is a community web site that hosts many open source communication projects and protocol stacks including VOCAL. There are a number of other open source VoIP solutions, most of which do different things than VOCAL. Bayonne (bayonne.sourceforge.net) and Asterisk (www.asteriskpbx.com) both support TDM interfaces, and added (or are in the process of adding) VoIP support, while VOCAL started VoIP-only (and SIP-centric, at that). OpenH323 (www.openh323.org) focuses on H.323, although the latest versions of the code also has SIP support. Probably the most similar stuff to VOCAL is the osip stack (www.fsf.org/software/osip/), the related proxies, and linphone (www.linphone.org). We have done some work to make linphone interoperate with VOCAL.

    As for the Slashdot community's comments about the article's opening line, "While most Open Source projects are applications and utilities intended for single users,..." we didn't intend to slam apache, sendmail, bind, mysql and other multiuser projects. We intended to show that VOCAL was interesting because it was built from the ground up as a distributed system that can easily load balance across multiple servers to scale.

    We've been trying to make VOCAL easier to install for new users. If you tried earlier versions of VOCAL and found it difficult to install, you might want to try the latest version. We have also built RPMs and Solaris packages so that people can try VOCAL without having to compile the source code. For those who are interested in acquiring the source code, it is available in tarballs and from CVS.

    As for the documentation, we have been working on a book for O'Reilly that includes not only user guide material but large amount of detailed information about the data structures and state machines. The book is called Practical VoIP: Using VOCAL and it is due out this summer. People who have seen advance copies have told that, from a developers' point of view, the material is very useful. Thank you for your patience and please stay tuned.

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