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Cross-Platform VoIP Software? 205

feilkin writes "With the release of Skype's Linux client, I'm wondering about alternatives. Namely, cross-platform solutions for voice communication. I've got friends who are using Windows, Linux and OSX, and I'm hoping that there is a way to communicate with all of them. I myself am using Linux, and I haven't been able to find any solutions that seem fitting to my situation completely. Does anyone have a solution that'll be useful on all three platforms, or solutions that may be coming in the near future?"
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Cross-Platform VoIP Software?

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  • SIP (Score:5, Informative)

    by mcrbids ( 148650 ) on Monday June 21, 2004 @08:28PM (#9490224) Journal
    • Re:SIP (Score:5, Informative)

      by Anonymous Coward on Monday June 21, 2004 @09:14PM (#9490506)
      for those who don't know what SIP is.

      A number of years ago, the telecom providers got together and tried to do VoIP. They came up with H.323, which was a terrible mess and near impossible to do anything with. To top that off, you have to pay for access to the spec (I'm only pretty sure about this, please correct me if I'm wrong) So VoIP didn't go anywhere for a while.

      Then the IP folks (the people who designed the internet protocols like IP, TCP, UDP, etc) came together and designed SIP. The entire protocol is described in a mere 150 page RFC []. Anyone who's implemented a standardized protocol from a spec knows what a godsend a short spec is.

      In short, SIP is a protocol designed by the Internet folks for the Internet. It's layered on RTP [], so the audio quality degrades gracefully with the link quality. You can operate it point-to-point by simply running two clients on two machines and pointing one at the other's IP address. Or, if you want an easy to remember URL, you can sign up for a free account at places like You'll then be accessible as

      Google for "SIP softphones" and you'll find quit a few clients. The big ones on linux are kphone [] and linphone []. Shtoom [] is making some headway also, and runs on linux, windows, and os x.

      Skype decided they don't like either H.323 or SIP, went off and designed their own proprietary protocol, and is keeping it secret from everyone else.
      • Re:SIP (Score:4, Informative)

        by pacman on prozac ( 448607 ) on Monday June 21, 2004 @09:59PM (#9490835)
        Disclaimer: I just finished studying this and haven't implemented any of it so could be miles out.

        The actual phone conversations run over RTP whether you're using SIP or H.323. It's how the call is set up that differs.

        SIP uses set-up mechanism that works over HTTP, arranging the caller, receiver and codecs etc. Because of this it is simpler than H.323, which uses loads of other protocols that run over a mix of UDP and TCP, such as H.225 (call setup/RAS), H.245 (call management) and a few more I should probably know but have forgotten. H.323 needs a separate "gatekeeper" to control connections whereas a SIP client can use the DNS to find it's destination, addresses in SIP look similar to email addresses but may include a port number aswell.
        • Re:SIP (Score:3, Informative)

          by muonzoo ( 106581 )

          SIP uses set-up mechanism that works over HTTP


          SIP (aka RFC3261 [] et al.) uses SIP to setup calls. The syntax of SIP is clearly inspired by HTTP, but HTTP it ain't.

          Location of SIP services is handled through DNS operations as described in RFC 3263 -- Locating SIP Servers [].

          Why, oh why we don't locate HTTP services using SRV or NAPTR records is really a sad question -- virtual hosting would work so much better.

          Most everything else you mention is fairly accurate. There are excellent SIP resour

          • I reckon the lecturor said based on HTTP not run over HTTP, damnit, I'm still waiting for the final exam results too :/

            SIP seems very nice and simple, I'm sure it wouldn't be too hard to add a "nat: <natport>" extension to it that could make the client send appropriate packets out to convince the NAT router that packets should be allowed back in on that particular port.

            The entire NAT problem is overrated imo, its not like we don't have the processing power to deal with a little address & port re
          • by bblackfrog ( 513266 ) on Monday June 21, 2004 @11:43PM (#9491658)
            The NAT problem is definitely NOT overrated for residential gateways. There is no SBC in this case.

            Sure, SIP can use TCP as a transport, so a client can punch a signaling connection out through a residential NAT/gateway, but a SIP client still needs to advertise an address and port to receive RTP on. He can't very well advertise his NATed address if he's connecting to a client outside of his NAT.

            STUN - RFC 3489 - ( partially addressed this problem, by allowing an application to discover its public address and port mappings.

            TURN ( erg-midcom-turn-04.txt) addressed the problem, but in an inelegant manner, by routing all RTP through a public server.

            ICE ( usic-ice-01.txt) describes a method of using STUN and/or TURN to discover, describe, and prioritize many potential addresses. Using ICE, two SIP clients can choose the best possible route for RTP, through several NATs that might separate them.

            And it's backwards compatible.

            It should also be noted that ICE is independent of SIP, and could also be applied by H.323 clients, or RTSP streaming client/server for that matter.
      • by __aadkms7016 ( 29860 ) on Monday June 21, 2004 @10:25PM (#9491052)
        The SIP RFC you linked to is obsoleted by RFC 3261 []
    • Re:SIP (Score:3, Interesting)

      by walt-sjc ( 145127 )
      I use SIP devices with Asterisk, FWD, etc. Sip has one major flaw - horrible support for NAT. If you are behind NAT, you frequently need to use an external proxy. Considering that SIP is not all that old, it boggles the mind that it handles NAT so poorly.
      • Re:SIP (Score:1, Interesting)

        by Anonymous Coward
        SIP can handle NAT situations just fine, you just need to ensure your endpoint as declared in the SIP packets is your real address, and the devices between do The Right Thing.

        * from 0.7.x onwards can force the address to be correct, and works fine for me behind a 1:1 NAT.
        • Re:SIP (Score:3, Informative)

          by walt-sjc ( 145127 )
          For more info on SIP and NAT issues, see this [] and this [] and other related pages at 1:1 nat is atypical. Most people use a much more simplistic nat (cisco calls it PAT) where you only have 1 real external IP address. I'm well aware of the tricks asterisk uses, but again, this is atypical of sip clients.
      • you're thinking of H323.

        SIP has specific support for NAT. H323 does not.
    • Better yet (Score:3, Informative)

      by swordboy ( 472941 )
      BudgeTone 101 - $75 []

      It is basically a phone with an ethernet port and SIP built in. Not bad.
      • Re:Better yet (Score:3, Informative)

        by Tony Hoyle ( 11698 )
        $75 is expensive for one of those...

        They're pretty horrible but a good intro to VOIP if you don't want to spend much. You could get an SPA2000 for a little more and be able to plug your DECT phone directly into it - wireless VOIP on a budget.
  • err... (Score:3, Insightful)

    by errgh ( 744846 ) on Monday June 21, 2004 @08:28PM (#9490227) Homepage
    • Re:err...Yes Skype (Score:5, Informative)

      by Tuna_Shooter ( 591794 ) on Monday June 21, 2004 @08:43PM (#9490329) Homepage Journal
      I've just tested the Linux version of Skype here -Local box to box - Linux to windows - 2 accounts as advertised. But just from a technical point of view being an old coder myself, i'd like to know how they minimize the lag.. Dam this thing works better than my cell from a lag-latency point of view.
      • Why would you think there'd be a huge lag working local-local connections?

        The problem with most voip is with network induced latencies. So a local-local test is less interesting as say a system - nearby network test.
        • The lag is minimal even for long-distance connections. I live near the west coast of Canada, and my girlfriend lives in north Florida. (oh, how I love the Internet 9_9) Judging by the rhythm and flow of our conversation, I would guess that the worst it got on a regular basis was a lag of less than one xecond. (of course, there'd be some severe peak latencies now and then, lasting three seconds or so.)

          I can understand how they reduced bandwidth-induced lag... their codec seems to be very well tuned for spee

          • Re:err...Yes Skype (Score:3, Interesting)

            by sumdumass ( 711423 )

            but once they reduce the number of bits, how on earth do they push those bits so damn fast across the continent?

            I'll take a stab at how.. This is just a guess but instead of doing a transmit and recive, they might open a stream like one from a streaming video server but smaller to only use enough bandwidth neccesary for low-medium quality voice. durring the non talking time they could just introduce sub audible noise to keep the conection alive and recive the same from the other end. with a stream like

            • Re:err...Yes Skype (Score:2, Interesting)

              by mindstrm ( 20013 )
              Half of streaming radio? Far, far less than that.

              The VoIP codecs I've commonly seen use about 6kbps. That's nothing by today's standards.. acceptable quality for voice is far, far below radio.

              You can make voip calls quite easily over a 28.8kbps modem dialup connection.

            • you're probably right. I imagine they use UDP to avoid the overhead of sending TCP packets... and actually, I think you're definitely on to something with the 'sub-audible noise' thing... in the periods of silence between conversation, I notice... well, it's not sub-audible, because I can hear it... just the tiniest bit of white noise.
    • Re:err... (Score:3, Informative)

      by MinutiaeMan ( 681498 )
      I believe that the original post specified cross-platform, including Macintosh. Skype does not work on the Mac.
      • Re:err... (Score:2, Informative)

        by ortcutt ( 711694 )
        Mac support is coming real soon if the reports a month ago were true.
  • by webperf ( 560195 ) on Monday June 21, 2004 @08:30PM (#9490242) it has clients on all platforms. the other one is called asterisk
  • FWD (Score:3, Informative)

    by Anonymous Coward on Monday June 21, 2004 @08:32PM (#9490260)

    I've got friends on it using windows and linux (I personally use both, and have clients installed on both). I'm pretty sure they've got osX clients aswell.
    • Re:FWD (Score:3, Informative)

      Yes there is a Mac OS X version of XTen X-Lite SIP soft phone ( I'm using it to call friends with windows machine and linux machine.. works great. For "commercial" SIP services there is also nikotel ( and they also are using XTen soft phone.
  • SIP solutions (Score:5, Informative)

    by p80 ( 771195 ) on Monday June 21, 2004 @08:33PM (#9490266) Homepage
    I know nikotel [] works great on windows and macOS and as it is SIP compliant it works with linphone [] and kphone [] on linux.
  • OpenH323 (Score:5, Informative)

    by klingens ( 147173 ) on Monday June 21, 2004 @08:33PM (#9490270)
    OpenH323 [] is available on all 3 platforms and has very good voice quality. It can do video as well. Setup is not always trivial: it needs lots of open ports, udp and tcp. The license is MPL.
    • Re:OpenH323 (Score:3, Interesting)

      by ciryon ( 218518 )
      OpenH323 is:
      • Microsoft Netmeeting on Windows (still included in XP, even if it's "hidden" inside Program Files)
      • Gnomemeeting [] on Linux
      • OphoneX [] for OS X.

      That's the bad solution. You need to open loads of ports and it is still tricky to get to work.

      I suggest using AIM/iChat on Windows/OS X since audio and video is the best in the business. Also works through firewalls without need to open ports. This solution isn't available for Linux afaik, but it might be in the future. The protocols are open.


      • I've tried to get iChat and AIM working together to do voice chat, but I've never succeeded. I've read that a few people have made it work, and many people have not. Does anybody know why, and how I can make it work? I'm very interested in this question.
  • Bayonne (Score:3, Interesting)

    by nacturation ( 646836 ) <nacturation@gm[ ].com ['ail' in gap]> on Monday June 21, 2004 @08:34PM (#9490273) Journal
    I haven't checked in on the project in a while, but Bayonne [] was coming along nicely in this area and is currently used in a few production facilities.

    You might have to roll your own, but the framework is certainly there.
  • by SteamyMobile ( 783822 ) <> on Monday June 21, 2004 @08:38PM (#9490296) Homepage
    SpeakFreely [] used to be a fairly good option. I tried it several years ago, and it did work ok so long as everyone was on broadband. The project has been abandoned, though, and no future releases are planned.

    At this point, all the tools needed to create an Open Source cross-platform VoIP system are easily available. The Speex [] codec is specifically designed for low-bit-rate voice, is BSD licensed, and is implemented in both C and Java. It would not be hard to take this codec, throw in some good sound libraries and some crypto libraries (OpenSSL perhaps) and roll up a VoIP client. In fact there is a Speex implementation for Java, so you could write one in Java, and yes, Java really is "write once run anywhere" these days. Someday when I have more time I might do this. As a Java applet it would be great because there would be nothing to install.

  • by Anonymous Coward on Monday June 21, 2004 @08:38PM (#9490297)
    Does anyone have a solution that'll be useful on all three platforms?

    Son, I have a solution! Pick up the phone and call your friends!
    • Actually, if you all get on one of the providers that do free mobile-to-mobile (I'm partial to Sprint), that'd require little work and probably not cost any more money than whatever phone service you've all got now. I have no land-line, which helps a lot.

      Or, just go with a standard-ish protocol that has clients on each platform, like SIP or H.323, and don't worry about getting the same program for all...
    • Thats not really a solution if you want to get rid of your phone bill.

      The best solution if you really want VoIP to be pratical is a hardware solution. Anything from d-links cheap 4 line VoIP gateway to the latest wifi VoIP phones. I'm not sure of any model numbers or even which brand name is best right now. We have alot of VoIP hardware in our new Telecom lab at Collage. I haven't had a chance to play with much of it yet.
  • by Anonymous Coward
    ... just install windows everywhere
  • SIP (Score:4, Informative)

    by pirodude ( 54707 ) on Monday June 21, 2004 @08:43PM (#9490328)
    You're looking for a standard protocol that can be used across all platforms, and that protocol is SIP. I've used several VOIP products that have SIP support and currently am using a Grandstream Budget Tone 100 VOIP phone ($65) to do my calling and can contact anyone on any platform that can support SIP.
  • Open Standards (Score:5, Informative)

    by Anonymous Coward on Monday June 21, 2004 @08:44PM (#9490334)
    There are two main standards in use for VOIP:

    SIP and h.323. There are lots of clients out there for both of them.

    There should be a checkbox next to the "ask slashdot" submission box that says "did you use Google first?"
  • by A.T. Hun ( 192737 ) on Monday June 21, 2004 @08:48PM (#9490356)
    UT2K4 has built in VoIP support and is available for Windows, Linux (32 and 64 bit), and Mac! "No, sir, we're not wasting company time, we're _collaborating_! We're enhancing shareholder value! It's a whole new paradigm!"
    • UT2K4 has built in VoIP support and is available for Windows, Linux (32 and 64 bit), and Mac! "No, sir, we're not wasting company time, we're _collaborating_! We're enhancing shareholder value! It's a whole new paradigm!"

      Sir, just 2 more frags and we'll close this contract.

    • kind of reminds me of all those network admin mods for doom were you did actual work but form a console that resembled the game doom and you actually shot object to activate and deactivate them. I think you could also shoot ports to close them off when there was srtrange activity on them.
      is one for system administration.. i'm not finding the one about network admin right now.
  • Vonage (Score:3, Insightful)

    by Gothmolly ( 148874 ) on Monday June 21, 2004 @08:53PM (#9490389)
    For 29.99 USD a month, Vonage [] is the way to go.
    No fscking around with codecs.
    No gcc bullshit.
    No patching, only to have an OS upgrade break your app.
    Pay for it, be done with it, move on. What's your life worth? Time = money.

    • I second Vonage. Vonage is one of the big reasons why so many DSL providers won't unbundle their DSL service from their local phone service. I'd much rather give Vonage $30/month for a pretty incredible VoIP solution over my DSL than buy Bell's gimpy bundled voice line that I'm forced into.
    • Huh. for $30 a month i hope it's worth it. I can not check for myself, since the page tells me i've got no flash (And no, I don't want it either:)
    • No kidding. Plug it in and rock on. Let's be productive now and then.

      The OS X client is either out now or imminent, too.

    • Re:Vonage (Score:2, Funny)

      by Anonymous Coward
      "No fscking around with codecs."

      Usually, codecs doesn't involve file system checking, I think you hard drive is fucked, you should consider reinstalling on a new one.
    • Re:Vonage (Score:4, Insightful)

      by dj245 ( 732906 ) on Monday June 21, 2004 @11:09PM (#9491424) Homepage
      Unfortunately vonage requires lots of open ports. Many of these ports are now being camped out by worms and trojans, and are blocked by some ISPs (mine included). Vonage is completely useless on many ISPs due to port blocking of these worm ports.
      • No problems here... (Score:3, Interesting)

        by artemis67 ( 93453 )
        I use Road Runner through Time-Warner Cable, and have no issues with port-blocking. Also, the Vonage VoIP-box goes outside of your router/firewall, so you won't compromise your own network, either.
    • Just get the $15/mo. package. 500 anywhere minutes, and all calls to other Vonage subscribers are free.

      I'm using Vonage, and I love it. The sound quality is not pindrop-clear, but it's good enough for general, everyday phone use. I also like that Vonage emails me all of my voicemail messages, so I know immediately who called my home while I'm at work.

    • For 0 dollars per month, Skype is the way to go, if your friends have computers.

      We need an open source Skype. Sooner or later Skype will start charging. For what? Just some programming, that could be done better if it were Open Source.
    • I just started using Vonage. It has it's advantages, especially if you want a land-line with an external phone number, something that Skype and the like cannot do. With Vonage I'm able to disband with my local and long distance phone companies, and pay a single $30 bill for unlimited calling. You can also get phone numbers from different area codes which is nice if you want your families to be able to call you without paying long distance bills.
  • TeamSpeak (Score:4, Insightful)

    by Apreche ( 239272 ) on Monday June 21, 2004 @08:54PM (#9490393) Homepage Journal
    Sure, you can't talk to people who only have POTS.

    Sure, its designed to be used with online games.

    But really, its just IRC with voice. I talk to people in Australia with it all the time, for free. You can turn the bitrate down and even talk to people with dial-up. I usually just use gaim to tell someone "hey, join the teamspeak server, let's talk".

    Also, this way I don't have to remember phone numbers.
  • While reasearching this to play online game with buddies, I found that ventrilo and teamspeak were the most popular. Ventrilo has clients for many platforms (Win32, OSX, Linux, BSD, Solaris) but only a client for Win32. OSX and Linux clients are in development. Teamspeak seems to have only Win32 and Linux client and servers.

    My friends and I couldn't get Teamspeak to work, while ventrilo worked straight from install (on Win32 client & server or linux client).

    To bad neither are Free Software/Open So
  • Ventrilo (Score:2, Informative)

    by trawg ( 308495 )

    Has Win32, Mac and Linux clients.

    It is client/server, so you'll need a server, but you can get 8 users (I think) on the regular server. It is relatively bandwidth-friendly and awesome quality.

    Probably a bit harder for computer illiterates to use but its very cool software.
    • Re:Ventrilo (Score:3, Informative)

      I've used the Windows version and it seems to work really well. We had 15-20 people on a server to help with organizing parties in Star Wars: Galaxies.

      Unfortunately, the don't have Mac or Linux clients. Or, at least they're not available. Both are listed as "In Development" on their download page.

    • Nope. No linux ventrilo.
  • h323, sip (Score:2, Informative)

    by w4rl5ck ( 531459 )
    as many others pointed out, natural joices would be h323 (very wide-spread) and sip.

    I don't know much about sip, but everyone tells me "stop using h323, use sip". Seems to be better, but never change a running system.

    h323 is only for VoIP, not for calling real phones - unless you have an gateway to the "real" world.

    There are many h323 programs available, like netmeeting (really hardcore connectivity problems through firewalls, better use...), openphone (openh323/windows), gnomemeeting(openh323/linux) and
  • It works across all platforms, including all future OSes, and it is cheap to buy and operate.

    If you want mobile, get a cell phone.
  • complete solution! (Score:4, Informative)

    by shokk ( 187512 ) <> on Monday June 21, 2004 @09:22PM (#9490549) Homepage Journal
    For a complete VoIP Linux solution, check out Asterisk [].
  • H323 is a huge ball of protocols (H245/Q931/etc) for communications technologies. See the OpenH323 project [] for more. You can use it for connecting, say, VoIP and a regular phone using a Cisco PSTN gateway. The two main reasons for using something like the H323 protocol set are for session initiation (can you do video? What audio codecs do you support?) and data format (G721/722/726/etc).

    NetMeeting and GnomeMeeting both support some subset of H323 that lets them talk. So you can do audio/video between plat
  • h.323 for all (Score:2, Informative)

    by marcjw ( 546823 )
    All of these will interoperate. They get tricky when used behind NAT. The best option I have found in that case is to use a gatekeeper.

    Gnomemeeting for Linux []

    OphoneX for OS X []

    Netmeeting for Windows []
  • Asterisk (Score:5, Informative)

    by Michael Spencer Jr. ( 39538 ) * <spam @ m s> on Monday June 21, 2004 @09:31PM (#9490606) Homepage

    This thing is a VoIP BEAST. It's an open source PBX which runs on Linux. This will solve your problems by connecting all of these incompatible VoIP clients, making them all seem like virtual telephones, each with their own extensions. (This is good, if you don't mind them using your bandwidth when they bounce off of your Asterisk server to communicate with each-other.)

    "PBX" seems scary -- it's the same kind of system large businesses use to manage tons of phone lines, both inside their company and connecting to the outside world.

    For the needs of people like you and I, don't think of it in terms of "a solution used by people with lots of phones" -- think of it in terms of the kinds of technology it uses and can connect with.

    "Physical layer" stuff: with dedicated hardware it can talk to existing phones and existing phone lines. There's even a PCI card that can communicate with four T1 lines, for nearly 100 phone lines out to the telephone company. It can also do VoIP using standard interfaces like SIP, using its own unique (but open-source, not proprietary at all) interface called IAX, with existing programs like Netmeeting or MS Messenger, or with any number of Linux programs. (There's even an IAX client for my Zaurus PDA. That's not all that practical for receiving calls, but I have successfully placed phone calls with that client, over 802.11b.)

    Logical stuff: each of these connections to the outside world is given a context, and you can do things with those contexts. A connection to your outside phone line will be used by unknown callers, so its context shouldn't have access to features that cost money. A connection to an inside phone is "trusted", so it should be given access to these features.

    The system has something like a "dialplan", which is a rather flexible set of scripts you use to handle calls. There's a lot of room for creativity here -- you can make your system do anything you want with any call.

    This is so flexible because you form your dialplan from a bunch of references to "applications", either built-in or external. Some are very simple: play this wav file, transfer to this extension, go to this voicemail box, etc; some are more complex, such as "shell out to this executable CGI-style and do whatever that executable tells you".

    Asterisk also comes with a bunch of audio samples recorded by a "professional PBX voice", and many of them are saying some rather funny things, only useful for a home user. "All representatives of the household are currently assisting other telemarketers. Please hold, and you call will be answered in the order it was received."

    Asterisk can email you your voicemail messages as wav files. This is a KILLER feature. But you weren't asking about voicemail, you were asking about VoIP.

    Pros: VoIP BEAST. Take all your friends with VoIP clients, give them signins and extensions and voicemail, give them conference capabilities, etc. (Then they all use your bandwidth.)

    Cons: Complexity. Even if all you want is a simple call routing tool to make incompatible VoIP systems talk to each other, you have to learn the entire system to make it work. This is a typical Linux problem: you have to read tons of documentation / visit forums / discuss with others to figure it out, but because it uses "real world" concepts and is designed intelligently, once you're finished you have spent 30% of your time learning the quirks of a single software package you could care less about, and 70% of your time learning about how the subject works, gaining knowledge about that field that will follow you to any other program.

    (That's definitely true here: Since playing with Asterisk I've talked with professional telecom guys, and found what few terms and concepts I've learned from Asterisk definitely overlap with their "real world" stuff.)

    Weird system service requirements. Some software features rely on a very high-resolution system timer, and (allegedly) can't get t
  • Its this really cool new protocol that lets you communicate and its supported by all major platforms. So far Microsoft has not embraced and perverted it so its cross platform future looks good. It great because it usually gets delivered right away but your friends don't expect a response within thirty seconds. You can send to an two way pager or a cell phone. You can even claim that you haven't been checking your email for a while if you don't feel like answering right now. And you can send HTML email that
    • Re:SMTP (Score:3, Insightful)

      by ZigMonty ( 524212 )
      Uh huh... now explain to me how the "voice" part of the requirements is met, preferably in the same condescending tone.
      • I recored my voice to an ogg encoded binary and send my digital communique using the simplistic mail transference protocol. It is called a simplex conversation because only one person can talk simultaneaosly. My voice passes over the internet using the "internet protocol" a very sophisticated method for many inter connected computers to send messages amongst themselfs. If this sounds like it is going over your head, I would be happy to explain it in laymans terms. Don't be afraid to ask for help if you need
  • shtoom (Score:5, Informative)

    by anthony_baxter ( 48233 ) on Monday June 21, 2004 @09:40PM (#9490660)
    Shameless self-promotion - check out the shtoom program. It's cross platform (although the Mac support is incomplete, it in theory works, thanks to portaudio[1]), it has user interfaces for command line, Tk, Gtk, Qt, and wxWindows. Audio support is via PortAudio and OSS. It handles most NATs correctly (using STUN).

    It also includes 'doug', an application server for writing voice apps. There's a simple voicemail and simple conference server implemented in doug.

    It's pretty rough - it's certainly not something you'd give to your mother to use, but hey, it's free software.

    It's also entirely in Python.

    At the moment, the best bet is to use the svn trunk.



    PyCon paper (also possibly useful for an overview of VoIP): on2004/

    [1] Native Mac support will be finished Soon, I have a mac being shipped to me.
    • Re:shtoom (Score:2, Informative)

      by Packets ( 8071 )
      Shtoom is the only voip client that *just works*

      I installed the libraries, installed the client, ran it, typed in a sip address, hit 'call' and *volla* I was talking to some dude from alabama (I'm in .au).

      Cred to anthony for writing it.
    • ``It handles most NATs correctly (using STUN).''

      I always thought the most obvious solution to connecting to a NATed node is to have the NAT box act as a relay to the nodes it performs NAT for. One way to implement this (without running out of address/port pairs) is to use IP-IP tunneling. This is described, for example, in RFC 2003 []).

      Nodes can then negotiate (or, with an extension to DNS, look up) the parameters to use to traverse all the NAT boxen in between them. This schema doesn't require traffic to go
  • re: software is lame (Score:2, Interesting)

    by Rage Maxis ( 24353 )
    don't fuck around with a stupid software phone.

    just get a grandstream budgetel or a sipura or a wisip or any number of other SIP hardware phones.

    You will be happy you did. I am.

    plus wearing stupid headsets looks retarded.
    • by Elentar ( 168685 )
      Walking around with your head crunched over onto your shoulder to squeeze the phone there, getting an ache in your ear from flattening it, and having to hold a cellular/cordless phone close to your head for hours-long conference calls all look very retarded to me. Having a quality headset (like those from Plantronics) and being able to use two hands, walk upright and avoid ear/headaches looks very smart.

      Of course, if you meant that wearing a stupid-looking headset looks retarded, you're absolutely right -
  • [] is pretty comprehensive. Thier Vovida Open Communications Application Library (VOCAL) is pretty comprehensive, and works with many different vendor's phones, soft phones, and even Cisco's high-capacity PSTN gateways (H.323, MGCP, and SIP).
  • Teamspeak2 (Score:2, Informative)

    by Bob Loblaw ( 545027 )
    I am a fan of teamspeak2 ... it it not open source but it is free, cross platform and works quite well for conference/gaming type communication.
  • On June 16, 2004, there was an internal demonstration at Skype of the alpha version of Skype for Mac.The alpha version worked well and the development team is working towards a beta launch of Skype for Mac in about 2-3 months. The other major OS that Skype doesn't support is Windows 98, and there aren't official plans for that as of yet.
  • Oh-Phone (Score:2, Informative)

    by DonBiroton ( 765031 )
    I have been using Oh-Phone on MACOSX it works well with OpenPhone and Gnomemeeting. God bless standards
  • by ajs318 ( 655362 ) <[ku.oc.dohshtrae] [ta] [2pser_ds]> on Tuesday June 22, 2004 @06:26AM (#9493210)
    Ever since my boss got friendly with the Asterisk developers, my company's internal telephone network is now almost entirely VOIP. We have a server running Asterisk, with a Zaptel line card (needed a 3V3, 6MHz, 32-bit expansion slot; something you apparently only find on high-end mobo's, as most of the low-cost ones are 33MHz and/or 5V) plugged into an E1 line giving 30 ISDN lines. But you only need this to connect to POTS phones -- connecting to other VOIP phones is just done over the internet. The Asterisk machine also currently runs our intranet, though I'm ordering a new server for all the non-telephonical functions as something keeps crashing (not often enough to be serious, but we need to narrow it down).

    As for phone clients ..... we use dedicated hardware telephones. The Grandstream BudgeTone 101 was the first we evaluated, be aware that this comes with a Continental-style mains adaptor so you may need to get a new power pack (regulated 5 volts 400mA DC + --o)-- - polarity). This works lovely as a SIP telephone but doesn't as standard allow for a headset, which we kind of need in a call centre. The handset does use a standard RJ01 connector, but there seemed no easy way to deal with the receiver switch. We also evaluated every softphone we could get our hands on. In general they seem to be a bitch to get to compile; I had the best result with Linphone, it wasn't as polished as KPhone but it seemed to crash less often; and got absolutely gnowhere with Gnophone. Bear in mind also that a telephone headset will reveal the limitations of the sound chipsets on modern mobo's: you will require a real SoundBlaster-compatiable if you want to be able to understand what anybody is saying. I am running Debian Sid, my boss is running some perversion of Mandrake with a load of stuff from Cooker, and all our workstations run Mandrake 9.2 (hackerish systems are fine for us hackers, but it's more important to have Stuff That Just Works for the masses). We also got a softphone client from Zultys, called LIPZ; which looked stunning but was problematic in practice. It seems to bogart memory and CPU cycles. And when I came to do some hacking on it, I found the real kicker: it doesn't include the source code, so who knows what the hell it's really doing? In the end, we wound up using Zultys ZIP4X4 hardware SIP telephones. These are very expensive for "just" a telephone, but they are stuffed with features, all known codecs, 4 virtual lines, even an integral 5 port (one for the phone, four brought out on RJ45 jacks) 100Mb/s switch, and they are hardware -- my favourite programming language is still 63% tin, 37% lead.

    My honest recommendation would be go for something like the Grandstream, which does everything an "ordinary" phone should do and, being hardware is truly cross-platform. But note, it doesn't have any integral switch so you will take up an extra jack on your ADSL router.
  • Asterisk is king (Score:2, Informative)

    by cfoster611 ( 219409 )
    I've spent the last few weeks putting together a home Asterisk box. VoIP with Asterisk is amazing; the fact that I have a fully functional IP-PBX sitting in my living room running on hardware I found at the dump is mind boggling.

    The IAX protocol, which is a Asterisk-specific VoIP protocol, is great behind my IPCop box since it effortlessly works with NAT, requiring not a STUN server or any other kind of help. I've bought pre-paid VoicePulse Connect service for long distance calls to PSTN, and since I don't
  • ...because it sounds like you're looking for Vonage []. If you decide to switch, please let me know [mailto], I'd like the referral bonux.

The unfacts, did we have them, are too imprecisely few to warrant our certitude.