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ohphone http://www.openh323.org/

The open h323 project was started in 1998 and had a soft phone by 2000. This should count as prior art. http://www.openh323.org/fom-serve/cache/3.html

It also stinks that they get to sit on it for 6 years from date of filing. Patents used to be valid for 20 years from date of filing, now a company can sit on it, tweek it, and get 17 years from date of issue (AFIK).

For the love of [insert your god's name here], please, someone make a better client for OpenH323 http://www.openh323.org/ Then I can stop using skype...

I only want it to look a little neater and to work easily with dynamic IP addresses (no, dyndns wont cut it for me, the Average Joe/Jane user)

Troll? No. Clever? Thanks, yes I know, but don't let that frighten you.
As an AC I don't get any real chance to defend my side but I can
promise you that I have built at least three _working_ systems.
Jitter can be solved many ways, simple reordering, windowed backbuffering,
lots of fancy stuff - or for a budget DIY system just don't bother about
it too much, it's not like the OP asked for a fully commercial system is it?
Here's a few obvious links to help. I'm guessing you are enraged by my suggestion
because you work for a commercial VoIP provider. What can I say? Please grow up.
I also guess you have never actually built such such a system either, try it, go on,
fire up that C compiler and amaze yourself at how easy it is. Really, you don't need a load of fancy stuff, VoIP hardly even needs a processor, man you could get a 4MHz Z80 to
do most of what is required.

here
here
here
here
here
and here

OpenH323 (for private VOIP systems)
IPSec Howto (for transparently encrypted VPNs to secure your private VOIP system)

Are there any simple (relatively speaking) SIP servers that can be pressed into service as a Voice-over-IP conferencing server, the way OpenH323's OpenMCU can? I wouldn't really care that it was SIP, except that SIP seems to be the protocol with the greatest selection of open and/or free clients available at the moment.

I'm not thinking here of a full hook-your-telephone-to-the-internet system (which Asterisk seems to be ideal for), just a simple open-standards server for a few people to point their computer-based voice phones at, running on my OWN server, for a casual conference, using readily-acquired free/open software. I've gotten OpenMCU to work for that before, but H.323 seems like it is slowly being replaced by SIP (and there appear to be more SIP clients available than H.323 ones.)

I don't actually know how well it performs but at least it is open source and probably doesn't hog your bandwidth with some supernode technique.

I regularly conduct video conferences between New York, LA, and Atlanta. Standard video conferencing equipment works quite well over 384k ISDN connections via a bridge, and even 128k isn't bad.

I'm guessing that the three pubs aren't that far apart, so 512k should be plenty even with the IP overhead.

I don't know much about running video conferencing over IP, but check into the H.323 standard. I've seen a bit about it on http://www.openh323.org/.

Skype is already mentioned but I'm not sure how it uses the bandwidth for its own purposes, since it is a P2P software. I wouldn't like the idea that skype uses my whole (or major part of) bandwidth for some obscure purposes. OTOH the ad on the skype page tells it has no spy/ad/malware.

Does anybody know good voip software? This one I know but it's a bit spartan and not so easy to use: Skype is well-known but I'm unsure how it uses P2P technique that uses bandwidth for its own purposes.

At my company we use open source projects such as BIND (for an ENUM / DNS based call routing directory) around the edges of our VoiceXML / VOIP IVR hosting service, but not in our core platform.

Originally we did use early open source VOIP projects such as OpenH323. OpenH323 was great, but it needed to be replaced as we moved to SIP and required reliability beyond what OpenH323 offered.

Asterisk is in a similar place - it is a great project that has seen some great early success in voip. I have heard that Vonage, for example, uses it in their voicemail system. I also use it at home and we have several projects at work in the research phase that incorporate it.

Asterisk is not reliable enough for our production environment today - reboots every few weeks to few months are common. As a project it is similar to where Linux was 5+ years ago - plenty of momentum but not quite ready for mission critical use. I have no doubt Asterisk will become as pervasive and reliable as Linux and other leading open source projects have though. Asterisk is an extremely flexible, easy to work with project; and the people involved are also easy to work with and know telephony very well.

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 platforms that way.

There's another protocol called "SIP," for Session Initiation Protocol. H323 is extremely complex (until OpenH323 you needed to pay $$$ to license an ASN1 compiler to compile the protocol into headers so you could code to the spec) and a pain to tunnel over HTTP. SIP was created by some folks who recognized the weaknesses in H323 and decided to create their own protocol.

Bottom line, you can use GnomeMeeting to talk to NetMeeting. I have no idea about to OS X world.

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.

It uses H323 so it's compatible with a lot of stuff. I use GnomeMeeting to chat to Windows users and OS X users. OpenPhone and NetMeeting for Windows, OhPhoneX for OS X.

Also I believe GnomeMeeting is now (just about) usable on OS X with Fink.

Like several of the others, I would recommend OpenH323 . It's fairly easy to set up.

You also have the option of setting up a gateway (hardware isn't cheap, however.) I've used the voicetronix board. This would give the option of calling locally if your or she had other friends in the area.

Also, I've found Voip far easier to use with real (or semi-real) phone hardware. Cisco has the ATA-186, though it operates best with proprietary codecs. I've ended up using a VoipVoice handset -- it's USB and appears as a USB audio device under linux so it's pretty easy to configure, though it's a little weak on the volume. I got mine through IPblue though I am sure there are other sources/options for open-source IP phones.

Creative labs used to sell the voipblaster, and this is still available from ebay, and that lets you use conventional phone hardware -- it's basically a USB-to-phone-jack converter with a hardware codec, and would likely be idea if you're working with anyone who is technology-challenged.

This being a standard protocol, these apps will communicate with each other. However, H.323 relies on UDP communication, which is always a problem with routers. Many routers (such as the 3Com OfficeConnect broadband router) come with built-in "NetMeeting support", ie. H.323 support.

Other applications I know about, but haven't tried: iVisit, Marratech, PictureTalk, vrvs (open source).

Have you looked at OpenH323? There's a multitude of clients that it can communicate with. I've used it to communicate with family who use Windows, and friends who use Macs. As a bonus, video works great too.

There's also an answermachine available compatible with gnomemeeting/netmeeting available. You can find it here: http://www.openh323.org/code.html.

OpenH323 used to support G.723.1 under windows and now it doesn't, what happened to it?
There was once some code that allowed you to use the Microsoft ACM based G.723.1 codec on any windows system with a sound card. This technically worked great, but legally was a bit dubious.

Essentially, the license Microsoft have with the patent holders of G.723.1 is that their codec can only be used with NetMeeting. This means that if anyone else uses it, eg OpenH323, they are in breach of the patent, and possibly Microsofts copyright, but the latter is less likely. The former is definite.

Now, previously we could provide the source and it is up to the user if they wanted to breach patent law. So it was "not our problem", but then came the DMCS legislation and all bets were off. It then meant WE could be held liable if SOMEONE ELSE used our source code!

This sucks badly, but the net result was we pulled the source code from the library.

OpenH323 used to support G.723.1 under windows and now it doesn't, what happened to it?
There was once some code that allowed you to use the Microsoft ACM based G.723.1 codec on any windows system with a sound card. This technically worked great, but legally was a bit dubious.

Essentially, the license Microsoft have with the patent holders of G.723.1 is that their codec can only be used with NetMeeting. This means that if anyone else uses it, eg OpenH323, they are in breach of the patent, and possibly Microsofts copyright, but the latter is less likely. The former is definite.

Now, previously we could provide the source and it is up to the user if they wanted to breach patent law. So it was "not our problem", but then came the DMCS legislation and all bets were off. It then meant WE could be held liable if SOMEONE ELSE used our source code!

This sucks badly, but the net result was we pulled the source code from the library.

If the IBM software

It's probably OpenH323. IBM is smart, they wouldn't bother to re-invent the wheel ... err ... gateway

1. There aren't many state-owned phone companies left in the world due to "Structural Adjustment Programs"
2. Where there are State-Owned Telcos, IBM will probably be paying them for internet connectivity
3. This is really just a sign to sell stock in companies that produce PBX equipment but not VoIP servers/handsets

OpenH323 for more info about VoIP PBX whatevers... or GnomeMeeting for a client so you can start getting your hands dirty now...

• H.323 is the old standard, which almost everybody supports. It's a bit complex and ugly, and looks a lot like the ISDN telco standards. Microsoft Netmeeting supports it for video as well as audio. If you just want to connect two things together, H.323 will work fine, but if you want to build any sort of complex system, it's pretty clumsy. If you want to connect two new fancy systems together, and they're not really compatible, they'll often fall back to H.323.
• SIP is the main new standard, and everybody says they're going to use it Real Soon Now (particularly the VOIP router and PBX folks), though many of them don't actually have it implemented on all of their products yet because they've got too much embedded base. SIP is a much simpler and cleaner protocol, which looks like something written by Internet Unix developers who weren't worried about their embedded base of ISDN telco code.
• The extra half is "Skinny", Cisco's proprietary protocol that most of their IP PBXs and IP phones use, developed before SIP was sufficiently standardized. H.323 was too much baggage, though most of that equipment can fall back to it, and most of it will handle SIP Real Soon Now.
• Yes, Skype is proprietary and closed. Too bad, because it seems to be trying some interesting approaches to user interaction and directory service.
• Speak Freely is one of the best open-but-non-standard systems out there - it was an early attempt to do a crypto phone. Unfortunately, its originator and main developer has decided that there's too much NAT in the world to make it worth continuing to develop it; getting around that takes a major redesign.
• A lot of Instant Messaging systems of various sorts have added VOIP capability.

There's a LOT of open standards VOIP work - see openh323.org and other usual suspects. It turns out that many of the VOIP hardware makers are really happy to fund open standards development so there's something for their equipment to talk to, whether they make voice cards for PCs (either single-user or small PBX cards), or IP PBXs that want more features to make them interesting to users, or boxes that provide some glue function, or whatever, and even Cisco is funding some of them, and some of the little software companies are happy to do open standards work as part of consulting to the hardware people.

New PBXs are pretty much all migrating to IP-based; it's much easier to reuse low-cost PC hardware platforms and build good tools that way. The big PBX makers are generally taking their old PBXs and adding IP features on the side (as opposed to the big router makers adding VOIP boards to connect to old PBXs and telcos), and the real question for most of their customers is when to rip out the old stuff and replace it (for new buildings that need PBXs, it's obvious that IP PBXs are the way to go), because you really start to get operational benefits when you can interconnect multiple locations that way. The PBX industry could have gone to quasi-open standards with ISDN in the late 80s, to take advantage of the reduced development costs and simplicity, but it mostly didn't happen.

The real complexities are the interactions with existing public switched phone companies. There's a huge amount of economic and regulatory baggage built around who pays who how much money when a phone call gets handed off between parties. In the US, there's the originating local telco, the long distance telco (if it's long distance), the delivering telco (if it's not the originating telco), and the Gore Tax folks, all of whom want their cut of the money, and the settlements and pricing aren't really appropriate to the much lower costs of IP telephony, and the prices and regulators are different for intra-state vs. inter-state calls. In the international calling market, this

This was announced months ago. About time Slashdot caught up. Try OpenH323 or something.

A proprietary version of OpenH323! Where do I sign up?

Quicknet Technologies

OpenH323 Project

Sure, that works. Go to Open H.323 for an overview. What you want is called a gateway.

Of course, the real innovation would be to create a Gnutella-like network of people interested in maintaining a local phone line to donate to such a cause. That's an idea I had a couple of years ago. I decided it would be limited to businesses, because the person would pretty much have to have a *spare* phone line. Also, the system would probably attract a lot of leeches, so making sure that each node contributes would be a problem.

One big advantage of Java is that it is a platform, not just a language. You don't have to reinvent the wheel for basic things like threading that modern systems do. In the pdf linked from the article Stroustrup proposes filling out the standard library in ways that Java already does, and think this is a good thing. STL was a start in that direction, but every C++ system I come accross seems to do threading and many other common operation over again (pwlib used in openh323 for example). I'm glad to see the C++ world recognize this type of developer need.

One big advantage of Java is that it is a platform, not just a language. You don't have to reinvent the wheel for basic things like threading that modern systems do. In the pdf linked from the article Stroustrup proposes filling out the standard library in ways that Java already does, and think this is a good thing. STL was a start in that direction, but every C++ system I come accross seems to do threading and many other common operation over again (pwlib used in openh323 for example). I'm glad to see the C++ world recognize this type of developer need.

Netmeeting is an example of an H.323 client, and is capable of communicating with other non-MS H.323 clients such as gnomemeeting with no problems at all.

openH323 has plenty of links and information about this subject, although a lot of it is rather cryptic.

-s

The main VoIP standard is H.323 - Check out OpenH323, an open-source implentation of this technology.

All these comments and not one FOSS reference?

Here is a duct tape and scripting solution:

Get the software at Open H323.

Setup a dedicated MCU server using the OpenMCU conference server (also on above site). Without an MCU server, you can only have one-on-one video conferences. The MCU server will handle multiple participant video conferences as well as multiple rooms for simultaneous but separate conferences.

Use OpenPhone (also at above site) as the conferencing software. Since this is all standards based, the OpenMCU server should also support Netmeeting, Gnomemeeting etc.

This is Slashdot.. so all the advice is gratis but unreliable! Let us know how it works out!

Adi Gadwale.

PS. I have not been able to get this to work for even a 2 person call - Only one of the parties can hear the audio stream.

Take a look at OpenH323.org, they've got a netmeeting compatible application, that also supports the speex codec (so this comment is even on topic! ;-).

It works pretty good (I've tried the windows version), but be aware that H323 needs open UDP ports.

For more on SIP, please see:

• SIP tutorial
• Some Open Source SIP software
• More Open Source SIP-capable software

Yet another perfect application for mesh networking! (And another oppotunity for a shameless plug, but I digress...)

One of the products my company makes is a software mesh for 802.11. We have ported this software to PocketPC, so a device like a Compag^H^H^H^H^H^HHP iPaq with a wireless card can mesh with other devices around it. As nodes go down or enter the network the devices seamlessly configure themselves and route traffic around breaks or congested areas. If the access point you were using went down, you could hop through a neighborss handheld and his neighbor's, so on, until you found an AP.

Of course, you could also do this with free software. Familiar + iPaq + AODV would be a viable open source alternative. Once you have the connectivity you could use just about any app. Gnomemeeting or OpenH323 would enable VoIP. Email apps are there too.

Complexity: I've implemented SIP, H.323, et al., and SIP is just as complex as H.323.

Not true. I've developed two commercial SIP stacks, each about one man year. While I haven't developed an H.323 stack, I've worked with both the OpenH323 and RadVision H.323 stacks, and SIP is much, much simpler.

To wit, RFC 3261 is the largest RFC ever produced and even larger than the H.323 Recommendation.

I don't have a copy of the H.323 Recommendation because I'm not willing to buy it. However, the ITU's H.323 download page shows that the H.323 spec is 2,112,158 bytes in pdf format, while the latest SIP spec is 1,231,871 bytes. This non-conclusive evidence suggests that the H.323 spec is in fact twice as large as the SIP spec.

Add on all of the supporting RFCs and I-Ds, and you have a real mess. Folks even talk about the mess on the SIP reflectors.

H.323 has numerous annexes as well. Most SIP developers speak highly of the protocol, and while there are problems, I certainly wouldn't call it "a mess".

Interoperability: SIP is not more interoperable than H.323. Geez, I think their up to 13 or 14 interop events so far, and I've heard from those in attendance that SIP interoperability is getting harder and harder to achieve.

I have not seen these kinds of interoperability issues at the interops. I have seen many stacks fail the various torture tests, but this does not preclude interoperability. The only reason H.323 is just now achieving interoperability is because of the consolidation of the video conferencing industry into a single dominate vendor: Polycom.

Patents: There are no patents that I am aware of regarding H.323. There are, however, patent claims against some codecs used by H.323, e.g., G.723.1 and G.729, but those same codecs are also used by SIP.

One difference: H.323 requires support of patened codecs, building in a revenue source for those controlling the standard. SIP has no such codec requirements, although I agree that in practice those same codecs are used by many SIP clients.

Moreover, there are several patent claims against SIP (http://www.aful.org/wws/arc/patents/2003-01/msg00 082.html).

This is certainly a potential liability, and it is indeed "aful". Interestingly, those filing patents against IETF standards tend to be frozen out of the standards process, which hopefully acts as a deterant.

Microsoft: I agree that they will hurt SIP more than help. Note that Messenger is merely SIP-based.

Hey, we agree about Microsoft! Yes, Messenger is currently SIP-based (in a bastardized way), but as I said, MS is soon moving to more fully embrace SIP. It will be interesting to see what happens...

CU30 is one choice for 30 fps live video conferencing.

Its under the GPL and MPL and has (or still is) been supported via open h323 as an external library and gnome meeting should work. (I havn't confirmed either)

Qvix Technologies has a propritary version of the same codec, only much more refined. (I used to work for them.)

I initially bought this for three reasons:

1) It's the only portable OGG player right now :-)

2) It's a reasonably full functional, and powerful PDA

3) It runs linux, thus has endless potential -- far more than any proprietary PDA

Already there is a completely opensource replacement OS for the Zaurus.

But... there is no "killer app" yet. A few possibilities though:

1) Real, usable, opensource GPS software using one of the many CF GPS cards out there.

2) Wireless portable OpenH323 terminal

For now, zaurus xmms and zmame are enough to keep me busy for a long time :-)

With a VOIP WiFi "cell phone" you could conceivable talk to anyone in range (peer-to-peer) at no cost, and to anyone connected to the internet if you are in range of a base station.

You even already have an MPL'd H.323 protocol library to provide communication with NetMeeting and GnomeMeeting users. In fact, I've been looking for something like this which could compile on the LinuxARM architecture, in order to turn my iPAQ running Linux into a WiFi cellular phone.

For the most part, voice is covered, we have cell phones, short wave radios, walkie-taklies with a 2 mile range and more! These things are all entrenched, they work, and they are now fairly cheap. Do we need anything else? Would WiFi voice provide better communication than those voice-based devices we are already using?

WiFi would seem like a far better solution than any of those you mention above once you add in the obvious component:

Voice over IP

With a VOIP WiFi "cell phone" you could conceivable talk to anyone in range (peer-to-peer) at no cost, and to anyone connected to the internet if you are in range of a base station.

You even already have an MPL'd H.323 protocol library to provide communication with NetMeeting and GnomeMeeting users. In fact, I've been looking for something like this which could compile on the LinuxARM architecture, in order to turn my iPAQ running Linux into a WiFi cellular phone.

--Cycon

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).

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 with the simple query "SIP" .

No idea about MacOS SIP apps, but the same though as above: MacOSX and Gnome ?

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?

You might want to look at H.323 - http://www.openh323.org/. It's got support for Linux, plus H.323 is used by NetMeeting on Win32, now all you need is a Mac client. Also look at http://www.packetizer.com/h323link.html. You may even be able to do a porting from the openh323 linux code to OSX?? (not sure on this one)

> I'll put dollars to donuts that no major company
> that is pricing their toys at this stratosphere
> will EVER consider anything but BSD.

Netscape. Quicknet. Countless others.

Where are my dollars? (No, you can't pay in stock options >:-> .)

I work for a company doing a lot of wireless VoIP stuff, and can imagine all sorts of cool stuff you could do with this. Just load up OpenH323 and you have a functioning VoIP phone for anywhere. A guy who works with me has written an open source H323 app for WinCE on the iPAQ, and I bet it would be simple to port to a device like this!

Just of note:

The Aplio/Pro is Linux based, and the Aplio/Phone is not (dedicated hardware). Aplio chose to head away from dedicated hardware simply because of the development costs.

All it seems as though they have done is to point the unit at their Gateway/Gatekeeper, which is how the calls get to/from the IP network to phone lines. These devices should quite easily be capable of talking to any gateway/gatekeeper that meets the specs, like Cisco, Ericsson, OpenH323, etc.

If you want to learn more about VOIP, and the H.323 protocol, check out http://www.openh323.org/ which has a wealth of information and links.

sipc

Video Conferencing for Linux

Voice over IP technologies are the same as those used for video conferencing, but with audio codecs only. The two VoIP/VideoConf standards for call setup and control are H.323 and SIP.

There is a suite of free university-developed multicast enabled tools that are time tested and have been in use since 1992. Research rat (robust audio tool) and sdr (session directory) on Freshmeat or Google.

There is also a suite of point-to-point H.323 audio conferencing tools called openh323 which you can find more information out at www.openh323.org. It provides great interoptability with Windows products (Intel, iPhone and Microsoft NetMeeting) as well and can use cisco Systems Voice over IP gatekeepers if you want to link your phone system trunks in.

That should get you started! Have fun!

-Pat

Eeerm, not exactly sure I agree with you on this. In my view Open Source is exactly what telephony needs. You quite rightly pointed out the telephony needs obscene amounts of reliability and uptime, but this is exactly what Open Source is good at.

You get many experts from all around the world to actually look at the code and you get less bugs. Even the thought that your code may be looked at by lots of people much cleverer than you makes you think twice when writing it.

Before you scoff, I've been there and done that. I've spent ten years writing telephony code and managing telephony projects, I've done the SS7 thing, and believe me, the code I've written as Open Source is better quality and has had more thought put into it than the closed source code I have which has millions of calls running through it. For example a lot of my code handles millions of calls a month on British Telecoms internal networks, it ain't pretty bit it works, but the Open Source code I've written not only works, it's pretty too.

If you want some examples of extremely reliable, well thought out and well tested Open Source code check out OpenH323 and OpenGatekeeper. (Okay the last bit is a blantant plug!)

This is probably too late to get noticed but....

Take a look at the OpenH323 effort at www.openh323.org, they have a well thought out, open source, cross platform H.323 stack that is being used by lots of people and is one of the most robust stacks available. In short, it's the dog's danglies.

Contrary to what a lot of people are saying in this discussion, H.323 is real, it works, and it is already in use. Sure it's not going to work great running of you Soundblaster over a congested 33.6 link, but given the right hardware and the right network it works perfectly!