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Vovida.org 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).

Although there is no perfect solution I have found that STUN works quite well. It suffers most from the fact that there doesn't seem to be a standard for NAT implementation that is used by the vendors. I have used Vovida's free STUN Server and have only had some issues with LinkSys products. Unfortunately it is required to run on a public address. Vovida provides 2 public STUN Servers for people to use: The IP address of the STUN servers are: 128.107.250.38 128.107.250.39 I have not tried these and they may have changed. Maybe someone can give them a try and report back how well they work.

Also, the Linux based SIP softswitch Vovida received significant dev time and resources from Cisco. They even had a contest for the ATA appliances to write the coolest Linux based voice applications. Cisco also has their own commercialized version of the Vovida softswitch, and a bulletproof carrier class SIP server that is meant to run in central offices or large enterprises. It supports Linux and Solaris.

• VOVIDA
• RFC 3261 -- Session Initiation Protocol
• RFC 3263 -- Location of SIP Services
• The SIP Working Group Charter
• Additional SIP WG Info

• The SIPPING Working Group
• Additional SIPPING information

Both SIP and XMPP are XML based.

There is an (apparently) open SIP implementation at www.vovida.org.

If you get the option to use some sort of standard like SIP for your
phone, you can set up your own software call distribution system where
some calls ring your phone, some go to voicemail, some get forwarded to
your mobile etc.

When I was at cisco, these sorts of services were the "bet the company
on it products of the future"

The funny thing is, some of the most interesting implementations of
this sort of thing are open source, one of which is vovidia which got bought by cisco
, but is still operating as an open source operation. The guy who
started has been aquired by cisco twice.

If you get the option to use some sort of standard like SIP for your
phone, you can set up your own software call distribution system where
some calls ring your phone, some go to voicemail, some get forwarded to
your mobile etc.

When I was at cisco, these sorts of services were the "bet the company
on it products of the future"

The funny thing is, some of the most interesting implementations of
this sort of thing are open source, one of which is vovidia which got bought by cisco
, but is still operating as an open source operation. The guy who
started has been aquired by cisco twice.

You are a complete fucktard.

• Telecom giants join Linux effort
• Carrier Grade Linux
• Nortel Succession Communication Server 2000
• Cisco SIP Proxy
• Mitel 3050ICP
• Morotola Carrier Grade Chassis
• Motorola HA Linux
• Motorola VoIP
• Zultys Phones
• Zultys PBX
• SNOM Phones
• SNOM Media Server
• Asterisk
• Vovida

Its amazing how open source voip is unknown. Unfortunately not many people know this even exists. What a shame !

• Linphone (http://www.linphone.org/)
• SIPSet (http://www.vovida.org/)

One thing I don't like about Vonage is that you have to use their Cisco ATA-186(the "POTS-to-Ethernet gizmo" you mentioned). Of course, they password protect it and provision it themselves, so it can only be used with their service. This means you don't get to play with this nifty device, 'cause they've locked you out. Goes against the hacker spirit, seems more like the Microsoft "we've set this up for you for your own protection" thing. I even emailed them to ask if I could use my own ATA, here is their response:

"We do not currently offer service on devices that we do not provide. We do include the Cisco ATA 186 free of charge. We do appreciate your interest in our service. Please do let us know if we may be of further assistance."

Why would you want to configure the Cisco ATA yourself? Well, you might want to try Free World Dialup, or you might want to play with VOCAL from vovida.org. Or whatever.

What I did was to buy a Cisco ATA-186 myself from YesMicro for about $170 with shipping. Then, I got an account at iconnecthere.com and set up my ATA using their setup instructions (it's a Word file, oh well...). I pick up the phone, and it works. When I make a call, they just charge me by the minute (2.9 cents to the U.S.). They have other plans that are cheaper, if you make a lot of calls. If you want to send and receive calls, you can do that for $8.95/month, or $10.95/month for a toll-free number (first hour is included, extra minutes at $0.10/minute). I don't, however, need my own phone number. So, here was my decision-making process, in a nutshell:

With Vonage, if I don't need my own phone number, too bad, no discount; I get a phone number anyway. I still can't tinker with the Cisco ATA, and I still need to give it back (it's not like I could do anything with it anyway, since it's locked down). $39.95 for unlimited calls to the U.S.

With iconnecthere, if I don't need my own phone number, then I don't pay the extra $8.95/month. However, I need to buy the Cisco ATA. Assuming a cost of $170, it would cost me $14.16/month to pay for it. Taking the cost of the Cisco into account, $39.95 buys me 1404 minutes/month, or about 47 minutes/day. Without the cost of the Cisco, it's 1767 minutes/month, or almost an hour/day.

However, I don't make a lot of calls every day. So, with iconnecthere, I can just pay by the minute. Assuming I make about 15 minutes of calls/day, that's $24.16/month including the cost of the Cisco as above, or $10/month not including the cost of the Cisco (with their 1000 minutes for $10 plan). Plus, I have the fun of being able to hack around on the Cisco ATA, and it's mine to keep.

So, in conclusion, if you don't want to hack around on your Cisco ATA, you don't mind giving it back, and you make over an hour's worth of calls every single day, go with Vonage. If you want to hack your Cisco ATA, own it, and make less than an hour's worth of calls a day, iconnecthere seems to be a better option.

How many of these books can one own? I have plenty of books, but the mile-wide-but-foot-deep-overview books get old. I have Glass and Ables' "Unix for Programmers and Users" and Oreilly's "Running Linux". I reference there every so often (in fact just yesterday while installing VOCAL), but I'm not sure there is much more I could get out of a book that wasn't specifically about some library or application.

Most of the quick reference stuff anyone needs is on various websites and discussion boards.

In the SIP community, Linux is used quite extensively. I just returned from an even called SIPIt which is the major interoperability event for SIP based telephony. There were around 50 vendors there -- everyone from big players like Cisco and Polycom to little startups. Many, many people there were using Linux for their products -- I would say at least 50%.

I also have worked with several SIP companies recently, Vovida, and open source SIP stack and suite of applications later aquired by Cisco, and Jasomi, a company that produces telephony boundary control products. These places used Linux extensively as the deployment platform, and there are real working deployments out there using these products.

So for SIP anyway, the answer is a resounding yes!

VOCAL

There is open source code for tons of the traditionally G.7xx CODECs around. The issues many of them require licensing various peoples patents. A casual look at speex would make me think that it is quite likely to infringe someone's CELP patents. Does anyone have any thoughts on this? It's really cool to see something like speex happening but there are a few other things that you might want to think about.

Global IP Sound put out a codec for voice called iLBC. It is specifically designed to avoid infringing known patents. It's sound quality vs. packet loss is very good for IP systems. This is being standardized by the IETF. All the source code is open source and in the draft which you can find at http://www.ietf.org/internet-drafts/draft-ietf-avt -ilbc-codec-00.txt.

Sun has a free implementation of CCITT compression types G.711, G.721 and G.723 at ftp://ftp.cwi.nl/pub/audio/ccitt-adpcm.tar.gz. This is just a free implementation - it does not give you a license to the patents.

Various people including Cisco have been working with the license holders of G.729 IPR to make it available for "pre-commercial" systems, developers, and education. http://www.vovida.org/applications/downloads/G729A /

Asterisk

VOCAL

I know I am just being critical, but this is a very corporate sponsored "open source" project. Look at the document control, the list of contributors, and the Cisco sponsorship. The people that made this are clearly being paid for their efforts, not that that's bad thing, but i think when people think "open source" they think more in the Eric Raymond/Cathedral and the Bizarre kind of way, not a typical corporate software project that happens to have a GPL license.

-Jon

Why wasn't a link to the project's actual webpage in the submission? Here it is.

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?

If you want to implement a VoIP system, there is a bunch of open source software at www.vovida.org that was put there to help make things like this happen.

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There is also some good infromation at www.iptel.org

"rtsp://" protocol, something only realnetworks software can understand

Wrong. RTSP is an open protocol. You can read RFC2326 here. Multiple implementations already exist, including an open-source one.

This linksys/net2phone SIP-in-a-box product was just announced yesterday. What great timing to get it published on /. :-) They haven't even updated their websites yet.

A slightly different version of this service was discussed recently on /.

We've been playing around with a SIP gateway server and a VoIP phone on our DSL connection here in Europe. It works, but phone quality to the US sucks at best. The problem is QoS. Without spending US$10,000++ per month on a dedicated IP pipe from Europe to the US with a guaranteed QoS end-to-end, VoIP just doesn't replace regular phone service. But for IP connections within Europe, we get reasonable quality. Now, if only there were more than 3 people who could call us (and two of those are inside cisco TAC who only call to test their SIP setups)

This linksys/net2phone service requires you to pay them a subscription to use their SIP gateway, and the units probably are not configurable to use alternate SIP services. So if your account expires, your box becomes an expensive blinking light source.

It should work in Europe, I doubt they care which IP block you are coming from. But all the sessions will pass to north america for processing on their VoIP network. If you do buy one of these boxes, drop me a note. I'd love to see what kind of "virtual phone number" they assign you.

the AC