How to run your own open source Skype replacement

Alternatively throw up an instance of apprtc[1], which is a few hundred lines of JS using the new WebRTC APIs. NAT traversal, background noise reduction, gain boosting, bandwidth adaptation all come for free.<p>[1] <a href="http://apprtc.appspot.com/" rel="nofollow">http://apprtc.appspot.com/</a> [2] source: <a href="http://code.google.com/p/webrtc-samples/source/browse/trunk/apprtc/" rel="nofollow">http://code.google.com/p/webrtc-samples/source/browse/trunk/...</a>

the idea that it's a skype replacement while simultaneously warning the reader about avoiding NAT is funny, 50% of the skype secret sauce is the NAT traversal.

I applaud this effort. Jitsi is nothing new and predates all the Skype hype. My guess is it may even be true "end to end," i.e. it relies on no third party "service" that Joe User would find a little too much hassle to run himself, such as XMPP (or even SIP, for that matter).<p>But the first problem I have with Jitsi is that the source is still not open. Looking at current website you have to be a "project member".<p>It's Java-based, right? I want to see that code. If the solution I'm using is less than a few hundred lines of C (quite manageable for any security analysis), why should I blindly (i.e. without seeing the code) switch to Jitsi?<p>These p2p threads are continually entertaining because they prove time and again how many people still think NAT traversal is some sort of "magic" requiring special expertise (e.g., that only Skype or some other private company has).<p>That might just be a myth.<p>Example: Proving a negative. If I can't find a piece of code to do some task does that mean it does not exist? Maybe I just can't find it?<p>agranig himself mentions a couple of things that are in wide use but "little known". Not every solution is going to be widely known. That does not mean such solutions do not exist.<p>re: p2p stuff<p>Read the code before you read the marketing copy.

note that it's just advertisement for a free+commercial solution based on Asterisk (<a href="http://www.asterisk.org/" rel="nofollow">http://www.asterisk.org/</a>)<p>It makes the setup and maintenance easy, basically.<p>Personally I use Asterisk with all the standard SIP clients (CSIPSimple on Android for example)

Why make it particularly SIP and not for example Jingle based? And there is no need to build anything. Just use regular XMPP servers infrastructure. Security can be achieved with ZRTP and OTR, though current implementation of secure XMPP/Jingle clients is still lacking.

I have a Skype-in regular phone number. I did not see how to set that up and without some sort of local number availability, for me at least, this is no "Skype Replacement"

A bit bold to talk about running your own server to replace Skype. You'd need to get 32 million users to switch over.

I think that the most likely thing to disrupt Skype is going to be WebRTC and getUserMedia/PeerConnection.

What about android clients for this?

It appears to be based on Asterisk, with Mysql and Apache.