On Jan 19, 2009, at 12:35 PM, sp4rc wrote: > Hello VoIP guys > > Sorry for being somewhat off-topic. At the moment I am studying > informatics in the seventh semester and I need to start thinking about > my thesis. As I am very interested in VoIP technologies I thought > about > picking this as my main topic. So far I have only little experience in > this area. I have been fiddling around with siproxd and pfSense and > have > red the one or the other packet dump containing SIP and RTP traffic, > had > a look into codecs, STUN, etc... but very cursorily, and that's the > reason why I am quite unsure on which track to go. I think I am quite > familiar with many network protocols and devices... so here comes the > question of the questions: > > What would be a great project for my thesis to work on in the VoIP > field? What are topics that still need special development? The time > frame should be around 300 hours but don't take this value too > seriously... > > An idea: contact synchronisation via SIP > Are there any (working or concept) extensions on using SIP to > synchronize contacts > in the way icq does it? (server-side contacts) > > Any ideas are welcome! > /sp4rc
I suppose there are a lot of questions here, actually, since this is a fairly broad topic area you've mentioned. - are you looking to write code to solve a problem? - are you looking for a particularly vexing problem about which to write an analysis paper but write no code? - in what areas have you done work already? Signal analysis? Packet protocols? Hotel management? (the last one is facetious but actually is not entirely non-relevant - Asterisk is lacking a good open-source SMDR interface.) So here are some projects you might look into: - Work with Kristian Kielhofner and make a better signal analysis engine for his Recqual system - SMDR for Asterisk (http://lists.digium.com/pipermail/asterisk-users/2004-June/042854.html ) - steganographic audio multiplexing (http://stegano.net/) - audio encryption/decryption codecs (analog PSTN compatible) - RTP multiplexing for bandwidth savings (see http://lists.digium.com/pipermail/asterisk-dev/2008-December/thread.html#35814) - open-source ZRTP implementation There's a start. Asterisk is a good platform for testing lots of new ideas. Let us know what you might be interested in, and I'm sure there will be good comments from the crowd. JT --- John Todd email:[email protected] Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
