[asterisk-users] [somewhat OT] seeking ideas/input for my thesis

[John Todd]

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:jtodd at digium.com
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/