asterisk users - Jan 2009 - [somewhat OT] seeking ideas/input for my thesis

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

sp4rc wrote:

> 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)
The OpenSER/Kamailio/OpenSIPS technology stack along with XCAP and XMPP 
provides pretty good solutions to this:

http://www.kamailio.org/pub/OpenSER-Summit-2008/15-openser_summit_2008-QuoVadis-anca_vamanu-presence.pdf

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Alex Balashov
Evariste Systems
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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/

John Todd schrieb:
> So here are some projects you might look into:
>   - open-source ZRTP implementation
libzrtpcpp is open source. GPL.
http://www.gnu.org/software/ccrtp/


   Philipp Kempgen

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