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https://jitsi.org/GSOC2010/Kamailio4575Accepted


http://opensips-open-sip-server.1449251.n2.nabble.com/No-Voice-Comm-in-Conference-call-td7580232.html


http://www.in2eps.com/fo-sip/tk-fo-sip-service-11.html


http://wiki.cs.columbia.edu/download/attachments/576/SIP+Conferencing.pdf

GSoC Student: Marius-Ovidiu Bucur - (Romania) 
Mentors: Daniel-Constantin Mierla (Romania/Germany) 

PROJECT REQUIREMENTS ( SHOW )

In case you’ve already participated in conference phone calls (which are basically confs with many participants) then you most probably had to simply dial a number and then somehow started hearing everyone. This is how things have been happening in conventional telephony for quite a while and this is how they happen today with VoIP.

In the case of VoIP, however, the approach is not all that sophisticated since VoIP clients would have the impression they are calling a regular participant and they would hence present you with their regular call interface. This works of course, but why settle for it when we could have more :). Wouldn’t it be nice for example if you could see who else is on the call? Wouldn’t it be even better to know who’s currently speaking?

We think this is important and so do the members of the popular Kamailio (OpenSER) development team. We are therefore joining up in this project and need your help to add the necessary code to Kamailio.

kamailio.png

In the SIP specification universe (or in other words in the IETF), conference calls are described by RFC 4353, and RFC 4575. The basic differences between these two are explained in these slides but you’d still need to have a look at the specs :).

So to sum it up, this project is about the implementation of conference signalling in the Kamailio (OpenSER) server. It means implementing support for the following standards:

  • RFC 4353: A Framework for Conferencing with SIP
  • RFC 4575: A SIP Event Package for Conference State

Interested? Then looking forward to reading your application!

Note that this project will be mentored by members of the Kamailio (OpenSER) development team so you’ll have all the expert help you need!

References:

Kamailio (OpenSER) – the Open Source SIP Server
http://kamailio.org

A SIP Event Package for Conference State
http://tools.ietf.org/html/rfc4575

A Framework for Conferencing with SIP 
http://tools.ietf.org/html/rfc4353

Support for conference calls in SIP Communicator
http://sip-communicator.org/gsoc2010/SIP.Communicator@FOSDEM-2010-02-06-updated.pdf

Other Jitsi GSoC Projects 
http://gsoc.jitsi.org

Jitsi Developer Documentation
http://www.jitsi.org/index.php/Documentation/DeveloperDocumentation

The official Jitsi website 
http://www.jitsi.org

조회 수 :
58613
등록일 :
2014.03.12
12:31:17 (*.251.139.148)
엮인글 :
http://webs.co.kr/index.php?document_srl=39231&act=trackback&key=3ce
게시글 주소 :
http://webs.co.kr/index.php?document_srl=39231
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49 telepresence: Open Source SIP Telepresence/MCU admin 123000   2014-03-12
 
48 SIP PBX - OpenSIPS and Asterisk configuration admin 105665   2014-03-12
 
» Conference Support in Kamailio (OpenSER) admin 58613   2014-03-12
https://jitsi.org/GSOC2010/Kamailio4575Accepted http://opensips-open-sip-server.1449251.n2.nabble.com/No-Voice-Comm-in-Conference-call-td7580232.html http://www.in2eps.com/fo-sip/tk-fo-sip-service-11.html http://wiki.cs.columbia.edu/do...  
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44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 26045   2014-03-12
 
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