Re: [Linphone-developers] Linphone.org SIP service - technical details

[Simon Morlat]

Hi Kristian,

Indeed I was pretty sure that somebody will ask this question, though it
finally came quicker than I thought :-)

The answer is "why not ?"

Indeed the SER/Kamailio/OpenSIPS has a very impressive feature list,
however quite a few media-procesing related features. But the
web/database ecosystem around it is very complete.

SER/Kamailio/OpenSIPS is the only SIP open-source proxy product, I think
it is always good to have a the choice, like ekiga and linphone for
example.

Before going further to the debate, I have one question for you: did you
try already personaly to install and setup a SER/Kamailio/OpenSIPS by
yourself to setup a SIP network on the public internet ? if yes what
were your impressions ?

Actually, some answers to this question we asked to several people
around us lead us to the conclusion that there could be a strong need
for another SIP proxy, based on different architures and concepts.

SER/Kamailio/OpenSIPS is mostly script oriented (even the SIP routing
and message processing logic is a script). For flexisip we thought that
the best language for doing SIP processing is to use a real language,
like C++, with perhaps java binding in the future. A real programming
language provides strong syntax and type checking, thus less errors.
One idea behind flexisip is: no scripts, just C++ code and interfaces,
and extend flexsip with new features by writing a flexisip C++ module.

Best regards,

Simon


Le mercredi 26 janvier 2011 à 10:57 -0500, Kristian Kielhofner a écrit :
> I'm sure you've expected this but my first question is: Why another SIP proxy?
> 
> The venerable SER family (SER/Kamailio/OpenSIPS), for example, has a
> decade head start on Flexsip.  I'm sure you're well aware but the
> modules/feature list speaks for itself:
> 
> http://www.kamailio.org/docs/modules/stable/
> 
> The stability, scalability, and feature set is unmatched and has been
> for some time.  When Cisco designed the platform for Linksys One four
> years ago, they chose OpenSIPS:
> 
> http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps7194/product_data_sheet0900aecd805c3cc1.pdf
> 
> Where does Flexsip fit in with all of this?
> 
> On Wed, Jan 26, 2011 at 10:38 AM, Simon Morlat
> <address@hidden> wrote:
> > Dear users and developers,
> >
> > This second email is to announce a few technical details about the newly 
> > launch sip.linphone.org SIP service.
> > The service is powered by a sip proxy software called "Flexisip", that we 
> > (Belledonne Communications SARL company) have developed over the past 
> > months.
> >
> > We plan to release it under the GNU Affeiro GPL open-source license as soon 
> > as it is ready for public distribution, that is when we'll enough have 
> > polished its configuration management and documentation.
> >
> > It is written in C/C++ and is based over the LGPL sofia-sip stack.
> >
> > The feature set at this time is:
> > - registration, call routing (the basics)
> > - digest authentication linked to a subscriber database
> > - NAT friendly: it implements all required SIP features required to 
> > workaround nat problems, that is contact fix up, Record-Routes, and of 
> > course media relay for both audio and video streams.
> > - transparent audio transcoding, based on mediastreamer2 media engine (but 
> > this option is not activated in the instance running on sip.linphone.org)
> >
> > Our intent is to make this Flexisip product a SIP proxy easy to deploy, 
> > robust, and easy to extend with media-oriented features.
> >
> > We'll be pleased to answer any questions on the linphone list regarding the 
> > ongoing flexisip development.
> >
> > Simon
> >
> >
> >
> > _______________________________________________
> > Linphone-developers mailing list
> > address@hidden
> > http://lists.nongnu.org/mailman/listinfo/linphone-developers
> >
> 
> 
>