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[Linphone-developers] SIP service - technical details

From: Simon Morlat
Subject: [Linphone-developers] SIP service - technical details
Date: Wed, 26 Jan 2011 16:38:13 +0100

Dear users and developers,

This second email is to announce a few technical details about the newly launch 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

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.


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