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-users mailing list [email protected] http://lists.nongnu.org/mailman/listinfo/linphone-users
