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[Linphone-users] Linphone.org SIP service - technical details
From: |
Simon Morlat |
Subject: |
[Linphone-users] Linphone.org 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.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
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