[Adium-devl] Jingle, Sip and IAX2 in one step

[Andrew Wellington]

Hi all,

FreeSWITCH can't be used.

It's licensed under the MPL 1.1 (http://www.mozilla.org/MPL/ 
MPL-1.1.html) which is not GPL compatible. Additionally, I'm not even  
sure it can be distributed as a binary as it is. FreeSWITCH as a  
whole claims to be MPL, but the following libraries it uses are of  
various different origins:

Public Domain:
sqlite-3.3.6
g77x codec

Apache License 2.0
apr-1.2.7
apr-util-1.2.7

BSD-like
g72x codec (Sun Microsystems)
gsm codec (although the directory also appears to contain a copy of  
the GPL as COPYING, the source refers to COPYRIGHT which is BSD-like.  
Would need to check that it hasn't come from something GPL somewhere  
and been modified from the BSD-like version)
pcre
srtp

IANA Reference Code
No clear license statement included in code. RFC 3978 appears to  
refer to the license granted to RFCs and is available at http:// 
www.ietf.org/rfc/rfc3978.txt. I have no idea what to file this code  
under, the license isn't very clear.
ilbc

LGPL
libiax
libresample

MPL
libdingaling
libteletone

LGPL/MPL dual license
libspeakup

No Clear License
lpc10 (Also used by Asterisk, OpenSIPStack, linphone. Has been  
generated with a Fortran to C translator.


It looks like in addition to this, there are options to use the  
following libraries:
BSD-like
PortAudio
libspeex

LGPL
libsndfile
iksemel


In conclusion, not only can Adium not use it (MPL is not GPL  
compatible), the licensing of it as a whole is pretty doubtful and  
certainly needs clarifying.

Regards,
Andrew

On 09/11/2006, at 5:13 AM, Brian West wrote:

> hey guys,
> 	As a mac users and one of the very few that have a working cross
> platform jingle library not based on LibJingle at all.  I was
> discussing the approach of Jingle in AdiumX in IRC and was pointed to
> this mailing list.  I was told that Quicktime was going to be used
> for RTP which throws a big monkey wrench into the mix because it
> doesn't do RTP-ICE which is a requirement for speaking with the
> google talk clients.  But on a side note why not just grab FreeSWITCH
> and create a plugin using libFreeSWITCH which does SIP, IAX2 and
> Jingle along with RTP-ICE and STUN and SRTP.  So you could gain all
> those voip protocols in one step.  In addition you could gain the
> ability to write an answering machine module in say JavaScript to
> answer the phone when you're away from your computer.  Any input? (We
> also support SIP SIMPLE)
>
> Thanks,
> Brian