Re: [Sipwitch-devel] Sipwitch and csipsimple failure over WAN

[sipwitch]

> Sipwitch and Yate are open source projects.
>
> You are more than welcome to improve on either to work as you desire. Many 
> brilliant users have.

Oh yes but YATE looks as a misconception and is Sangoma sponsored project, the 
US company with NSA pressure to avoid security :-) FreeSwitch and Yate and 
Asterisk want to preprocess voice streams, transcode them, offer voicemail. The 
first thing people do when they get a new phone is disabling a voicemail :-) 
The way the big VoIP software works is a fundamental misconception.
Sipwitch is the best as far as conception is concerned but it is not finished 
project. I have been using Sipwitch on the local LAN/24 for several months and 
it works. Maybe TLS obfuscation of the SIP data exchange could be useful but in 
a company when we mostly call to each other the ZRTP is enough to talk freely 
about everything. We don’t need to hide identity but just the voice stream need 
to be private. I wanted to step forward and make SIP server under my control. 
For outside we use ostel.co, Linphone, Ekiga and XMPP/Jitsi protocols but I 
wanted to have call logs, at leas part of them under my control :-)
I don’t have idea why doesn’t Sipwitch work from the public IP when others are 
also on the public Ips but a different subnets. Maybe it is simply not finished 
and we all wasting time trying it.
Freeswitch works until you try to use ZRTP.

Is there Eclipse environment for compiling Sipwiych? Programming in pure text 
mode is not convenient. If you help me to setup GUI compiler under Debian and 
compile Sipwitch under it I could look at it :-)