sipwitch-devel
[Top][All Lists]
Advanced

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

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


From: David Sugar
Subject: Re: [Sipwitch-devel] Sipwitch and csipsimple failure over WAN
Date: Sun, 15 Jun 2014 10:25:45 -0400
User-agent: Mozilla/5.0 (X11; Linux x86_64; rv:24.0) Gecko/20100101 Thunderbird/24.5.0

In regard to Yate, asterisk, etc, many of these designed around some
form of rather generic multi-protocol support including traditional
trunking, etc, and certainly incorporating (unencrypted) media through a
central point becomes necessary for doing that.  Each are I am sure
excellent projects for what they choose to do.  Sipwitch was expressly
designed around the question of having encrypted peer media paths and a
single common network signalling/session protocol (sip).  Connectivity
to other kinds of networks cannot be guaranteed as secure, but could be
provided through sip gateways talking to sipwitch, there is support for
routing rules for doing that.

Again if there is more we can do to make sipwitch easier for others to
work with, I would be happy to do that.

On 06/15/2014 10:13 AM, David Sugar wrote:
> Generally I had been working on making it friendly to also build with
> cmake and recently reorganized namespaces and some other things to make
> it (and ucommon) easier to use directly with qt creator as an ide.  We
> also have a single checkout bootstrap that creates a top level cmake
> project for all the packages, which can then be used in a single qt
> creator project.  We have our own build server for debian/raspian, and
> use osc for some other platforms.  We also have been able to use mingw
> to build for windows.  I am not sure what else we could do to make it
> easier to work with the code or deploy packages, other than document
> things better...
> 
> On 06/15/2014 07:38 AM, address@hidden wrote:
>>> 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 :-)
>>

Attachment: dyfet.vcf
Description: Vcard


reply via email to

[Prev in Thread] Current Thread [Next in Thread]