[Free-RTC] Escape from PRISM!

Sylvain Berfini sylvain.berfini at linphone.org
Tue Oct 8 16:36:27 CEST 2013


Hi Lee,

I tried your service with Linphone (one I compiled myself with ZRTP
enabled, the one in the Google Play Store doesn't embed it yet), and it
works great (at least with your echo service).

Regards,
Sylvain


2013/6/14 Lee Azzarello <lee at rockingtiger.com>

> I operate a secure SIP service at https://ostel.co
>
> I'm using rtpproxy for NAT traversal. Since the audio stream is
> encrypted with ZRTP, the server doesn't have access to the content of
> calls.
>
> During the course of this project, I have discovered numerous
> complications. The nut I'm trying to crack right now is SIP
> federation. I have alias calling working via SRV records but calling
> between SIP domains is pending. The closest I came so far is a call
> between ostel.co and ekiga.net succeeded but one of the two providers
> is mangling the rtp stream so that the ZRTP key agreement never
> happens.
>
> I'll post my progress when I get that nailed down. I also have a
> milestone on my roadmap to build a Chef cookbook to automate the
> assembly of a full secure SIP stack.
>
> Regards,
> Lee
>
> On Tue, Jun 11, 2013 at 11:12 AM, MJ Ray <mjr at phonecoop.coop> wrote:
> > Daniel Pocock <daniel at pocock.com.au>
> >> There are also some sore points, e.g. the Debian 7 release includes an
> >> Empathy version that only works with Google's gmail TURN server and
> >> not the Debian packaged TURN server.  Things like that are going to
> >> continue frustrating users for some time to come but will eventually
> >> be ironed out.
> >
> > I think it's a far bigger frustration that we can't actually call
> > anyone who's not using the same system yet, and sometimes it won't
> > even work on the same system, while the debugging logs on either
> > client or provider often seem to be insufficient to actually find out
> > why something doesn't work.
> >
> > I've got jitsi working over XMPP, but damned if I can see how to let
> > others call in, or call outside the same system with it :-/
> >
> > The lumicall site says things like "If the VoIP provider supports SIP,
> > TLS and ICE, it should work" which is great, but what UK VoIP DDI
> > provider supports all three?  In short, part of the necessary
> > information for mass use seems to be missing still.  If anyone has it
> > working, please post step-by-step instructions so others can set
> > things up and help spread the spoken word.
> >
> > Thanks,
> > --
> > MJ Ray (slef), member of www.software.coop, a for-more-than-profit
> co-op.
> > http://koha-community.org supporter, web and library systems developer.
> > In My Opinion Only: see http://mjr.towers.org.uk/email.html
> > Available for hire (including development) at http://www.software.coop/
> > _______________________________________________
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> > Free-RTC at lists.fsfe.org
> > https://lists.fsfe.org/mailman/listinfo/free-rtc
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